From 102646085ab530a0e155d2bde424589b83ef5a89 Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Fri, 1 Feb 2008 13:12:12 +0100 Subject: [PATCH 001/250] [ALSA] soc - Mono voice playback volume for WM8753 Voice playback volume is in register bits 0:2, not 4:6. From: Mike Montour Signed-off-by: Mark Brown Cc: Werner Almesberger Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ddd9c71b3fde..02882758415a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -279,7 +279,7 @@ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0 SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1), +SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), SOC_ENUM("Bass Boost", wm8753_enum[0]), From dc952e693e3653d0fa50ee1986a47d88de3465b7 Mon Sep 17 00:00:00 2001 From: Kristoffer Ericson Date: Fri, 1 Feb 2008 13:16:10 +0100 Subject: [PATCH 002/250] [ALSA] Add SUPERH depends to sound/soc/sh/Kconfig Currently you will see an empty "SoC Audio support for SuperH" menu when building for other archs (example pxa). This patch adds "depends on SUPERH" to remove that empty menu. Signed-off-by: Kristoffer Ericson Signed-off-by: Takashi Iwai --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index f03220d23e73..4c1e013381c9 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,4 +1,5 @@ menu "SoC Audio support for SuperH" + depends on SUPERH config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" From bf277785d6921b8a9f8339ad5ad632aef6cae73c Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Sun, 3 Feb 2008 20:31:47 +0100 Subject: [PATCH 003/250] [ALSA] HDA - Add support for the OQO Model 2 This patch adds support for the OQO Model 2 Ultra Mobile PC. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index caf48edaa921..4c3c4e6ce3d6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -39,6 +39,7 @@ enum { STAC_REF, + STAC_9200_OQO, STAC_9200_DELL_D21, STAC_9200_DELL_D22, STAC_9200_DELL_D23, @@ -1052,9 +1053,15 @@ static unsigned int dell9200_m27_pin_configs[8] = { 0x90170310, 0x04a11020, 0x90170310, 0x40f003fc, }; +static unsigned int oqo9200_pin_configs[8] = { + 0x40c000f0, 0x404000f1, 0x0221121f, 0x02211210, + 0x90170111, 0x90a70120, 0x400000f2, 0x400000f3, +}; + static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_REF] = ref9200_pin_configs, + [STAC_9200_OQO] = oqo9200_pin_configs, [STAC_9200_DELL_D21] = dell9200_d21_pin_configs, [STAC_9200_DELL_D22] = dell9200_d22_pin_configs, [STAC_9200_DELL_D23] = dell9200_d23_pin_configs, @@ -1069,6 +1076,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_REF] = "ref", + [STAC_9200_OQO] = "oqo", [STAC_9200_DELL_D21] = "dell-d21", [STAC_9200_DELL_D22] = "dell-d22", [STAC_9200_DELL_D23] = "dell-d23", @@ -1153,6 +1161,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { STAC_9200_GATEWAY), SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707", STAC_9200_GATEWAY), + /* OQO Mobile */ + SND_PCI_QUIRK(0x1106, 0x3288, "OQO Model 2", STAC_9200_OQO), {} /* terminator */ }; @@ -3147,7 +3157,8 @@ static int patch_stac9200(struct hda_codec *codec) spec->num_adcs = 1; spec->num_pwrs = 0; - if (spec->board_config == STAC_9200_GATEWAY) + if (spec->board_config == STAC_9200_GATEWAY || + spec->board_config == STAC_9200_OQO) spec->init = stac9200_eapd_init; else spec->init = stac9200_core_init; From 09f99701393c7b66bde01df6c292fe5d9f843033 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Feb 2008 12:31:13 +0100 Subject: [PATCH 004/250] [ALSA] hda-codec - Allow multiple SPDIF devices The current code doesn't allow multiple SPDIF devices, and causes errors when multiple SPDIF devices are found (e.g. SPDIF out and HDMI). This patch allows multiple SPDIF devices by incrementing the index automatically. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 35 +++++++++++++++++++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 37c413923db8..ab3bb7997cd2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1037,16 +1037,24 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, } /* find a mixer control element with the given name */ -struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, - const char *name) +static struct snd_kcontrol * +_snd_hda_find_mixer_ctl(struct hda_codec *codec, + const char *name, int idx) { struct snd_ctl_elem_id id; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id.index = idx; strcpy(id.name, name); return snd_ctl_find_id(codec->bus->card, &id); } +struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, + const char *name) +{ + return _snd_hda_find_mixer_ctl(codec, name, 0); +} + /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves) @@ -1481,6 +1489,8 @@ static struct snd_kcontrol_new dig_mixes[] = { { } /* end */ }; +#define SPDIF_MAX_IDX 4 /* 4 instances should be enough to probe */ + /** * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls * @codec: the HDA codec @@ -1496,9 +1506,20 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) int err; struct snd_kcontrol *kctl; struct snd_kcontrol_new *dig_mix; + int idx; + for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { + if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Playback Switch", + idx)) + break; + } + if (idx >= SPDIF_MAX_IDX) { + printk(KERN_ERR "hda_codec: too many IEC958 outputs\n"); + return -EBUSY; + } for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + kctl->id.index = idx; kctl->private_value = nid; err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) @@ -1595,7 +1616,17 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) int err; struct snd_kcontrol *kctl; struct snd_kcontrol_new *dig_mix; + int idx; + for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { + if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Capture Switch", + idx)) + break; + } + if (idx >= SPDIF_MAX_IDX) { + printk(KERN_ERR "hda_codec: too many IEC958 inputs\n"); + return -EBUSY; + } for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); kctl->private_value = nid; From 12a733e56c0f1c78bd34bf36e9765463fd51c88e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Feb 2008 12:32:20 +0100 Subject: [PATCH 005/250] [ALSA] hda-codec - Add SI HDMI codec support Added the support of SI HDMI codec, found in ASUS machines. ALSA bug#3654 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3654 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_atihdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 9a8bb4ce3f8d..27d2e007404b 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -58,6 +58,10 @@ static int atihdmi_build_controls(struct hda_codec *codec) static int atihdmi_init(struct hda_codec *codec) { snd_hda_sequence_write(codec, atihdmi_basic_init); + /* SI codec requires to unmute the pin */ + if (get_wcaps(codec, 0x03) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, 0x03, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); return 0; } @@ -158,5 +162,6 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, + { .id = 0x10951392, .name = "SI HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; From f12462c5224bf992f5ed4d37af4d42622f7d5934 Mon Sep 17 00:00:00 2001 From: Mirco Tischler Date: Mon, 4 Feb 2008 12:33:59 +0100 Subject: [PATCH 006/250] [ALSA] hda-codec - Add support of Zepto laptops Adds support for zepto laptops with alc268 intel_hda codec. Signed-off-by: Mirco Tischler Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index e985cf5e0410..9a56b9b273cd 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -826,6 +826,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. toshiba Toshiba A205 acer Acer laptops dell Dell OEM laptops (Vostro 1200) + zepto Zepto laptops test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33282f9c01c7..45e661e42c0b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ enum { ALC268_TOSHIBA, ALC268_ACER, ALC268_DELL, + ALC268_ZEPTO, #ifdef CONFIG_SND_DEBUG ALC268_TEST, #endif @@ -10105,6 +10106,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", [ALC268_DELL] = "dell", + [ALC268_ZEPTO] = "zepto", #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = "test", #endif @@ -10122,6 +10124,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), + SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), {} }; @@ -10182,6 +10185,22 @@ static struct alc_config_preset alc268_presets[] = { .init_hook = alc268_dell_init_hook, .input_mux = &alc268_capture_source, }, + [ALC268_ZEPTO] = { + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc268_toshiba_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .dig_out_nid = ALC268_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc268_toshiba_unsol_event, + .init_hook = alc268_toshiba_automute + }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { .mixers = { alc268_test_mixer, alc268_capture_mixer }, From f339eb0f30e6598c1d3f91b01a3e634364fab7a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Feb 2008 12:34:59 +0100 Subject: [PATCH 007/250] [ALSA] ice1724 - Enable AK4114 support for Audiophile192 Fixed and enabled the support of AK4114 chip on Audiophile192. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/revo.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 301bf929acd9..1d3b1ebf9c9b 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -353,28 +353,20 @@ static struct snd_ak4xxx_private akm_ap192_priv __devinitdata = { .cif = 0, .data_mask = VT1724_REVO_CDOUT, .clk_mask = VT1724_REVO_CCLK, - .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS3, - .cs_addr = VT1724_REVO_CS3, - .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS3, + .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1, + .cs_addr = VT1724_REVO_CS1, + .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1, .add_flags = VT1724_REVO_CCLK, /* high at init */ .mask_flags = 0, }; -#if 0 -/* FIXME: ak4114 makes the sound much lower due to some confliction, - * so let's disable it right now... - */ -#define BUILD_AK4114_AP192 -#endif - -#ifdef BUILD_AK4114_AP192 /* AK4114 support on Audiophile 192 */ /* CDTO (pin 32) -- GPIO2 pin 52 * CDTI (pin 33) -- GPIO3 pin 53 (shared with AK4358) * CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358) * CSN (pin 35) -- GPIO7 pin 59 */ -#define AK4114_ADDR 0x00 +#define AK4114_ADDR 0x02 static void write_data(struct snd_ice1712 *ice, unsigned int gpio, unsigned int data, int idx) @@ -428,7 +420,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) tmp = snd_ice1712_gpio_read(ice); tmp |= VT1724_REVO_CCLK; /* high at init */ tmp |= VT1724_REVO_CS0; - tmp &= ~VT1724_REVO_CS3; + tmp &= ~VT1724_REVO_CS1; snd_ice1712_gpio_write(ice, tmp); udelay(1); return tmp; @@ -436,7 +428,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp) { - tmp |= VT1724_REVO_CS3; + tmp |= VT1724_REVO_CS1; tmp |= VT1724_REVO_CS0; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -485,13 +477,13 @@ static int __devinit ap192_ak4114_init(struct snd_ice1712 *ice) struct ak4114 *ak; int err; - return snd_ak4114_create(ice->card, + err = snd_ak4114_create(ice->card, ap192_ak4114_read, ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &ak); + return 0; /* error ignored; it's no fatal error */ } -#endif /* BUILD_AK4114_AP192 */ static int __devinit revo_init(struct snd_ice1712 *ice) { @@ -588,11 +580,9 @@ static int __devinit revo_add_controls(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; -#ifdef BUILD_AK4114_AP192 err = ap192_ak4114_init(ice); if (err < 0) return err; -#endif break; } return 0; From a76969228a5b341f9c968abbc6eb7655ac3734e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Feb 2008 12:36:32 +0100 Subject: [PATCH 008/250] [ALSA] ice1724 - Add ADC setup in set_rate callback for Audiophile192 Added the missing GPIO setup for the AK5385A ADC codec on Audiophile192. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/revo.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1d3b1ebf9c9b..7c930cc05f1d 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -322,17 +322,23 @@ static struct snd_pt2258 ptc_revo51_volume; static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) { struct snd_ice1712 *ice = ak->private_data[0]; + int dfs; revo_set_rate_val(ak, rate); -#if 1 /* FIXME: do we need this procedure? */ - /* reset DFS pin of AK5385A for ADC, too */ - /* DFS0 (pin 18) -- GPIO10 pin 77 */ - snd_ice1712_save_gpio_status(ice); - snd_ice1712_gpio_write_bits(ice, 1 << 10, - rate > 48000 ? (1 << 10) : 0); - snd_ice1712_restore_gpio_status(ice); -#endif + /* reset CKS */ + snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 : 0); + /* reset DFS pins of AK5385A for ADC, too */ + if (rate > 96000) + dfs = 2; + else if (rate > 48000) + dfs = 1; + else + dfs = 0; + snd_ice1712_gpio_write_bits(ice, 3 << 9, dfs << 9); + /* reset ADC */ + snd_ice1712_gpio_write_bits(ice, 1 << 11, 0); + snd_ice1712_gpio_write_bits(ice, 1 << 11, 1); } static const struct snd_akm4xxx_dac_channel ap192_dac[] = { @@ -549,6 +555,9 @@ static int __devinit revo_init(struct snd_ice1712 *ice) if (err < 0) return err; + /* unmute all codecs */ + snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, + VT1724_REVO_MUTE); break; } From 6c4cc3a8ed15aacc06a5fd369639fef633cee2bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Feb 2008 12:44:11 +0100 Subject: [PATCH 009/250] [ALSA] Add more fallbacks to OSS PHONEOUT mixer map Added more fallbacks to OSS PHONEOUT mixer mapping. This corresponds to the speaker output in general, so now "Mono" and "Speaker" are assigned. Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 75daed298a15..581aa2c60e65 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1257,6 +1257,8 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_DIGITAL3, "Digital", 2 }, { SOUND_MIXER_PHONEIN, "Phone", 0 }, { SOUND_MIXER_PHONEOUT, "Master Mono", 0 }, + { SOUND_MIXER_PHONEOUT, "Speaker", 0 }, /*fallback*/ + { SOUND_MIXER_PHONEOUT, "Mono", 0 }, /*fallback*/ { SOUND_MIXER_PHONEOUT, "Phone", 0 }, /* fallback */ { SOUND_MIXER_VIDEO, "Video", 0 }, { SOUND_MIXER_RADIO, "Radio", 0 }, From 7ba72ba1fe891a94b1e9d506236507e4dc50e872 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Feb 2008 14:03:20 +0100 Subject: [PATCH 010/250] [ALSA] hda-intel - Fix PCM device number assignment In the current scheme, PCM device numbers are assigned incrementally in the order of codecs. This causes problems when the codec number is irregular, e.g. codec #0 for HDMI and codec #1 for analog. Then the HDMI becomes the first PCM, which is picked up as the default output device. Unfortuantely this doesn't work well with normal setups. This patch introduced the fixed device numbers for the PCM types, namely, analog, SPDIF, HDMI and modem. The PCM devices are assigned according to the corresponding PCM type. After this patch, HDMI will be always assigned to PCM #3, SPDIF to PCM #1, and the first analog to PCM #0, etc. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 12 ++++- sound/pci/hda/hda_intel.c | 89 ++++++++++++++++++---------------- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_atihdmi.c | 1 + sound/pci/hda/patch_cmedia.c | 1 + sound/pci/hda/patch_conexant.c | 1 + sound/pci/hda/patch_realtek.c | 1 + sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + sound/pci/hda/patch_via.c | 1 + 10 files changed, 66 insertions(+), 44 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index f14871151be9..301b5227bfb1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -590,11 +590,21 @@ struct hda_pcm_stream { struct hda_pcm_ops ops; }; +/* PCM types */ +enum { + HDA_PCM_TYPE_AUDIO, + HDA_PCM_TYPE_SPDIF, + HDA_PCM_TYPE_HDMI, + HDA_PCM_TYPE_MODEM, + HDA_PCM_NTYPES +}; + /* for PCM creation */ struct hda_pcm { char *name; struct hda_pcm_stream stream[2]; - unsigned int is_modem; /* modem codec? */ + unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */ + int device; /* assigned device number */ }; /* codec information */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4be36c84b36c..18475de074b2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -211,9 +211,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) /* max number of PCM devics per card */ -#define AZX_MAX_AUDIO_PCMS 6 -#define AZX_MAX_MODEM_PCMS 2 -#define AZX_MAX_PCMS (AZX_MAX_AUDIO_PCMS + AZX_MAX_MODEM_PCMS) +#define AZX_MAX_PCMS 8 /* RIRB int mask: overrun[2], response[0] */ #define RIRB_INT_RESPONSE 0x01 @@ -350,7 +348,6 @@ struct azx { struct azx_dev *azx_dev; /* PCM */ - unsigned int pcm_devs; struct snd_pcm *pcm[AZX_MAX_PCMS]; /* HD codec */ @@ -1386,7 +1383,7 @@ static void azx_pcm_free(struct snd_pcm *pcm) } static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, - struct hda_pcm *cpcm, int pcm_dev) + struct hda_pcm *cpcm) { int err; struct snd_pcm *pcm; @@ -1400,7 +1397,7 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_assert(cpcm->name, return -EINVAL); - err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, + err = snd_pcm_new(chip->card, cpcm->name, cpcm->device, cpcm->stream[0].substreams, cpcm->stream[1].substreams, &pcm); @@ -1423,59 +1420,67 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 1024 * 64, 1024 * 1024); - chip->pcm[pcm_dev] = pcm; - if (chip->pcm_devs < pcm_dev + 1) - chip->pcm_devs = pcm_dev + 1; - + chip->pcm[cpcm->device] = pcm; return 0; } static int __devinit azx_pcm_create(struct azx *chip) { + static const char *dev_name[HDA_PCM_NTYPES] = { + "Audio", "SPDIF", "HDMI", "Modem" + }; + /* starting device index for each PCM type */ + static int dev_idx[HDA_PCM_NTYPES] = { + [HDA_PCM_TYPE_AUDIO] = 0, + [HDA_PCM_TYPE_SPDIF] = 1, + [HDA_PCM_TYPE_HDMI] = 3, + [HDA_PCM_TYPE_MODEM] = 6 + }; + /* normal audio device indices; not linear to keep compatibility */ + static int audio_idx[4] = { 0, 2, 4, 5 }; struct hda_codec *codec; int c, err; - int pcm_dev; + int num_devs[HDA_PCM_NTYPES]; err = snd_hda_build_pcms(chip->bus); if (err < 0) return err; /* create audio PCMs */ - pcm_dev = 0; + memset(num_devs, 0, sizeof(num_devs)); list_for_each_entry(codec, &chip->bus->codec_list, list) { for (c = 0; c < codec->num_pcms; c++) { - if (codec->pcm_info[c].is_modem) - continue; /* create later */ - if (pcm_dev >= AZX_MAX_AUDIO_PCMS) { - snd_printk(KERN_ERR SFX - "Too many audio PCMs\n"); - return -EINVAL; + struct hda_pcm *cpcm = &codec->pcm_info[c]; + int type = cpcm->pcm_type; + switch (type) { + case HDA_PCM_TYPE_AUDIO: + if (num_devs[type] >= ARRAY_SIZE(audio_idx)) { + snd_printk(KERN_WARNING + "Too many audio devices\n"); + continue; + } + cpcm->device = audio_idx[num_devs[type]]; + break; + case HDA_PCM_TYPE_SPDIF: + case HDA_PCM_TYPE_HDMI: + case HDA_PCM_TYPE_MODEM: + if (num_devs[type]) { + snd_printk(KERN_WARNING + "%s already defined\n", + dev_name[type]); + continue; + } + cpcm->device = dev_idx[type]; + break; + default: + snd_printk(KERN_WARNING + "Invalid PCM type %d\n", type); + continue; } - err = create_codec_pcm(chip, codec, - &codec->pcm_info[c], pcm_dev); + num_devs[type]++; + err = create_codec_pcm(chip, codec, cpcm); if (err < 0) return err; - pcm_dev++; - } - } - - /* create modem PCMs */ - pcm_dev = AZX_MAX_AUDIO_PCMS; - list_for_each_entry(codec, &chip->bus->codec_list, list) { - for (c = 0; c < codec->num_pcms; c++) { - if (!codec->pcm_info[c].is_modem) - continue; /* already created */ - if (pcm_dev >= AZX_MAX_PCMS) { - snd_printk(KERN_ERR SFX - "Too many modem PCMs\n"); - return -EINVAL; - } - err = create_codec_pcm(chip, codec, - &codec->pcm_info[c], pcm_dev); - if (err < 0) - return err; - chip->pcm[pcm_dev]->dev_class = SNDRV_PCM_CLASS_MODEM; - pcm_dev++; } } return 0; @@ -1587,7 +1592,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - for (i = 0; i < chip->pcm_devs; i++) + for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) snd_hda_suspend(chip->bus, state); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index c8649282c2cf..7286ab86ecc4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -359,6 +359,7 @@ static int ad198x_build_pcms(struct hda_codec *codec) info++; codec->num_pcms++; info->name = "AD198x Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; if (spec->dig_in_nid) { diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 27d2e007404b..e0e9ea995684 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -116,6 +116,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = "ATI HDMI"; + info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback; return 0; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 3d6097ba1d68..99ce74b4e9fc 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -571,6 +571,7 @@ static int cmi9880_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "CMI9880 Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7206b30cbf94..bb915ede0ceb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -284,6 +284,7 @@ static int conexant_build_pcms(struct hda_codec *codec) info++; codec->num_pcms++; info->name = "Conexant Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 45e661e42c0b..85ea3f82de19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2499,6 +2499,7 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 2; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid && spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index d22f5a6b850f..598ee2119bbe 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -206,7 +206,7 @@ static int si3054_build_pcms(struct hda_codec *codec) info->name = "Si3054 Modem"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; - info->is_modem = 1; + info->pcm_type = HDA_PCM_TYPE_MODEM; return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4c3c4e6ce3d6..f693011d25a0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1899,6 +1899,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4e5dd4cf36f5..d9a5c6a2dd9f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -523,6 +523,7 @@ static int via_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = spec->stream_name_digital; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); From 7d664ed58fb54bc421d5fe8e5e0adec736fd0558 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Feb 2008 14:41:59 +0100 Subject: [PATCH 011/250] [ALSA] hda-codec - Add ID for an unknown HDMI codec chip Added the ID for an unknown HDMI codec chip on Jetway J9F2. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_atihdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index e0e9ea995684..e6fbd5d8fba4 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -164,5 +164,6 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, { .id = 0x10951392, .name = "SI HDMI", .patch = patch_atihdmi }, + { .id = 0x17e80047, .name = "Unknown HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; From d043143d803ad9a9f4a66d6a833876735fb7b869 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Wed, 6 Feb 2008 14:43:54 +0100 Subject: [PATCH 012/250] [ALSA] ice1712 - Fix hoontech MIDI input Fixes the problems with Midi In on Hoontech/STA dsp24 cards, for example with DSP2000 box, without restricting the box configurations available. Also adds mpu_401 name strings. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai --- sound/pci/ice1712/hoontech.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index cf5c7c0898fd..6914189073a4 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -208,6 +208,19 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) /* ICE1712_STDSP24_MUTE | ICE1712_STDSP24_INSEL | ICE1712_STDSP24_DAREAR; */ + /* These boxconfigs have caused problems in the past. + * The code is not optimal, but should now enable a working config to + * be achieved. + * ** MIDI IN can only be configured on one box ** + * ICE1712_STDSP24_BOX_MIDI1 needs to be set for that box. + * Tests on a ADAC2000 box suggest the box config flags do not + * work as would be expected, and the inputs are crossed. + * Setting ICE1712_STDSP24_BOX_MIDI1 and ICE1712_STDSP24_BOX_MIDI2 + * on the same box connects MIDI-In to both 401 uarts; both outputs + * are then active on all boxes. + * The default config here sets up everything on the first box. + * Alan Horstmann 5.2.2008 + */ spec->boxconfig[0] = ICE1712_STDSP24_BOX_CHN1 | ICE1712_STDSP24_BOX_CHN2 | ICE1712_STDSP24_BOX_CHN3 | @@ -223,14 +236,14 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) (spec->config & ICE1712_STDSP24_MUTE) ? 1 : 0); snd_ice1712_stdsp24_insel(ice, (spec->config & ICE1712_STDSP24_INSEL) ? 1 : 0); - for (box = 0; box < 1; box++) { + for (box = 0; box < 4; box++) { if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2) snd_ice1712_stdsp24_midi2(ice, 1); for (chn = 0; chn < 4; chn++) snd_ice1712_stdsp24_box_channel(ice, box, chn, (spec->boxconfig[box] & (1 << chn)) ? 1 : 0); - snd_ice1712_stdsp24_box_midi(ice, box, - (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) ? 1 : 0); + if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) + snd_ice1712_stdsp24_box_midi(ice, box, 1); } return 0; @@ -322,6 +335,8 @@ struct snd_ice1712_card_info snd_ice1712_hoontech_cards[] __devinitdata = { .name = "Hoontech SoundTrack Audio DSP24", .model = "dsp24", .chip_init = snd_ice1712_hoontech_init, + .mpu401_1_name = "MIDI-1 Hoontech/STA DSP24", + .mpu401_2_name = "MIDI-2 Hoontech/STA DSP24", }, { .subvendor = ICE1712_SUBDEVICE_STDSP24_VALUE, /* a dummy id */ From b76c850fbc280d6c0ff786653915f3a9700b5912 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 6 Feb 2008 14:49:44 +0100 Subject: [PATCH 013/250] [ALSA] hda: STAC927x power down inactive DACs On several laptops that have STAC9228 codecs have unused DACs, this powers them down to a D3 state. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f693011d25a0..7f506ef0accb 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -136,6 +136,7 @@ struct sigmatel_spec { /* power management */ unsigned int num_pwrs; hda_nid_t *pwr_nids; + hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; @@ -291,6 +292,10 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dac_nids[6] = { + 0x02, 0x03, 0x04, 0x05, 0x06, 0 +}; + static hda_nid_t stac927x_dmux_nids[1] = { 0x1b, }; @@ -2877,6 +2882,18 @@ static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) return 0; /* nid is not a HP-Out */ }; +static void stac92xx_power_down(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + /* power down inactive DACs */ + hda_nid_t *dac; + for (dac = spec->dac_list; *dac; dac++) + if (!is_in_dac_nids(spec, *dac)) + snd_hda_codec_write_cache(codec, *dac, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -2929,7 +2946,8 @@ static int stac92xx_init(struct hda_codec *codec) enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); } - + if (spec->dac_list) + stac92xx_power_down(codec); if (cfg->dig_out_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, AC_PINCTL_OUT_EN); @@ -3102,6 +3120,9 @@ static int stac92xx_resume(struct hda_codec *codec) spec->gpio_dir, spec->gpio_data); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + /* power down inactive DACs */ + if (spec->dac_list) + stac92xx_power_down(codec); /* invoke unsolicited event to reset the HP state */ if (spec->hp_detect) codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); @@ -3589,6 +3610,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->dac_list = stac927x_dac_nids; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config) { From 4ce107b990d994a0fccea9b1e885b08a0daea495 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Feb 2008 14:50:19 +0100 Subject: [PATCH 014/250] [ALSA] hda-intel - Use SG buffer Use SG buffers for the HD-audio instead of linear buffers. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 114 ++++++++++++++++++++++++-------------- 1 file changed, 71 insertions(+), 43 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 18475de074b2..b38a5a70ff08 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -206,8 +206,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define MAX_AZX_DEV 16 /* max number of fragments - we may use more if allocating more pages for BDL */ -#define BDL_SIZE PAGE_ALIGN(8192) -#define AZX_MAX_FRAG (BDL_SIZE / (MAX_AZX_DEV * 16)) +#define BDL_SIZE 4096 +#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) +#define AZX_MAX_FRAG 32 /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) /* max number of PCM devics per card */ @@ -282,12 +283,10 @@ enum { */ struct azx_dev { - u32 *bdl; /* virtual address of the BDL */ - dma_addr_t bdl_addr; /* physical address of the BDL */ + struct snd_dma_buffer bdl; /* BDL buffer */ u32 *posbuf; /* position buffer pointer */ unsigned int bufsize; /* size of the play buffer in bytes */ - unsigned int fragsize; /* size of each period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ @@ -358,8 +357,7 @@ struct azx { struct azx_rb corb; struct azx_rb rirb; - /* BDL, CORB/RIRB and position buffers */ - struct snd_dma_buffer bdl; + /* CORB/RIRB and position buffers */ struct snd_dma_buffer rb; struct snd_dma_buffer posbuf; @@ -962,30 +960,57 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* * set up BDL entries */ -static void azx_setup_periods(struct azx_dev *azx_dev) +static int azx_setup_periods(struct snd_pcm_substream *substream, + struct azx_dev *azx_dev) { - u32 *bdl = azx_dev->bdl; - dma_addr_t dma_addr = azx_dev->substream->runtime->dma_addr; - int idx; + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + u32 *bdl; + int i, ofs, periods, period_bytes; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); + period_bytes = snd_pcm_lib_period_bytes(substream); + periods = azx_dev->bufsize / period_bytes; + /* program the initial BDL entries */ - for (idx = 0; idx < azx_dev->frags; idx++) { - unsigned int off = idx << 2; /* 4 dword step */ - dma_addr_t addr = dma_addr + idx * azx_dev->fragsize; - /* program the address field of the BDL entry */ - bdl[off] = cpu_to_le32((u32)addr); - bdl[off+1] = cpu_to_le32(upper_32bit(addr)); - - /* program the size field of the BDL entry */ - bdl[off+2] = cpu_to_le32(azx_dev->fragsize); - - /* program the IOC to enable interrupt when buffer completes */ - bdl[off+3] = cpu_to_le32(0x01); + bdl = (u32 *)azx_dev->bdl.area; + ofs = 0; + azx_dev->frags = 0; + for (i = 0; i < periods; i++) { + int size, rest; + if (i >= AZX_MAX_BDL_ENTRIES) { + snd_printk(KERN_ERR "Too many BDL entries: " + "buffer=%d, period=%d\n", + azx_dev->bufsize, period_bytes); + /* reset */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + return -EINVAL; + } + rest = period_bytes; + do { + dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32bit(addr)); + /* program the size field of the BDL entry */ + size = PAGE_SIZE - (ofs % PAGE_SIZE); + if (rest < size) + size = rest; + bdl[2] = cpu_to_le32(size); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + rest -= size; + bdl[3] = rest ? 0 : cpu_to_le32(0x01); + bdl += 4; + azx_dev->frags++; + ofs += size; + } while (rest > 0); } + return 0; } /* @@ -1034,9 +1059,9 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) /* program the BDL address */ /* lower BDL address */ - azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl_addr); + azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr); /* upper BDL address */ - azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr)); + azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr)); /* enable the position buffer */ if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) @@ -1272,8 +1297,6 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); - azx_dev->fragsize = snd_pcm_lib_period_bytes(substream); - azx_dev->frags = azx_dev->bufsize / azx_dev->fragsize; azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, runtime->format, @@ -1288,7 +1311,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, " "format=0x%x\n", azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val); - azx_setup_periods(azx_dev); + if (azx_setup_periods(substream, azx_dev) < 0) + return -EINVAL; azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; @@ -1375,6 +1399,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, }; static void azx_pcm_free(struct snd_pcm *pcm) @@ -1417,7 +1442,7 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops); if (cpcm->stream[1].substreams) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops); - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), 1024 * 64, 1024 * 1024); chip->pcm[cpcm->device] = pcm; @@ -1507,10 +1532,7 @@ static int __devinit azx_init_stream(struct azx *chip) * and initialize */ for (i = 0; i < chip->num_streams; i++) { - unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4); struct azx_dev *azx_dev = &chip->azx_dev[i]; - azx_dev->bdl = (u32 *)(chip->bdl.area + off); - azx_dev->bdl_addr = chip->bdl.addr + off; azx_dev->posbuf = (u32 __iomem *)(chip->posbuf.area + i * 8); /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); @@ -1646,8 +1668,9 @@ static int azx_resume(struct pci_dev *pci) */ static int azx_free(struct azx *chip) { + int i; + if (chip->initialized) { - int i; for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); azx_stop_chip(chip); @@ -1662,8 +1685,11 @@ static int azx_free(struct azx *chip) if (chip->remap_addr) iounmap(chip->remap_addr); - if (chip->bdl.area) - snd_dma_free_pages(&chip->bdl); + if (chip->azx_dev) { + for (i = 0; i < chip->num_streams; i++) + if (chip->azx_dev[i].bdl.area) + snd_dma_free_pages(&chip->azx_dev[i].bdl); + } if (chip->rb.area) snd_dma_free_pages(&chip->rb); if (chip->posbuf.area) @@ -1745,7 +1771,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, struct azx **rchip) { struct azx *chip; - int err; + int i, err; unsigned short gcap; static struct snd_device_ops ops = { .dev_free = azx_dev_free, @@ -1857,13 +1883,15 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } - /* allocate memory for the BDL for each stream */ - err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - BDL_SIZE, &chip->bdl); - if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); - goto errout; + for (i = 0; i < chip->num_streams; i++) { + /* allocate memory for the BDL for each stream */ + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + BDL_SIZE, &chip->azx_dev[i].bdl); + if (err < 0) { + snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + goto errout; + } } /* allocate memory for the position buffer */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, From cf7aaca8bae3a719db47ff6eca5f6f2f42eba05a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Feb 2008 15:05:57 +0100 Subject: [PATCH 015/250] [ALSA] hda-intel - Support 64bit buffer allocation The HD-audio hardware usually supports 64bit address for DMA and other buffers. The patch enables the feature if supported. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b38a5a70ff08..ec3ddda30beb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1843,6 +1843,10 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd("chipset global capabilities = 0x%x\n", gcap); + /* allow 64bit DMA address if supported by H/W */ + if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK)) + pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK); + if (gcap) { /* read number of streams from GCAP register instead of using * hardcoded value From a60567d13c047b03167df4aed6b7a8730f267234 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 6 Feb 2008 15:48:06 +0100 Subject: [PATCH 016/250] [ALSA] Added support for Delta1010E (newer revisions of Delta1010) For more details, see ALSA bug#3327 . Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/delta.c | 9 +++++++++ sound/pci/ice1712/delta.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index efd180b40e56..c78894f110be 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -86,6 +86,7 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) unsigned char tmp; tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); switch (ice->eeprom.subvendor) { + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: tmp &= ~ICE1712_DELTA_1010LT_CS; tmp |= ICE1712_DELTA_1010LT_CCLK | ICE1712_DELTA_1010LT_CS_CS8427; @@ -109,6 +110,7 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp) { switch (ice->eeprom.subvendor) { + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: tmp &= ~ICE1712_DELTA_1010LT_CS; tmp |= ICE1712_DELTA_1010LT_CS_NONE; @@ -534,6 +536,9 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) int err; struct snd_akm4xxx *ak; + if (ice->eeprom.subvendor && ice->eeprom.gpiodir == 0x7b) + ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA1010E; + /* determine I2C, DACs and ADCs */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_AUDIOPHILE: @@ -550,6 +555,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) ice->num_total_adcs = ice->omni ? 8 : 4; break; case ICE1712_SUBDEVICE_DELTA1010: + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_MEDIASTATION: ice->num_total_dacs = 8; @@ -568,6 +574,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_AUDIOPHILE: case ICE1712_SUBDEVICE_DELTA410: + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_VX442: if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) { @@ -601,6 +608,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) /* no analog? */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_DELTA1010: + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTADIO2496: case ICE1712_SUBDEVICE_MEDIASTATION: return 0; @@ -674,6 +682,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; break; + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_select, ice)); if (err < 0) diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index 26ea05a32f56..d07c49b4b66f 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -36,6 +36,7 @@ "{Lionstracs,Mediastation}," #define ICE1712_SUBDEVICE_DELTA1010 0x121430d6 +#define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6 #define ICE1712_SUBDEVICE_DELTADIO2496 0x121431d6 #define ICE1712_SUBDEVICE_DELTA66 0x121432d6 #define ICE1712_SUBDEVICE_DELTA44 0x121433d6 From ef2cd2ccad66b4aba518eca7514eface267ee0f3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 6 Feb 2008 20:04:49 +0100 Subject: [PATCH 017/250] [ALSA] ice1712 - added support for M-Audio Delta 66E See ALSA bug#3327 for more details. Experimental. Also fix support for M-Audio Delta 1010E - subdevice check. Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/delta.c | 15 ++++++++++++--- sound/pci/ice1712/delta.h | 1 + 2 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index c78894f110be..0ed96c178059 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -1,8 +1,8 @@ /* * ALSA driver for ICEnsemble ICE1712 (Envy24) * - * Lowlevel functions for M-Audio Delta 1010, 44, 66, Dio2496, Audiophile - * Digigram VX442 + * Lowlevel functions for M-Audio Delta 1010, 1010E, 44, 66, 66E, Dio2496, + * Audiophile, Digigram VX442 * * Copyright (c) 2000 Jaroslav Kysela * @@ -536,9 +536,14 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) int err; struct snd_akm4xxx *ak; - if (ice->eeprom.subvendor && ice->eeprom.gpiodir == 0x7b) + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 && + ice->eeprom.gpiodir == 0x7b) ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA1010E; + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA66 && + ice->eeprom.gpiodir == 0xfb) + ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA66E; + /* determine I2C, DACs and ADCs */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_AUDIOPHILE: @@ -565,6 +570,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) ice->num_total_dacs = 4; /* two AK4324 codecs */ break; case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: /* omni not suported yet */ ice->num_total_dacs = 4; ice->num_total_adcs = 4; break; @@ -577,6 +583,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) { snd_printk(KERN_ERR "unable to create I2C bus\n"); return err; @@ -635,6 +642,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice); break; case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice); break; default: @@ -725,6 +733,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA44: case ICE1712_SUBDEVICE_DELTA66: case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index d07c49b4b66f..ea7116c304c0 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -39,6 +39,7 @@ #define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6 #define ICE1712_SUBDEVICE_DELTADIO2496 0x121431d6 #define ICE1712_SUBDEVICE_DELTA66 0x121432d6 +#define ICE1712_SUBDEVICE_DELTA66E 0xff1432d6 #define ICE1712_SUBDEVICE_DELTA44 0x121433d6 #define ICE1712_SUBDEVICE_AUDIOPHILE 0x121434d6 #define ICE1712_SUBDEVICE_DELTA410 0x121438d6 From 21c7b0819f0d04788b2d3341f5062744373589a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Feb 2008 12:06:32 +0100 Subject: [PATCH 018/250] [ALSA] hda-intel - Fix a compile error with CONFIG_SND_DEBUG_DETECT=y Forgot to get rid of the obsolete fragsize field from a debug print. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec3ddda30beb..f3242e1a7315 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1308,9 +1308,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) return -EINVAL; } - snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, " - "format=0x%x\n", - azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val); + snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + azx_dev->bufsize, azx_dev->format_val); if (azx_setup_periods(substream, azx_dev) < 0) return -EINVAL; azx_setup_controller(chip, azx_dev); From cc4d13873aeacf89901861706a7a083d5a82e26a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Feb 2008 17:12:01 +0100 Subject: [PATCH 019/250] [ALSA] hda-codec - Correct HDMI transmitter names Give better names to the new HDMI transmitter chips. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_atihdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index e6fbd5d8fba4..45a2e30cbf42 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -163,7 +163,7 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, - { .id = 0x10951392, .name = "SI HDMI", .patch = patch_atihdmi }, - { .id = 0x17e80047, .name = "Unknown HDMI", .patch = patch_atihdmi }, + { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi }, + { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; From aa27a44395c3d35bc16e52f1e709e0fb2a3709e4 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Fri, 8 Feb 2008 12:44:17 +0100 Subject: [PATCH 020/250] [ALSA] hda-codec - remove duplicate controls in alc268 test mixer I've just noticed that there are a handful of duplicate controls in the ALC268 test model mixer. This patch (against alsa-driver 1.0.16) removes them. Signed-off-by: Jonathan Woithe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 85ea3f82de19..e4a7c50e9f73 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9840,11 +9840,6 @@ static struct hda_input_mux alc268_capture_source = { #ifdef CONFIG_SND_DEBUG static struct snd_kcontrol_new alc268_test_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - /* Volume widgets */ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT), From fdafad6fc24a11070bcd0885100be7143cc038f8 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Mon, 11 Feb 2008 14:48:06 +0100 Subject: [PATCH 021/250] [ALSA] AK4114 - listing regs in proc A simple patch for listing AK4114 regs in proc. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4114.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 15061bd72776..9a90e830c423 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -27,6 +27,7 @@ #include #include #include +#include MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei"); @@ -446,6 +447,26 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = { } }; + +static void snd_ak4114_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct ak4114 *ak4114 = entry->private_data; + int reg, val; + /* all ak4114 registers 0x00 - 0x1f */ + for (reg = 0; reg < 0x20; reg++) { + val = reg_read(ak4114, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void snd_ak4114_proc_init(struct ak4114 *ak4114) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ak4114->card, "ak4114", &entry)) + snd_info_set_text_ops(entry, ak4114, snd_ak4114_proc_regs_read); +} + int snd_ak4114_build(struct ak4114 *ak4114, struct snd_pcm_substream *ply_substream, struct snd_pcm_substream *cap_substream) @@ -478,6 +499,7 @@ int snd_ak4114_build(struct ak4114 *ak4114, return err; ak4114->kctls[idx] = kctl; } + snd_ak4114_proc_init(ak4114); /* trigger workq */ schedule_delayed_work(&ak4114->work, HZ / 10); return 0; From 937b416027d8f79d7b37bb63b6585ea8fdf125de Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Feb 2008 14:52:36 +0100 Subject: [PATCH 022/250] [ALSA] hda-codec - Fix ALC880 F1734 model Fixed some issues with ALC880 F1734 model - fix capture via mic - enable volume-wheel control Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e4a7c50e9f73..3a735870a071 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1320,11 +1320,19 @@ static struct snd_kcontrol_new alc880_f1734_mixer[] = { HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; +static struct hda_input_mux alc880_f1734_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "CD", 0x4 }, + }, +}; + /* * ALC880 ASUS model @@ -1936,6 +1944,9 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = { {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT}, + { } }; @@ -3059,7 +3070,9 @@ static struct alc_config_preset alc880_presets[] = { .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, + .input_mux = &alc880_f1734_capture_source, + .unsol_event = alc880_uniwill_p53_unsol_event, + .init_hook = alc880_uniwill_p53_hp_automute, }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, From f0824812af1bf4f7d27e054a2ca2686385d770bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Feb 2008 15:54:34 +0100 Subject: [PATCH 023/250] [ALSA] hda-codec - Fix automute of AD1981HD hp model Reprogram the speaker-pin setting at each HP pin plug to make sure the spekaer auto-muting on AD1981HD hp model. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7286ab86ecc4..9d0d2a1bbd64 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1366,7 +1366,10 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, if (! ad198x_eapd_put(kcontrol, ucontrol)) return 0; - + /* change speaker pin appropriately */ + snd_hda_codec_write(codec, 0x05, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->cur_eapd ? PIN_OUT : 0); /* toggle HP mute appropriately */ snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, HDA_AMP_MUTE, From e1406348129dc2db60ccad079c3d014200590557 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Feb 2008 18:32:32 +0100 Subject: [PATCH 024/250] [ALSA] hda-codec - Clean up capture source selection of Realtek codecs Clean up the codes of the capture source selection for Realtek codecs. Now using common helper functions with the new capsrc_nids field. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 216 +++++++--------------------------- 1 file changed, 42 insertions(+), 174 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a735870a071..f1fa1d249369 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -238,6 +238,7 @@ struct alc_spec { /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ /* capture source */ @@ -291,6 +292,7 @@ struct alc_config_preset { hda_nid_t hp_nid; /* optional */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; @@ -337,9 +339,10 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; + hda_nid_t nid = spec->capsrc_nids ? + spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol, - spec->adc_nids[adc_idx], - &spec->cur_mux[adc_idx]); + nid, &spec->cur_mux[adc_idx]); } @@ -708,6 +711,7 @@ static void setup_preset(struct alc_spec *spec, spec->num_adc_nids = preset->num_adc_nids; spec->adc_nids = preset->adc_nids; + spec->capsrc_nids = preset->capsrc_nids; spec->dig_in_nid = preset->dig_in_nid; spec->unsol_event = preset->unsol_event; @@ -5219,6 +5223,9 @@ static hda_nid_t alc882_dac_nids[4] = { #define alc882_adc_nids alc880_adc_nids #define alc882_adc_nids_alt alc880_adc_nids_alt +static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; +static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -5241,15 +5248,10 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; const struct hda_input_mux *imux = spec->input_mux; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 }; - hda_nid_t nid; + hda_nid_t nid = spec->capsrc_nids[adc_idx]; unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; - if (spec->num_adc_nids < 3) - nid = capture_mixers[adc_idx + 1]; - else - nid = capture_mixers[adc_idx]; idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; @@ -6126,6 +6128,7 @@ static struct alc_config_preset alc882_presets[] = { .dig_out_nid = ALC882_DIGOUT_NID, .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), .channel_mode = alc882_3ST_6ch_modes, .need_dac_fix = 1, @@ -6142,6 +6145,7 @@ static struct alc_config_preset alc882_presets[] = { .dig_out_nid = ALC882_DIGOUT_NID, .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), .channel_mode = alc882_3ST_6ch_modes, .need_dac_fix = 1, @@ -6387,12 +6391,14 @@ static int patch_alc882(struct hda_codec *codec) if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc882_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt); + spec->capsrc_nids = alc882_capsrc_nids_alt; spec->mixers[spec->num_mixers] = alc882_capture_alt_mixer; spec->num_mixers++; } else { spec->adc_nids = alc882_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); + spec->capsrc_nids = alc882_capsrc_nids; spec->mixers[spec->num_mixers] = alc882_capture_mixer; spec->num_mixers++; } @@ -6435,6 +6441,8 @@ static hda_nid_t alc883_adc_nids[2] = { 0x08, 0x09, }; +static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -6468,33 +6476,8 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { #define alc883_mux_enum_info alc_mux_enum_info #define alc883_mux_enum_get alc_mux_enum_get - -static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[2] = { 0x23, 0x22 }; - hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} +/* ALC883 has the ALC882-type input selection */ +#define alc883_mux_enum_put alc882_mux_enum_put /* * 2ch mode @@ -7667,8 +7650,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, @@ -7680,8 +7661,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, @@ -7693,8 +7672,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, @@ -7706,8 +7683,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, @@ -7719,8 +7694,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, @@ -7734,8 +7707,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7752,8 +7723,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7764,8 +7733,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7779,8 +7746,6 @@ static struct alc_config_preset alc883_presets[] = { alc883_medion_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -7791,8 +7756,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7804,8 +7767,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7815,8 +7776,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_lenovo_101e_capture_source, @@ -7828,8 +7787,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, @@ -7843,8 +7800,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, @@ -7858,8 +7813,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7872,8 +7825,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, @@ -7884,8 +7835,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes), .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, @@ -7897,8 +7846,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, @@ -7911,8 +7858,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -8072,10 +8017,9 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->adc_nids && spec->input_mux) { - spec->adc_nids = alc883_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - } + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + spec->capsrc_nids = alc883_capsrc_nids; spec->vmaster_nid = 0x0c; @@ -9532,6 +9476,8 @@ static hda_nid_t alc268_adc_nids_alt[1] = { 0x08 }; +static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 }; + static struct snd_kcontrol_new alc268_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), @@ -9787,21 +9733,7 @@ static struct hda_verb alc268_volume_init_verbs[] = { #define alc268_mux_enum_info alc_mux_enum_info #define alc268_mux_enum_get alc_mux_enum_get - -static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; - hda_nid_t nid = capture_mixers[adc_idx]; - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - nid, - &spec->cur_mux[adc_idx]); -} +#define alc268_mux_enum_put alc_mux_enum_put static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), @@ -10145,6 +10077,7 @@ static struct alc_config_preset alc268_presets[] = { .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10159,6 +10092,7 @@ static struct alc_config_preset alc268_presets[] = { .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, @@ -10174,6 +10108,7 @@ static struct alc_config_preset alc268_presets[] = { .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, @@ -10202,6 +10137,7 @@ static struct alc_config_preset alc268_presets[] = { .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10219,6 +10155,7 @@ static struct alc_config_preset alc268_presets[] = { .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10294,6 +10231,7 @@ static int patch_alc268(struct hda_codec *codec) alc268_capture_mixer; spec->num_mixers++; } + spec->capsrc_nids = alc268_capsrc_nids; } spec->vmaster_nid = 0x02; @@ -11850,6 +11788,8 @@ static hda_nid_t alc861vd_adc_nids[1] = { 0x09, }; +static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ static struct hda_input_mux alc861vd_capture_source = { @@ -11881,33 +11821,8 @@ static struct hda_input_mux alc861vd_hp_capture_source = { #define alc861vd_mux_enum_info alc_mux_enum_info #define alc861vd_mux_enum_get alc_mux_enum_get - -static int alc861vd_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[1] = { 0x22 }; - hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} +/* ALC861VD has the ALC882-type input selection (but has only one ADC) */ +#define alc861vd_mux_enum_put alc882_mux_enum_put /* * 2ch mode @@ -12390,8 +12305,6 @@ static struct alc_config_preset alc861vd_presets[] = { alc861vd_3stack_init_verbs }, .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), .dac_nids = alc660vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, @@ -12403,8 +12316,6 @@ static struct alc_config_preset alc861vd_presets[] = { .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), .dac_nids = alc660vd_dac_nids, .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, @@ -12449,8 +12360,6 @@ static struct alc_config_preset alc861vd_presets[] = { alc861vd_lenovo_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), .dac_nids = alc660vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, @@ -12462,8 +12371,6 @@ static struct alc_config_preset alc861vd_presets[] = { .init_verbs = { alc861vd_dallas_verbs }, .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), .dac_nids = alc861vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_dallas_capture_source, @@ -12475,9 +12382,7 @@ static struct alc_config_preset alc861vd_presets[] = { .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), .dac_nids = alc861vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), .dig_out_nid = ALC861VD_DIGOUT_NID, - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_hp_capture_source, @@ -12779,6 +12684,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); + spec->capsrc_nids = alc861vd_capsrc_nids; spec->mixers[spec->num_mixers] = alc861vd_capture_mixer; spec->num_mixers++; @@ -12820,9 +12726,11 @@ static hda_nid_t alc662_adc_nids[1] = { /* ADC1-2 */ 0x09, }; + +static hda_nid_t alc662_capsrc_nids[1] = { 0x23 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ - static struct hda_input_mux alc662_capture_source = { .num_items = 4, .items = { @@ -12851,33 +12759,8 @@ static struct hda_input_mux alc662_eeepc_capture_source = { #define alc662_mux_enum_info alc_mux_enum_info #define alc662_mux_enum_get alc_mux_enum_get +#define alc662_mux_enum_put alc882_mux_enum_put -static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[2] = { 0x23, 0x22 }; - hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} /* * 2ch mode */ @@ -13354,8 +13237,6 @@ static struct alc_config_preset alc662_presets[] = { .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .dig_in_nid = ALC662_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, @@ -13368,8 +13249,6 @@ static struct alc_config_preset alc662_presets[] = { .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .dig_in_nid = ALC662_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, @@ -13382,8 +13261,6 @@ static struct alc_config_preset alc662_presets[] = { .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .need_dac_fix = 1, @@ -13396,8 +13273,6 @@ static struct alc_config_preset alc662_presets[] = { .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .dig_in_nid = ALC662_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes), .channel_mode = alc662_5stack_modes, @@ -13408,8 +13283,6 @@ static struct alc_config_preset alc662_presets[] = { .init_verbs = { alc662_init_verbs, alc662_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, @@ -13422,8 +13295,6 @@ static struct alc_config_preset alc662_presets[] = { alc662_eeepc_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, .input_mux = &alc662_eeepc_capture_source, @@ -13437,8 +13308,6 @@ static struct alc_config_preset alc662_presets[] = { alc662_eeepc_ep20_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, @@ -13750,10 +13619,9 @@ static int patch_alc662(struct hda_codec *codec) spec->stream_digital_playback = &alc662_pcm_digital_playback; spec->stream_digital_capture = &alc662_pcm_digital_capture; - if (!spec->adc_nids && spec->input_mux) { - spec->adc_nids = alc662_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); - } + spec->adc_nids = alc662_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); + spec->capsrc_nids = alc662_capsrc_nids; spec->vmaster_nid = 0x02; From 5d5d5f43f1b835c375de9bd270cce030d16e2871 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Feb 2008 12:11:36 +0100 Subject: [PATCH 025/250] [ALSA] hda-codec - Implement auto-mic jack sensing on Samsung laptops Implemented the auto-mic jack sensing for Samsung laptops with AD1986A codec chip (model=laptop-eapd). The hardware uses pin 0x1d and 0x1f for the internal and external mics, respectively. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 53 +++++++++++++++++++++++++++++++++--- 1 file changed, 49 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9d0d2a1bbd64..266c35e32b64 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -612,13 +612,19 @@ static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { }, }; +static struct hda_input_mux ad1986a_automic_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Mix", 0x5 }, + }, +}; + static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), @@ -642,6 +648,33 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { } /* end */ }; +/* re-connect the mic boost input according to the jack sensing */ +static void ad1986a_automic(struct hda_codec *codec) +{ + unsigned int present; + present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0); + /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */ + snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL, + (present & AC_PINSENSE_PRESENCE) ? 0 : 2); +} + +#define AD1986A_MIC_EVENT 0x36 + +static void ad1986a_automic_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1986A_MIC_EVENT) + return; + ad1986a_automic(codec); +} + +static int ad1986a_automic_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1986a_automic(codec); + return 0; +} + /* laptop-automute - 2ch only */ static void ad1986a_update_hp(struct hda_codec *codec) @@ -845,6 +878,15 @@ static struct hda_verb ad1986a_eapd_init_verbs[] = { {} }; +static struct hda_verb ad1986a_automic_verbs[] = { + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/ + {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT}, + {} +}; + /* Ultra initialization */ static struct hda_verb ad1986a_ultra_init[] = { /* eapd initialization */ @@ -987,14 +1029,17 @@ static int patch_ad1986a(struct hda_codec *codec) break; case AD1986A_LAPTOP_EAPD: spec->mixers[0] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 2; + spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->init_verbs[2] = ad1986a_automic_verbs; spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = ad1986a_laptop_dac_nids; if (!is_jack_available(codec, 0x25)) spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_laptop_eapd_capture_source; + spec->input_mux = &ad1986a_automic_capture_source; + codec->patch_ops.unsol_event = ad1986a_automic_unsol_event; + codec->patch_ops.init = ad1986a_automic_init; break; case AD1986A_LAPTOP_AUTOMUTE: spec->mixers[0] = ad1986a_laptop_automute_mixers; From 5832fcf8b55cfdbd7d8511f747d15fd20ed4703d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Feb 2008 18:30:12 +0100 Subject: [PATCH 026/250] [ALSA] hda-codec - More fix-up for auto-configuration In some cases, the BIOS sets up only the HP pins with different assoc and sequence numbers, e.g. on FSC Esprimo with ALC262. This patch adds a fix-up for such a case. When multiple HPs are defined and no line-outs is found, the configurator tries to re-assign some pins from HP list to line-out, judging from the sequence number. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ab3bb7997cd2..af2c8943b303 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2821,6 +2821,30 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } } + /* FIX-UP: + * If no line-out is defined but multiple HPs are found, + * some of them might be the real line-outs. + */ + if (!cfg->line_outs && cfg->hp_outs > 1) { + int i = 0; + while (i < cfg->hp_outs) { + /* The real HPs should have the sequence 0x0f */ + if ((sequences_hp[i] & 0x0f) == 0x0f) { + i++; + continue; + } + /* Move it to the line-out table */ + cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i]; + sequences_line_out[cfg->line_outs] = sequences_hp[i]; + cfg->line_outs++; + cfg->hp_outs--; + memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, + sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); + memmove(sequences_hp + i - 1, sequences_hp + i, + sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); + } + } + /* sort by sequence */ sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out, cfg->line_outs); From f6c7e5461e9046445d50c5c7a9a4587824239623 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Feb 2008 18:32:23 +0100 Subject: [PATCH 027/250] [ALSA] hda-codec - Fix auto-configuration of Realtek codecs This patch fixes some bugs in the auto-configurator of Realtek codecs: - add missing pin set-up for speaker pins - fix the speaker auto-mute function not to conflict with the existing "Speaker" mixer switch Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 111 +++++++++++++++++++--------------- 1 file changed, 63 insertions(+), 48 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1fa1d249369..d41eafacd86d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -746,7 +746,6 @@ static struct hda_verb alc_gpio3_init_verbs[] = { static void alc_sku_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int mute; unsigned int present; unsigned int hp_nid = spec->autocfg.hp_pins[0]; unsigned int sp_nid = spec->autocfg.speaker_pins[0]; @@ -756,16 +755,8 @@ static void alc_sku_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_GET_PIN_SENSE, 0); spec->jack_present = (present & 0x80000000) != 0; - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, hp_nid, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } + snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->jack_present ? 0 : PIN_OUT); } /* unsolicited event for HP jack sensing */ @@ -3486,15 +3477,20 @@ static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, return 0; } +static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, + unsigned int pin_type) +{ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_type); + /* unmute pin */ + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, 0xff, 0x00); +} + static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ if (alc880_is_multi_pin(nid)) { struct alc_spec *spec = codec->spec; @@ -3616,9 +3612,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc880_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -4814,11 +4813,7 @@ static void alc260_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int sel_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ if (nid >= 0x12) { int idx = nid - 0x12; @@ -4965,8 +4960,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc260_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -6201,15 +6199,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, struct alc_spec *spec = codec->spec; int idx; + alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; else idx = spec->multiout.dac_nids[dac_idx] - 2; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -6238,6 +6232,9 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ /* use dac 0 */ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc882_is_input_pin(nid) alc880_is_input_pin(nid) @@ -6313,9 +6310,12 @@ static int alc882_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc882_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc882_auto_init_multi_out(codec); alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc882(struct hda_codec *codec) @@ -7878,15 +7878,11 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec, struct alc_spec *spec = codec->spec; int idx; + alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; else idx = spec->multiout.dac_nids[dac_idx] - 2; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -7915,6 +7911,9 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ /* use dac 0 */ alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc883_is_input_pin(nid) alc880_is_input_pin(nid) @@ -7966,9 +7965,12 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc883_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc883_auto_init_multi_out(codec); alc883_auto_init_hp_out(codec); alc883_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc883(struct hda_codec *codec) @@ -9144,9 +9146,12 @@ static int alc262_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static void alc262_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc262_auto_init_multi_out(codec); alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -10033,10 +10038,13 @@ static int alc268_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static void alc268_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc268_auto_init_multi_out(codec); alc268_auto_init_hp_out(codec); alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -10505,9 +10513,12 @@ static int alc269_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static void alc269_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc269_auto_init_multi_out(codec); alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -11429,13 +11440,7 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); - + alc_set_pin_output(codec, nid, pin_type); } static void alc861_auto_init_multi_out(struct hda_codec *codec) @@ -11462,6 +11467,9 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } static void alc861_auto_init_analog_input(struct hda_codec *codec) @@ -11534,9 +11542,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc861_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -12397,11 +12408,7 @@ static struct alc_config_preset alc861vd_presets[] = { static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); } static void alc861vd_auto_init_multi_out(struct hda_codec *codec) @@ -12428,6 +12435,9 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front and use dac 0 */ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc861vd_is_input_pin(nid) alc880_is_input_pin(nid) @@ -12631,9 +12641,12 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc861vd_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc861vd_auto_init_multi_out(codec); alc861vd_auto_init_hp_out(codec); alc861vd_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc861vd(struct hda_codec *codec) @@ -13453,11 +13466,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ if (alc880_is_multi_pin(nid)) { struct alc_spec *spec = codec->spec; @@ -13492,6 +13501,9 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ /* use dac 0 */ alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc662_is_input_pin(nid) alc880_is_input_pin(nid) @@ -13569,9 +13581,12 @@ static int alc662_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc662_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc662_auto_init_multi_out(codec); alc662_auto_init_hp_out(codec); alc662_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc662(struct hda_codec *codec) From 9a08160bdbe3148a405f72798f76e2a5d30bd243 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Feb 2008 18:37:26 +0100 Subject: [PATCH 028/250] [ALSA] hda-codec - Add "IEC958 Default PCM" switch Added a new mixer switch to enable/disable the sharing of the default PCM stream with analog and SPDIF outputs. When "IEC958 Default PCM" switch is on, the PCM stream is routed both to analog and SPDIF outputs. This is the behavior in the earlier version. Turning this switch off has a merit for some codecs, though. Some codec chips don't support 24bit formats for SPDIF but only for analog outputs. In this case, you can use 24bit format by disabling this switch. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 71 ++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_local.h | 13 ++++++- sound/pci/hda/patch_analog.c | 8 +++- sound/pci/hda/patch_cmedia.c | 8 +++- sound/pci/hda/patch_conexant.c | 8 +++- sound/pci/hda/patch_realtek.c | 8 +++- sound/pci/hda/patch_sigmatel.c | 8 +++- sound/pci/hda/patch_via.c | 8 +++- 8 files changed, 122 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index af2c8943b303..853e5c786c37 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1532,6 +1532,43 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) return 0; } +/* + * SPDIF sharing with analog output + */ +static int spdif_share_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = mout->share_spdif; + return 0; +} + +static int spdif_share_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol); + mout->share_spdif = !!ucontrol->value.integer.value[0]; + return 0; +} + +static struct snd_kcontrol_new spdif_share_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Default PCM Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = spdif_share_sw_get, + .put = spdif_share_sw_put, +}; + +int snd_hda_create_spdif_share_sw(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + if (!mout->dig_out_nid) + return 0; + /* ATTENTION: here mout is passed as private_data, instead of codec */ + return snd_ctl_add(codec->bus->card, + snd_ctl_new1(&spdif_share_sw, mout)); +} + /* * SPDIF input */ @@ -2557,9 +2594,36 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, */ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + struct hda_pcm_stream *hinfo) { - substream->runtime->hw.channels_max = mout->max_channels; + struct snd_pcm_runtime *runtime = substream->runtime; + runtime->hw.channels_max = mout->max_channels; + if (mout->dig_out_nid) { + if (!mout->analog_rates) { + mout->analog_rates = hinfo->rates; + mout->analog_formats = hinfo->formats; + mout->analog_maxbps = hinfo->maxbps; + } else { + runtime->hw.rates = mout->analog_rates; + runtime->hw.formats = mout->analog_formats; + hinfo->maxbps = mout->analog_maxbps; + } + if (!mout->spdif_rates) { + snd_hda_query_supported_pcm(codec, mout->dig_out_nid, + &mout->spdif_rates, + &mout->spdif_formats, + &mout->spdif_maxbps); + } + mutex_lock(&codec->spdif_mutex); + if (mout->share_spdif) { + runtime->hw.rates &= mout->spdif_rates; + runtime->hw.formats &= mout->spdif_formats; + if (mout->spdif_maxbps < hinfo->maxbps) + hinfo->maxbps = mout->spdif_maxbps; + } + } + mutex_unlock(&codec->spdif_mutex); return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } @@ -2579,7 +2643,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, int i; mutex_lock(&codec->spdif_mutex); - if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { + if (mout->dig_out_nid && mout->share_spdif && + mout->dig_out_used != HDA_DIG_EXCLUSIVE) { if (chs == 2 && snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ad0014ab71f9..ce2ad42a8a8a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -228,8 +228,18 @@ struct hda_multi_out { int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ + int share_spdif; /* share SPDIF pin */ + /* PCM information for both analog and SPDIF DACs */ + unsigned int analog_rates; + unsigned int analog_maxbps; + u64 analog_formats; + unsigned int spdif_rates; + unsigned int spdif_maxbps; + u64 spdif_formats; }; +int snd_hda_create_spdif_share_sw(struct hda_codec *codec, + struct hda_multi_out *mout); int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, @@ -241,7 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, - struct snd_pcm_substream *substream); + struct snd_pcm_substream *substream, + struct hda_pcm_stream *hinfo); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 266c35e32b64..1f2102860fe8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -171,6 +171,11 @@ static int ad198x_build_controls(struct hda_codec *codec) err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -217,7 +222,8 @@ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 99ce74b4e9fc..9794d4166ae4 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -329,6 +329,11 @@ static int cmi9880_build_controls(struct hda_codec *codec) err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -432,7 +437,8 @@ static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bb915ede0ceb..2bb9a58db9fa 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -98,7 +98,8 @@ static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -372,6 +373,11 @@ static int conexant_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d41eafacd86d..6c605813fc6e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1520,6 +1520,11 @@ static int alc_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -2325,7 +2330,8 @@ static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f506ef0accb..7901e76f2690 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -916,6 +916,11 @@ static int stac92xx_build_controls(struct hda_codec *codec) err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -1748,7 +1753,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d9a5c6a2dd9f..3515a3fb5d9d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -357,7 +357,8 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -493,6 +494,11 @@ static int via_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); From c8cd1281171602033861d0888273e0512f9b165c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Feb 2008 16:59:29 +0100 Subject: [PATCH 029/250] [ALSA] hda-codec - Add more names to vendor list Added more known names to the vendor id list. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 853e5c786c37..8ab88d9ba3b5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -51,13 +51,18 @@ struct hda_vendor_id { /* codec vendor labels */ static struct hda_vendor_id hda_vendor_ids[] = { - { 0x10ec, "Realtek" }, + { 0x1002, "ATI" }, { 0x1057, "Motorola" }, + { 0x1095, "Silicon Image" }, + { 0x10ec, "Realtek" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, + { 0x11c1, "LSI" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, { 0x14f1, "Conexant" }, + { 0x17e8, "Chrontel" }, + { 0x1854, "LG" }, { 0x434d, "C-Media" }, { 0x8384, "SigmaTel" }, {} /* terminator */ From d260cdf65657382c4cde366a1c7d4ddce669a427 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Feb 2008 17:19:35 +0100 Subject: [PATCH 030/250] [ALSA] hda-codec - Fix breakage of resume in auto-config of realtek codecs The last patch for fixing the auto-config pin setting breaks the resume due to a wrong use of snd_hda_codec_amp_stereo(). The code in the init hook shouldn't touch the amp cache. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6c605813fc6e..355d88d07ea2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3489,7 +3489,8 @@ static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); /* unmute pin */ - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, 0xff, 0x00); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); } static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, From 814b1a5ce6b35bafd583ae813da97e8254bb1930 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Thu, 14 Feb 2008 12:55:13 +0100 Subject: [PATCH 031/250] [ALSA] HDA-Intel - Patch to support RV7xx HDMI Audio This patch is to add R7xx HDMI audio support. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f3242e1a7315..66dceffa1217 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2051,6 +2051,9 @@ static struct pci_device_id azx_ids[] = { { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */ { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */ { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */ + { 0x1002, 0xaa38, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV730 HDMI */ + { 0x1002, 0xaa40, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV710 HDMI */ + { 0x1002, 0xaa48, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV740 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ From 88c71a9974693f2b2824b09340269511dd7cbe18 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Feb 2008 17:27:17 +0100 Subject: [PATCH 032/250] [ALSA] hda-codec - Fix missing capsrc_nids for ALC262 ALC262 must have capsrc_nids defined as well as in ALC882. Also, add a NULL check in alc882_mux_enum_put to avoid Oops. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 355d88d07ea2..eea18b3336d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5253,7 +5253,8 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; const struct hda_input_mux *imux = spec->input_mux; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - hda_nid_t nid = spec->capsrc_nids[adc_idx]; + hda_nid_t nid = spec->capsrc_nids ? + spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -8053,6 +8054,8 @@ static int patch_alc883(struct hda_codec *codec) #define alc262_dac_nids alc260_dac_nids #define alc262_adc_nids alc882_adc_nids #define alc262_adc_nids_alt alc882_adc_nids_alt +#define alc262_capsrc_nids alc882_capsrc_nids +#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt #define alc262_modes alc260_modes #define alc262_capture_source alc882_capture_source @@ -9443,12 +9446,14 @@ static int patch_alc262(struct hda_codec *codec) if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc262_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt); + spec->capsrc_nids = alc262_capsrc_nids_alt; spec->mixers[spec->num_mixers] = alc262_capture_alt_mixer; spec->num_mixers++; } else { spec->adc_nids = alc262_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids); + spec->capsrc_nids = alc262_capsrc_nids; spec->mixers[spec->num_mixers] = alc262_capture_mixer; spec->num_mixers++; } From 83ac08c0846bc6106d6c7fbb342eab02b32dd399 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 15 Feb 2008 16:43:11 +0100 Subject: [PATCH 033/250] [ALSA] ASoC: WM9713 driver This patch adds an ASoC driver for the WM9713 AC97 codec. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9713.c | 1289 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9713.h | 53 ++ 4 files changed, 1348 insertions(+) create mode 100644 sound/soc/codecs/wm9713.c create mode 100644 sound/soc/codecs/wm9713.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 898a7d363284..3903ab7dfa4a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,10 @@ config SND_SOC_WM9712 tristate depends on SND_SOC +config SND_SOC_WM9713 + tristate + depends on SND_SOC + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c6e5338c2666..4e1314c9d3ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o +snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -11,5 +12,6 @@ obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o +obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c new file mode 100644 index 000000000000..c3d0afdc0997 --- /dev/null +++ b/sound/soc/codecs/wm9713.c @@ -0,0 +1,1289 @@ +/* + * wm9713.c -- ALSA Soc WM9713 codec support + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 4th Feb 2006 Initial version. + * + * Features:- + * + * o Support for AC97 Codec, Voice DAC and Aux DAC + * o Support for DAPM + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm9713.h" + +#define WM9713_VERSION "0.15" + +struct wm9713_priv { + u32 pll_in; /* PLL input frequency */ + u32 pll_out; /* PLL output frequency */ +}; + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg); +static int ac97_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val); + +/* + * WM9713 register cache + * Reg 0x3c bit 15 is used by touch driver. + */ +static const u16 wm9713_reg[] = { + 0x6174, 0x8080, 0x8080, 0x8080, + 0xc880, 0xe808, 0xe808, 0x0808, + 0x00da, 0x8000, 0xd600, 0xaaa0, + 0xaaa0, 0xaaa0, 0x0000, 0x0000, + 0x0f0f, 0x0040, 0x0000, 0x7f00, + 0x0405, 0x0410, 0xbb80, 0xbb80, + 0x0000, 0xbb80, 0x0000, 0x4523, + 0x0000, 0x2000, 0x7eff, 0xffff, + 0x0000, 0x0000, 0x0080, 0x0000, + 0x0000, 0x0000, 0xfffe, 0xffff, + 0x0000, 0x0000, 0x0000, 0xfffe, + 0x4000, 0x0000, 0x0000, 0x0000, + 0xb032, 0x3e00, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0006, + 0x0001, 0x0000, 0x574d, 0x4c13, + 0x0000, 0x0000, 0x0000 +}; + +/* virtual HP mixers regs */ +#define HPL_MIXER 0x80 +#define HPR_MIXER 0x82 +#define MICB_MUX 0x82 + +static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; +static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; +static const char *wm9713_rec_src[] = + {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker", + "Mono Out", "Zh"}; +static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; +static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", + "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_spk_pga[] = + {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", + "Speaker Vmid", "Inv Vmid"}; +static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone", + "Headphone Vmid"}; +static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"}; +static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"}; +static const char *wm9713_dac_inv[] = + {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone", + "Headphone Mono", "NC", "Vmid"}; +static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"}; +static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"}; +static const char *wm9713_micb_select[] = {"MPB", "MPA"}; + +static const struct soc_enum wm9713_enum[] = { +SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/ +SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ +SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ +SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ +SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ +}; + +static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { +SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), +SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1), +SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), +SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), +SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), +SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), + +SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), +SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), + +SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), +SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), +SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0), +SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), + +SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), +SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), + +SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), +SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), +SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), +SOC_ENUM("ALC Function", wm9713_enum[6]), +SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), +SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), +SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), +SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), +SOC_ENUM("ALC NG Type", wm9713_enum[17]), +SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), + +SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0), +SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), + +SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), +SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), +SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), + +SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), +SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), +SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), + +SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), +SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), + +SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), + +SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), +SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), +SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), + +SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), +SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), +SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), + +SOC_ENUM("Bass Control", wm9713_enum[16]), +SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), +SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), +SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), +SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), +SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), + +SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), +SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), +SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), +}; + +/* add non dapm controls */ +static int wm9713_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm9713_snd_ac97_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + return 0; +} + +/* We have to create a fake left and right HP mixers because + * the codec only has a single control that is shared by both channels. + * This makes it impossible to determine the audio path using the current + * register map, thus we add a new (virtual) register to help determine the + * audio route within the device. + */ +static int mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 l, r, beep, tone, phone, rec, pcm, aux; + + l = ac97_read(w->codec, HPL_MIXER); + r = ac97_read(w->codec, HPR_MIXER); + beep = ac97_read(w->codec, AC97_PC_BEEP); + tone = ac97_read(w->codec, AC97_MASTER_TONE); + phone = ac97_read(w->codec, AC97_PHONE); + rec = ac97_read(w->codec, AC97_REC_SEL); + pcm = ac97_read(w->codec, AC97_PCM); + aux = ac97_read(w->codec, AC97_AUX); + + if (event & SND_SOC_DAPM_PRE_REG) + return 0; + if ((l & 0x1) || (r & 0x1)) + ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); + else + ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); + + if ((l & 0x2) || (r & 0x2)) + ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); + else + ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); + + if ((l & 0x4) || (r & 0x4)) + ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); + else + ac97_write(w->codec, AC97_PHONE, phone | 0x8000); + + if ((l & 0x8) || (r & 0x8)) + ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); + else + ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); + + if ((l & 0x10) || (r & 0x10)) + ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); + else + ac97_write(w->codec, AC97_PCM, pcm | 0x8000); + + if ((l & 0x20) || (r & 0x20)) + ac97_write(w->codec, AC97_AUX, aux & 0x7fff); + else + ac97_write(w->codec, AC97_AUX, aux | 0x8000); + + return 0; +} + +/* Left Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), +}; + +/* Right Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), +}; + +/* headphone capture mux */ +static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[1]); + +/* headphone mic mux */ +static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[0]); + +/* Speaker Mixer */ +static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1), +SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1), +SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1), +}; + +/* mono mic mux */ +static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[2]); + +/* mono output mux */ +static const struct snd_kcontrol_new wm9713_mono_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[7]); + +/* speaker left output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[8]); + +/* speaker right output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[9]); + +/* headphone left output mux */ +static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[10]); + +/* headphone right output mux */ +static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[11]); + +/* Out3 mux */ +static const struct snd_kcontrol_new wm9713_out3_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[12]); + +/* Out4 mux */ +static const struct snd_kcontrol_new wm9713_out4_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[13]); + +/* DAC inv mux 1 */ +static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[14]); + +/* DAC inv mux 2 */ +static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[15]); + +/* Capture source left */ +static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[3]); + +/* Capture source right */ +static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[4]); + +/* mic source */ +static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[18]); + +/* mic source B virtual control */ +static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[19]); + +static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = { +SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_rec_mux_controls), +SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_mic_mux_controls), +SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mic_mux_controls), +SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mux_controls), +SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkl_mux_controls), +SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkr_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out3_mux_controls), +SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out4_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv1_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv2_mux_controls), +SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcl_mux_controls), +SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcr_mux_controls), +SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, + &wm9713_mic_sel_mux_controls), +SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, + &wm9713_micb_sel_mux_controls), +SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, + &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, + &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, + &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, + &wm9713_speaker_mixer_controls[0], + ARRAY_SIZE(wm9713_speaker_mixer_controls)), +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1), +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1), +SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1), +SND_SOC_DAPM_OUTPUT("MONO"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKL"), +SND_SOC_DAPM_OUTPUT("SPKR"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_INPUT("LINEL"), +SND_SOC_DAPM_INPUT("LINER"), +SND_SOC_DAPM_INPUT("MONOIN"), +SND_SOC_DAPM_INPUT("PCBEEP"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2A"), +SND_SOC_DAPM_INPUT("MIC2B"), +SND_SOC_DAPM_VMID("VMID"), +}; + +static const char *audio_map[][3] = { + /* left HP mixer */ + {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, + {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Left HP Mixer", NULL, "Capture Headphone Mux"}, + + /* right HP mixer */ + {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, + {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Right HP Mixer", NULL, "Capture Headphone Mux"}, + + /* virtual mixer - mixes left & right channels for spk and mono */ + {"AC97 Mixer", NULL, "Left DAC"}, + {"AC97 Mixer", NULL, "Right DAC"}, + {"Line Mixer", NULL, "Right Line In"}, + {"Line Mixer", NULL, "Left Line In"}, + {"HP Mixer", NULL, "Left HP Mixer"}, + {"HP Mixer", NULL, "Right HP Mixer"}, + {"Capture Mixer", NULL, "Left Capture Source"}, + {"Capture Mixer", NULL, "Right Capture Source"}, + + /* speaker mixer */ + {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, + + /* mono mixer */ + {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", NULL, "Capture Mono Mux"}, + + /* DAC inv mux 1 */ + {"DAC Inv Mux 1", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"}, + + /* DAC inv mux 2 */ + {"DAC Inv Mux 2", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"}, + + /* headphone left mux */ + {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"}, + + /* headphone right mux */ + {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"}, + + /* speaker left mux */ + {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"}, + {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* speaker right mux */ + {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"}, + {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"}, + + /* mono mux */ + {"Mono Out Mux", "Mono", "Mono Mixer"}, + {"Mono Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* out 3 mux */ + {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"}, + + /* out 4 mux */ + {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Out Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Out Mux"}, + {"OUT3", NULL, "Out 3"}, + {"Out 3", NULL, "Out 3 Mux"}, + {"OUT4", NULL, "Out 4"}, + {"Out 4", NULL, "Out 4 Mux"}, + {"SPKL", NULL, "Left Speaker"}, + {"Left Speaker", NULL, "Left Speaker Out Mux"}, + {"SPKR", NULL, "Right Speaker"}, + {"Right Speaker", NULL, "Right Speaker Out Mux"}, + {"MONO", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono Out Mux"}, + + /* input pga */ + {"Left Line In", NULL, "LINEL"}, + {"Right Line In", NULL, "LINER"}, + {"Mono In", NULL, "MONOIN"}, + {"Mic A PGA", NULL, "Mic A Pre Amp"}, + {"Mic B PGA", NULL, "Mic B Pre Amp"}, + + /* left capture select */ + {"Left Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Left Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Left Capture Source", "Line", "LINEL"}, + {"Left Capture Source", "Mono In", "MONOIN"}, + {"Left Capture Source", "Headphone", "Left HP Mixer"}, + {"Left Capture Source", "Speaker", "Speaker Mixer"}, + {"Left Capture Source", "Mono Out", "Mono Mixer"}, + + /* right capture select */ + {"Right Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Right Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Right Capture Source", "Line", "LINER"}, + {"Right Capture Source", "Mono In", "MONOIN"}, + {"Right Capture Source", "Headphone", "Right HP Mixer"}, + {"Right Capture Source", "Speaker", "Speaker Mixer"}, + {"Right Capture Source", "Mono Out", "Mono Mixer"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Source"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Source"}, + + /* mic */ + {"Mic A Pre Amp", NULL, "Mic A Source"}, + {"Mic A Source", "Mic 1", "MIC1"}, + {"Mic A Source", "Mic 2 A", "MIC2A"}, + {"Mic A Source", "Mic 2 B", "Mic B Source"}, + {"Mic B Pre Amp", "MPB", "Mic B Source"}, + {"Mic B Source", NULL, "MIC2B"}, + + /* headphone capture */ + {"Capture Headphone Mux", "Stereo", "Capture Mixer"}, + {"Capture Headphone Mux", "Left", "Left Capture Source"}, + {"Capture Headphone Mux", "Right", "Right Capture Source"}, + + /* mono capture */ + {"Capture Mono Mux", "Stereo", "Capture Mixer"}, + {"Capture Mono Mux", "Left", "Left Capture Source"}, + {"Capture Mono Mux", "Right", "Right Capture Source"}, + + {NULL, NULL, NULL}, +}; + +static int wm9713_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); + + /* set up audio path audio_mapnects */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || + reg == AC97_CD) + return soc_ac97_ops.read(codec->ac97, reg); + else { + reg = reg >> 1; + + if (reg > (ARRAY_SIZE(wm9713_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg <= (ARRAY_SIZE(wm9713_reg))) + cache[reg] = val; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 divsel:1; + u32 divctl:1; + u32 lf:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the PLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 22) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + + /* The the PLL output is always 98.304MHz. */ + target = 98304000; + + /* If the input frequency is over 14.4MHz then scale it down. */ + if (source > 14400000) { + source >>= 1; + pll_div->divsel = 1; + + if (source > 14400000) { + source >>= 1; + pll_div->divctl = 1; + } else + pll_div->divctl = 0; + + } else { + pll_div->divsel = 0; + pll_div->divctl = 0; + } + + /* Low frequency sources require an additional divide in the + * loop. + */ + if (source < 8192000) { + pll_div->lf = 1; + target >>= 2; + } else + pll_div->lf = 0; + + Ndiv = target / source; + if ((Ndiv < 5) || (Ndiv > 12)) + printk(KERN_WARNING + "WM9713 PLL N value %d out of recommended range!\n", + Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +/** + * Please note that changing the PLL input frequency may require + * resynchronisation with the AC97 controller. + */ +static int wm9713_set_pll(struct snd_soc_codec *codec, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct wm9713_priv *wm9713 = codec->private_data; + u16 reg, reg2; + struct _pll_div pll_div; + + /* turn PLL off ? */ + if (freq_in == 0 || freq_out == 0) { + /* disable PLL power and select ext source */ + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + wm9713->pll_out = 0; + return 0; + } + + pll_factors(&pll_div, freq_in); + + if (pll_div.k == 0) { + reg = (pll_div.n << 12) | (pll_div.lf << 11) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } else { + /* write the fractional k to the reg 0x46 pages */ + reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + + /* K [21:20] */ + reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [19:16] */ + reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [15:12] */ + reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [11:8] */ + reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [7:4] */ + reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } + + /* turn PLL on and select as source */ + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + wm9713->pll_out = freq_out; + wm9713->pll_in = freq_in; + + /* wait 10ms AC97 link frames for the link to stabilise */ + schedule_timeout_interruptible(msecs_to_jiffies(10)); + return 0; +} + +static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + return wm9713_set_pll(codec, pll_id, freq_in, freq_out); +} + +/* + * Tristate the PCM DAI lines, tristate can be disabled by calling + * wm9713_set_dai_fmt() + */ +static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, + int tristate) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; + + if (tristate) + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + + return 0; +} + +/* + * Configure WM9713 clock dividers. + * Voice DAC needs 256 FS + */ +static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM9713_PCMCLK_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKA_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKB_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_HIFI_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_PCMBCLK_DIV: + reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + break; + case WM9713_PCMCLK_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + break; + case WM9713_HIFI_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 reg = 0x8000; + + /* clock masters */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= 0x4000; + gpio |= 0x0010; + break; + case SND_SOC_DAIFMT_CBM_CFS: + reg |= 0x6000; + gpio |= 0x0018; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= 0x0200; + gpio |= 0x001a; + break; + case SND_SOC_DAIFMT_CBS_CFM: + gpio |= 0x0012; + break; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + reg |= 0x00c0; + break; + case SND_SOC_DAIFMT_IB_NF: + reg |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= 0x0040; + break; + } + + /* DAI format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + reg |= 0x0043; + break; + } + + ac97_write(codec, AC97_GPIO_CFG, gpio); + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 0x000c; + break; + } + + /* enable PCM interface in master mode */ + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 status; + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; + ac97_write(codec, AC97_HANDSET_RATE, 0x0280); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, 0x0F80); + ac97_write(codec, AC97_EXTENDED_MID, status); +} + +static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +static int ac97_aux_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 vra, xsle; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + xsle = ac97_read(codec, AC97_PCI_SID); + ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + + return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); +} + +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_FORMATS \ + (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_codec_dai wm9713_dai[] = { +{ + .name = "AC97 HiFi", + .type = SND_SOC_DAI_AC97_BUS, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_hifi_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_aux_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "WM9713 Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .ops = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, + }, + }, +}; +EXPORT_SYMBOL_GPL(wm9713_dai); + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (!(ac97_read(codec, 0) & 0x8000)) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) & 0x8000) + return -EIO; + return 0; +} +EXPORT_SYMBOL_GPL(wm9713_reset); + +static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +{ + u16 reg; + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* enable thermal shutdown */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* enable master bias and vmid */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* disable everything including AC link */ + ac97_write(codec, AC97_EXTENDED_MID, 0xffff); + ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->dapm_state = event; + return 0; +} + +static int wm9713_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int wm9713_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm9713_priv *wm9713 = codec->private_data; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* do we need to re-start the PLL ? */ + if (wm9713->pll_out) + wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + + /* only synchronise the codec if warm reset failed */ + if (ret == 0) { + for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { + if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || + i == AC97_EXTENDED_MSTATUS || i > 0x66) + continue; + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + } + + if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + + return ret; +} + +static int wm9713_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0, reg; + + printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9713_reg); + codec->reg_cache_step = 2; + + codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL); + if (codec->private_data == NULL) { + ret = -ENOMEM; + goto priv_err; + } + + codec->name = "WM9713"; + codec->owner = THIS_MODULE; + codec->dai = wm9713_dai; + codec->num_dai = ARRAY_SIZE(wm9713_dai); + codec->write = ac97_write; + codec->read = ac97_read; + codec->dapm_event = wm9713_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + wm9713_reset(codec, 0); + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "AC97 link error\n"); + goto reset_err; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* unmute the adc - move to kcontrol */ + reg = ac97_read(codec, AC97_CD) & 0x7fff; + ac97_write(codec, AC97_CD, reg); + + wm9713_add_controls(codec); + wm9713_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); + +pcm_err: + snd_soc_free_ac97_codec(codec); + +codec_err: + kfree(codec->private_data); + +priv_err: + kfree(codec->reg_cache); + +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9713_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec->dai); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9713 = { + .probe = wm9713_soc_probe, + .remove = wm9713_soc_remove, + .suspend = wm9713_soc_suspend, + .resume = wm9713_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713); + +MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h new file mode 100644 index 000000000000..d357b6c8134b --- /dev/null +++ b/sound/soc/codecs/wm9713.h @@ -0,0 +1,53 @@ +/* + * wm9713.h -- WM9713 Soc Audio driver + */ + +#ifndef _WM9713_H +#define _WM9713_H + +/* clock inputs */ +#define WM9713_CLKA_PIN 0 +#define WM9713_CLKB_PIN 1 + +/* clock divider ID's */ +#define WM9713_PCMCLK_DIV 0 +#define WM9713_CLKA_MULT 1 +#define WM9713_CLKB_MULT 2 +#define WM9713_HIFI_DIV 3 +#define WM9713_PCMBCLK_DIV 4 +#define WM9713_PCMCLK_PLL_DIV 5 +#define WM9713_HIFI_PLL_DIV 6 + +/* Calculate the appropriate bit mask for the external PCM clock divider */ +#define WM9713_PCMDIV(x) ((x - 1) << 8) + +/* Calculate the appropriate bit mask for the external HiFi clock divider */ +#define WM9713_HIFIDIV(x) ((x - 1) << 12) + +/* MCLK clock mulitipliers */ +#define WM9713_CLKA_X1 (0 << 1) +#define WM9713_CLKA_X2 (1 << 1) +#define WM9713_CLKB_X1 (0 << 2) +#define WM9713_CLKB_X2 (1 << 2) + +/* MCLK clock MUX */ +#define WM9713_CLK_MUX_A (0 << 0) +#define WM9713_CLK_MUX_B (1 << 0) + +/* Voice DAI BCLK divider */ +#define WM9713_PCMBCLK_DIV_1 (0 << 9) +#define WM9713_PCMBCLK_DIV_2 (1 << 9) +#define WM9713_PCMBCLK_DIV_4 (2 << 9) +#define WM9713_PCMBCLK_DIV_8 (3 << 9) +#define WM9713_PCMBCLK_DIV_16 (4 << 9) + +#define WM9713_DAI_AC97_HIFI 0 +#define WM9713_DAI_AC97_AUX 1 +#define WM9713_DAI_PCM_VOICE 2 + +extern struct snd_soc_codec_device soc_codec_dev_wm9713; +extern struct snd_soc_codec_dai wm9713_dai[3]; + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm); + +#endif From c5059259688ab76f14f2f69a93e13575a36b614b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 16 Feb 2008 09:43:56 +0100 Subject: [PATCH 034/250] [ALSA] hda-codec - Add support of AD1883/1884A/1984A/1984B Added the support of new AD codecs: AD1883, AD1884A, AD1984A and AD1984B. These are almost compatible except for additional digital pins, etc. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 4 + sound/pci/hda/patch_analog.c | 327 ++++++++++++++++++ 2 files changed, 331 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 9a56b9b273cd..bfc6d486ad99 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -912,6 +912,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack mode (default) 6stack 6-stack mode + AD1884A / AD1883 / AD1984A / AD1984B + desktop 3-stack desktop (default) + laptop laptop with HP jack sensing + AD1884 N/A diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1f2102860fe8..b037fca1b44e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3410,6 +3410,329 @@ static int patch_ad1984(struct hda_codec *codec) } +/* + * AD1883 / AD1884A / AD1984A / AD1984B + * + * port-B (0x14) - front mic-in + * port-E (0x1c) - rear mic-in + * port-F (0x16) - CD / ext out + * port-C (0x15) - rear line-in + * port-D (0x12) - rear line-out + * port-A (0x11) - front hp-out + * + * AD1984A = AD1884A + digital-mic + * AD1883 = equivalent with AD1984A + * AD1984B = AD1984A + extra SPDIF-out + * + * FIXME: + * We share the single DAC for both HP and line-outs (see AD1884/1984). + */ + +static hda_nid_t ad1884a_dac_nids[1] = { + 0x03, +}; + +#define ad1884a_adc_nids ad1884_adc_nids +#define ad1884a_capsrc_nids ad1884_capsrc_nids + +#define AD1884A_SPDIF_OUT 0x02 + +static struct hda_input_mux ad1884a_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x0 }, + { "Mic", 0x4 }, + { "Line", 0x1 }, + { "CD", 0x2 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884a_base_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1884a_init_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-D (Line-out) mixer - route only from analog mixer */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer - route only from analog mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C (rear line-in) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-E (rear mic) pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */ + /* Port-F (CD) pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* SPDIF output amp */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1884a_loopbacks[] = { + { 0x20, HDA_INPUT, 0 }, /* Front Mic */ + { 0x20, HDA_INPUT, 1 }, /* Mic */ + { 0x20, HDA_INPUT, 2 }, /* CD */ + { 0x20, HDA_INPUT, 4 }, /* Docking */ + { } /* end */ +}; +#endif + +/* + * Laptop model + * + * Port A: Headphone jack + * Port B: MIC jack + * Port C: Internal MIC + * Port D: Dock Line Out (if enabled) + * Port E: Dock Line In (if enabled) + * Port F: Internal speakers + */ + +static struct hda_input_mux ad1884a_laptop_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, /* port-B */ + { "Internal Mic", 0x1 }, /* port-C */ + { "Dock Mic", 0x4 }, /* port-E */ + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* mute internal speaker if HP is plugged */ +static void ad1884a_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x11, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, + present ? 0x00 : 0x02); +} + +#define AD1884A_HP_EVENT 0x37 + +/* unsolicited event for HP jack sensing */ +static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1884a_hp_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1884a_hp_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + return 0; +} + +/* additional verbs for laptop model */ +static struct hda_verb ad1884a_laptop_verbs[] = { + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + { } /* end */ +}; + +/* + */ + +enum { + AD1884A_DESKTOP, + AD1884A_LAPTOP, + AD1884A_MODELS +}; + +static const char *ad1884a_models[AD1884A_MODELS] = { + [AD1884A_DESKTOP] = "desktop", + [AD1884A_LAPTOP] = "laptop", +}; + +static int patch_ad1884a(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); + spec->multiout.dac_nids = ad1884a_dac_nids; + spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids); + spec->adc_nids = ad1884a_adc_nids; + spec->capsrc_nids = ad1884a_capsrc_nids; + spec->input_mux = &ad1884a_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1884a_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1884a_init_verbs; + spec->spdif_route = 0; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1884a_loopbacks; +#endif + codec->patch_ops = ad198x_patch_ops; + + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, AD1884A_MODELS, + ad1884a_models, NULL); + switch (board_config) { + case AD1884A_LAPTOP: + spec->mixers[0] = ad1884a_laptop_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1884a_laptop_capture_source; + codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; + codec->patch_ops.init = ad1884a_hp_init; + break; + } + + return 0; +} + + /* * AD1882 * @@ -3709,8 +4032,12 @@ static int patch_ad1882(struct hda_codec *codec) * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, + { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, + { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a }, + { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, From b40b04ad380ad641e5740486e4b9a56fd32b64cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 16 Feb 2008 09:44:56 +0100 Subject: [PATCH 035/250] [ALSA] hda-codec - Add model=mobile for AD1884A & co Added the new model mobile for AD1884A and compatible codecs. It's a reduced version of model=laptop. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_analog.c | 48 ++++++++++++++++++- 2 files changed, 48 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index bfc6d486ad99..2cfb8b469c53 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -915,6 +915,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. AD1884A / AD1883 / AD1984A / AD1984B desktop 3-stack desktop (default) laptop laptop with HP jack sensing + mobile mobile devices with HP jack sensing AD1884 N/A diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b037fca1b44e..3f3905cc4e01 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3623,6 +3623,36 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; +static struct hda_input_mux ad1884a_mobile_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, /* port-C */ + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + /* mute internal speaker if HP is plugged */ static void ad1884a_hp_automute(struct hda_codec *codec) { @@ -3677,12 +3707,19 @@ static struct hda_verb ad1884a_laptop_verbs[] = { enum { AD1884A_DESKTOP, AD1884A_LAPTOP, + AD1884A_MOBILE, AD1884A_MODELS }; static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_DESKTOP] = "desktop", [AD1884A_LAPTOP] = "laptop", + [AD1884A_MOBILE] = "mobile", +}; + +static struct snd_pci_quirk ad1884a_cfg_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), + {} }; static int patch_ad1884a(struct hda_codec *codec) @@ -3717,7 +3754,8 @@ static int patch_ad1884a(struct hda_codec *codec) /* override some parameters */ board_config = snd_hda_check_board_config(codec, AD1884A_MODELS, - ad1884a_models, NULL); + ad1884a_models, + ad1884a_cfg_tbl); switch (board_config) { case AD1884A_LAPTOP: spec->mixers[0] = ad1884a_laptop_mixers; @@ -3727,6 +3765,14 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; break; + case AD1884A_MOBILE: + spec->mixers[0] = ad1884a_mobile_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1884a_mobile_capture_source; + codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; + codec->patch_ops.init = ad1884a_hp_init; + break; } return 0; From 310355c111dbae005269fe3fc39afdd60779bf5d Mon Sep 17 00:00:00 2001 From: Vladimir Barinov Date: Mon, 18 Feb 2008 11:40:22 +0100 Subject: [PATCH 036/250] [ALSA] Davinci ASoC support Add ASoC support for the TI Davinci SoC and the Davicni-EVM reference board. It includes: - ASoC Davinci DMA driver - ASoC Davinci I2S (Davinci McBSP module based) driver - ASoC Davinci-EVM reference board Signed-off-by: Vladimir Barinov Signed-off-by: Takashi Iwai --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/davinci/Kconfig | 19 ++ sound/soc/davinci/Makefile | 11 + sound/soc/davinci/davinci-evm.c | 208 ++++++++++++++++ sound/soc/davinci/davinci-i2s.c | 407 ++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s.h | 17 ++ sound/soc/davinci/davinci-pcm.c | 389 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-pcm.h | 29 +++ 9 files changed, 1082 insertions(+), 1 deletion(-) create mode 100644 sound/soc/davinci/Kconfig create mode 100644 sound/soc/davinci/Makefile create mode 100644 sound/soc/davinci/davinci-evm.c create mode 100644 sound/soc/davinci/davinci-i2s.c create mode 100644 sound/soc/davinci/davinci-i2s.h create mode 100644 sound/soc/davinci/davinci-pcm.c create mode 100644 sound/soc/davinci/davinci-pcm.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 276585215160..a3b51df2bea1 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -29,6 +29,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/fsl/Kconfig" +source "sound/soc/davinci/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 4869c9ae7a03..e489dbdde458 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig new file mode 100644 index 000000000000..20680c551aab --- /dev/null +++ b/sound/soc/davinci/Kconfig @@ -0,0 +1,19 @@ +config SND_DAVINCI_SOC + tristate "SoC Audio for the TI DAVINCI chip" + depends on ARCH_DAVINCI && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the DAVINCI AC97 or I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_DAVINCI_SOC_I2S + tristate + +config SND_DAVINCI_SOC_EVM + tristate "SoC Audio support for DaVinci EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci EVM platform. diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile new file mode 100644 index 000000000000..ca772e5b4637 --- /dev/null +++ b/sound/soc/davinci/Makefile @@ -0,0 +1,11 @@ +# DAVINCI Platform Support +snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-i2s-objs := davinci-i2s.o + +obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o + +# DAVINCI Machine Support +snd-soc-evm-objs := davinci-evm.o + +obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c new file mode 100644 index 000000000000..fcd165240333 --- /dev/null +++ b/sound/soc/davinci/davinci-evm.c @@ -0,0 +1,208 @@ +/* + * ASoC driver for TI DAVINCI EVM platform + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "../codecs/tlv320aic3x.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +#define EVM_CODEC_CLOCK 22579200 + +static int evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops evm_ops = { + .hw_params = evm_hw_params, +}; + +/* davinci-evm machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +/* davinci-evm machine audio_mapnections to the codec pins */ +static const char *audio_map[][3] = { + /* Headphone connected to HPLOUT, HPROUT */ + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LLOUT"}, + {"Line Out", NULL, "RLOUT"}, + + /* Mic connected to (MIC3L | MIC3R) */ + {"MIC3L", NULL, "Mic Bias 2V"}, + {"MIC3R", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "Mic Jack"}, + + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ + {"LINE1L", NULL, "Line In"}, + {"LINE2L", NULL, "Line In"}, + {"LINE1R", NULL, "Line In"}, + {"LINE2R", NULL, "Line In"}, + + {NULL, NULL, NULL}, +}; + +/* Logic for a aic3x as connected on a davinci-evm */ +static int evm_aic3x_init(struct snd_soc_codec *codec) +{ + int i; + + /* Add davinci-evm specific widgets */ + for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + + /* Set up davinci-evm specific audio path audio_map */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + /* not connected */ + snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); + snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); + snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + + /* always connected */ + snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line In", 1); + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +/* davinci-evm digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +/* davinci-evm audio machine driver */ +static struct snd_soc_machine snd_soc_machine_evm = { + .name = "DaVinci EVM", + .dai_link = &evm_dai, + .num_links = 1, +}; + +/* evm audio private data */ +static struct aic3x_setup_data evm_aic3x_setup = { + .i2c_address = 0x1b, +}; + +/* evm audio subsystem */ +static struct snd_soc_device evm_snd_devdata = { + .machine = &snd_soc_machine_evm, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &evm_aic3x_setup, +}; + +static struct resource evm_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data evm_snd_data = { + .tx_dma_ch = DM644X_DMACH_MCBSP_TX, + .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +}; + +static struct platform_device *evm_snd_device; + +static int __init evm_init(void) +{ + int ret; + + evm_snd_device = platform_device_alloc("soc-audio", 0); + if (!evm_snd_device) + return -ENOMEM; + + platform_set_drvdata(evm_snd_device, &evm_snd_devdata); + evm_snd_devdata.dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = &evm_snd_data; + + ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, + ARRAY_SIZE(evm_snd_resources)); + if (ret) { + platform_device_put(evm_snd_device); + return ret; + } + + ret = platform_device_add(evm_snd_device); + if (ret) + platform_device_put(evm_snd_device); + + return ret; +} + +static void __exit evm_exit(void) +{ + platform_device_unregister(evm_snd_device); +} + +module_init(evm_init); +module_exit(evm_exit); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c new file mode 100644 index 000000000000..c421774b33ee --- /dev/null +++ b/sound/soc/davinci/davinci-i2s.c @@ -0,0 +1,407 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "davinci-pcm.h" + +#define DAVINCI_MCBSP_DRR_REG 0x00 +#define DAVINCI_MCBSP_DXR_REG 0x04 +#define DAVINCI_MCBSP_SPCR_REG 0x08 +#define DAVINCI_MCBSP_RCR_REG 0x0c +#define DAVINCI_MCBSP_XCR_REG 0x10 +#define DAVINCI_MCBSP_SRGR_REG 0x14 +#define DAVINCI_MCBSP_PCR_REG 0x24 + +#define DAVINCI_MCBSP_SPCR_RRST (1 << 0) +#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4) +#define DAVINCI_MCBSP_SPCR_XRST (1 << 16) +#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20) +#define DAVINCI_MCBSP_SPCR_GRST (1 << 22) +#define DAVINCI_MCBSP_SPCR_FRST (1 << 23) +#define DAVINCI_MCBSP_SPCR_FREE (1 << 25) + +#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_XCR_XFIG (1 << 18) +#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) +#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) +#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) + +#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) +#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) +#define DAVINCI_MCBSP_PCR_FSRP (1 << 2) +#define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) +#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) +#define DAVINCI_MCBSP_PCR_FSRM (1 << 10) +#define DAVINCI_MCBSP_PCR_FSXM (1 << 11) + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +enum { + DAVINCI_MCBSP_WORD_8 = 0, + DAVINCI_MCBSP_WORD_12, + DAVINCI_MCBSP_WORD_16, + DAVINCI_MCBSP_WORD_20, + DAVINCI_MCBSP_WORD_24, + DAVINCI_MCBSP_WORD_32, +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { + .name = "I2S PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { + .name = "I2S PCM Stereo in", +}; + +struct davinci_mcbsp_dev { + void __iomem *base; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_mcbsp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Start frame sync */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST | + DAVINCI_MCBSP_SPCR_FRST, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} + +static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; + case SND_SOC_DAIFMT_CBM_CFM: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_IB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP | + DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_interval *i = NULL; + int mcbsp_word_length; + u32 w; + + /* general line settings */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + DAVINCI_MCBSP_SPCR_RINTM(3) | + DAVINCI_MCBSP_SPCR_XINTM(3) | + DAVINCI_MCBSP_SPCR_FREE); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + mcbsp_word_length = DAVINCI_MCBSP_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + mcbsp_word_length = DAVINCI_MCBSP_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + mcbsp_word_length = DAVINCI_MCBSP_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + + return 0; +} + +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcbsp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcbsp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int davinci_i2s_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *pdata; + int ret; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "McBSP region already claimed\n"); + return -EBUSY; + } + + dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_region; + } + + cpu_dai->private_data = dev; + + dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_free_mem; + } + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + pdata = pdev->dev.platform_data; + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + + return 0; + +err_free_mem: + kfree(dev); +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + return ret; +} + +static void davinci_i2s_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct resource *mem; + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); +} + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +struct snd_soc_cpu_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .type = SND_SOC_DAI_I2S, + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params,}, + .dai_ops = { + .set_fmt = davinci_i2s_set_dai_fmt, + }, +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h new file mode 100644 index 000000000000..9592d17db320 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _DAVINCI_I2S_H +#define _DAVINCI_I2S_H + +extern struct snd_soc_cpu_dai davinci_i2s_dai; + +#endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c new file mode 100644 index 000000000000..6a76927c9971 --- /dev/null +++ b/sound/soc/davinci/davinci-pcm.c @@ -0,0 +1,389 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include + +#include "davinci-pcm.h" + +#define DAVINCI_PCM_DEBUG 0 +#if DAVINCI_PCM_DEBUG +#define DPRINTK(x...) printk(KERN_DEBUG x) +#else +#define DPRINTK(x...) +#endif + +static struct snd_pcm_hardware davinci_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +struct davinci_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ + int master_lch; /* Master DMA channel */ + int slave_lch; /* Slave DMA channel */ + struct davinci_pcm_dma_params *params; /* DMA params */ +}; + +static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int lch = prtd->slave_lch; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + unsigned short src_bidx, dst_bidx; + unsigned int data_type; + unsigned int count; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x " + "period_size=%x\n", lch, dma_pos, period_size); + + data_type = prtd->params->data_type; + count = period_size / data_type; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = prtd->params->dma_addr; + src_bidx = data_type; + dst_bidx = 0; + } else { + src = prtd->params->dma_addr; + dst = dma_pos; + src_bidx = 0; + dst_bidx = data_type; + } + + davinci_set_dma_src_params(lch, src, INCR, W8BIT); + davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); + davinci_set_dma_src_index(lch, src_bidx, 0); + davinci_set_dma_dest_index(lch, dst_bidx, 0); + davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +{ + struct snd_pcm_substream *substream = data; + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + DPRINTK("lch=%d, status=0x%x\n", lch, ch_status); + + if (unlikely(ch_status != DMA_COMPLETE)) + return; + + if (snd_pcm_running(substream)) { + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } +} + +static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int tcc = TCC_ANY; + int ret; + + if (!dma_data) + return -ENODEV; + + prtd->params = dma_data; + + /* Request master DMA channel */ + ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + davinci_pcm_dma_irq, substream, + &prtd->master_lch, &tcc, EVENTQ_0); + if (ret) + return ret; + + /* Request slave DMA channel */ + ret = davinci_request_dma(PARAM_ANY, "Link", + NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); + if (ret) { + davinci_free_dma(prtd->master_lch); + return ret; + } + + /* Link slave DMA channel in loopback */ + davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + + return 0; +} + +static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_start_dma(prtd->master_lch); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_stop_dma(prtd->master_lch); + break; + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static int davinci_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct paramentry_descriptor temp; + + prtd->period = 0; + davinci_pcm_enqueue_dma(substream); + + /* Get slave channel dma params for master channel startup */ + davinci_get_dma_params(prtd->slave_lch, &temp); + davinci_set_dma_params(prtd->master_lch, &temp); + + return 0; +} + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + unsigned int offset; + dma_addr_t count; + dma_addr_t src, dst; + + spin_lock(&prtd->lock); + + davinci_dma_getposition(prtd->master_lch, &src, &dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = src - runtime->dma_addr; + else + count = dst - runtime->dma_addr;; + + spin_unlock(&prtd->lock); + + offset = bytes_to_frames(runtime, count); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + + prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + ret = davinci_pcm_dma_request(substream); + if (ret) { + printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); + kfree(prtd); + } + + return ret; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + + davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + + davinci_free_dma(prtd->slave_lch); + davinci_free_dma(prtd->master_lch); + + kfree(prtd); + + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .prepare = davinci_pcm_prepare, + .trigger = davinci_pcm_trigger, + .pointer = davinci_pcm_pointer, + .mmap = davinci_pcm_mmap, +}; + +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = davinci_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void davinci_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->capture.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform davinci_soc_platform = { + .name = "davinci-audio", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, + .pcm_free = davinci_pcm_free, +}; +EXPORT_SYMBOL_GPL(davinci_soc_platform); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h new file mode 100644 index 000000000000..8d6a45e75a6e --- /dev/null +++ b/sound/soc/davinci/davinci-pcm.h @@ -0,0 +1,29 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _DAVINCI_PCM_H +#define _DAVINCI_PCM_H + +struct davinci_pcm_dma_params { + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ +}; + +struct evm_snd_platform_data { + int tx_dma_ch; + int rx_dma_ch; +}; + +extern struct snd_soc_platform davinci_soc_platform; + +#endif From 2eef1258e54722b1c4efac6e5760d2153f96c4b4 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Mon, 18 Feb 2008 11:44:56 +0100 Subject: [PATCH 037/250] [ALSA] Add __devinit macro to at73c213 sound driver probe functions This patch adds __devinit to the functions used when probing. Will also reduce the memory footprint a bit if CONFIG_HOTPLUG is not enabled. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 89d6e9c35140..b8860b26fc62 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -737,7 +737,7 @@ cleanup: /* * Device functions */ -static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) { /* * Continuous clock output. @@ -767,7 +767,7 @@ static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) return 0; } -static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) { int retval; unsigned char dac_ctrl = 0; @@ -933,7 +933,7 @@ out: return retval; } -static int snd_at73c213_probe(struct spi_device *spi) +static int __devinit snd_at73c213_probe(struct spi_device *spi) { struct snd_card *card; struct snd_at73c213 *chip; From 4235a31784f59c9be5ff71534743c055091f9735 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Feb 2008 12:23:13 +0100 Subject: [PATCH 038/250] [ALSA] intel8x0 - Add support of 8 channel sound Added the support of 8 channel sound for codecs that are known to work. So far, only ALC850 is marked as a 8ch-support codec. This fix is a modified version of the patch on ALSA BTS#2097 by Martin Ellis: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097 Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 1 + sound/pci/ac97/ac97_patch.c | 46 ++++++++++++++++++++++++++----------- sound/pci/intel8x0.c | 28 +++++++++++++++++++--- 3 files changed, 59 insertions(+), 16 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 01480581f825..049edc5e6461 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -397,6 +397,7 @@ #define AC97_HAS_NO_TONE (1<<16) /* no Tone volume */ #define AC97_HAS_NO_STD_PCM (1<<17) /* no standard AC97 PCM volume and mute */ #define AC97_HAS_NO_AUX (1<<18) /* no standard AC97 AUX volume and mute */ +#define AC97_HAS_8CH (1<<19) /* supports 8-channel output */ /* rates indexes */ #define AC97_RATES_FRONT_DAC 0 diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 50c637e55ffa..39198e505b12 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -114,10 +114,9 @@ static int ac97_surround_jack_mode_put(struct snd_kcontrol *kcontrol, struct snd static int ac97_channel_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[] = { "2ch", "4ch", "6ch" }; - if (kcontrol->private_value) - return ac97_enum_text_info(kcontrol, uinfo, texts, 2); /* 4ch only */ - return ac97_enum_text_info(kcontrol, uinfo, texts, 3); + static const char *texts[] = { "2ch", "4ch", "6ch", "8ch" }; + return ac97_enum_text_info(kcontrol, uinfo, texts, + kcontrol->private_value); } static int ac97_channel_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -133,13 +132,8 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned char mode = ucontrol->value.enumerated.item[0]; - if (kcontrol->private_value) { - if (mode >= 2) - return -EINVAL; - } else { - if (mode >= 3) - return -EINVAL; - } + if (mode >= kcontrol->private_value) + return -EINVAL; if (mode != ac97->channel_mode) { ac97->channel_mode = mode; @@ -158,6 +152,7 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .get = ac97_surround_jack_mode_get, \ .put = ac97_surround_jack_mode_put, \ } +/* 6ch */ #define AC97_CHANNEL_MODE_CTL \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -165,7 +160,9 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .info = ac97_channel_mode_info, \ .get = ac97_channel_mode_get, \ .put = ac97_channel_mode_put, \ + .private_value = 3, \ } +/* 4ch */ #define AC97_CHANNEL_MODE_4CH_CTL \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -173,7 +170,17 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .info = ac97_channel_mode_info, \ .get = ac97_channel_mode_get, \ .put = ac97_channel_mode_put, \ - .private_value = 1, \ + .private_value = 2, \ + } +/* 8ch */ +#define AC97_CHANNEL_MODE_8CH_CTL \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Channel Mode", \ + .info = ac97_channel_mode_info, \ + .get = ac97_channel_mode_get, \ + .put = ac97_channel_mode_put, \ + .private_value = 4, \ } static inline int is_surround_on(struct snd_ac97 *ac97) @@ -210,6 +217,10 @@ static inline int is_shared_micin(struct snd_ac97 *ac97) return !ac97->indep_surround && !is_clfe_on(ac97); } +static inline int alc850_is_aux_back_surround(struct snd_ac97 *ac97) +{ + return is_surround_on(ac97); +} /* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */ /* Modified for YMF743 by Keita Maehara */ @@ -2816,10 +2827,12 @@ static int patch_alc655(struct snd_ac97 * ac97) #define AC97_ALC850_JACK_SELECT 0x76 #define AC97_ALC850_MISC1 0x7a +#define AC97_ALC850_MULTICH 0x6a static void alc850_update_jacks(struct snd_ac97 *ac97) { int shared; + int aux_is_back_surround; /* shared Line-In / Surround Out */ shared = is_shared_surrout(ac97); @@ -2837,13 +2850,18 @@ static void alc850_update_jacks(struct snd_ac97 *ac97) /* MIC-IN = 1, CENTER-LFE = 5 */ snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 4, shared ? (5<<4) : (1<<4)); + + aux_is_back_surround = alc850_is_aux_back_surround(ac97); + /* Aux is Back Surround */ + snd_ac97_update_bits(ac97, AC97_ALC850_MULTICH, 1 << 10, + aux_is_back_surround ? (1<<10) : (0<<10)); } static const struct snd_kcontrol_new snd_ac97_controls_alc850[] = { AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0), AC97_SINGLE("Mic Front Input Switch", AC97_ALC850_JACK_SELECT, 15, 1, 1), AC97_SURROUND_JACK_MODE_CTL, - AC97_CHANNEL_MODE_CTL, + AC97_CHANNEL_MODE_8CH_CTL, }; static int patch_alc850_specific(struct snd_ac97 *ac97) @@ -2869,6 +2887,7 @@ static int patch_alc850(struct snd_ac97 *ac97) ac97->build_ops = &patch_alc850_ops; ac97->spec.dev_flags = 0; /* for IEC958 playback route - ALC655 compatible */ + ac97->flags |= AC97_HAS_8CH; /* assume only page 0 for writing cache */ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR); @@ -2878,6 +2897,7 @@ static int patch_alc850(struct snd_ac97 *ac97) spdif-in monitor off, spdif-in PCM off center on mic off, surround on line-in off duplicate front off + NB default bit 10=0 = Aux is Capture, not Back Surround */ snd_ac97_write_cache(ac97, AC97_ALC650_MULTICH, 1<<15); /* SURR_OUT: on, Surr 1kOhm: on, Surr Amp: off, Front 1kOhm: off diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c52abd0bf22e..07782ba9c74d 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -155,7 +155,8 @@ DEFINE_REGSET(SP, 0x60); /* SPDIF out */ #define ICH_PCM_SPDIF_69 0x80000000 /* s/pdif pcm on slots 6&9 */ #define ICH_PCM_SPDIF_1011 0xc0000000 /* s/pdif pcm on slots 10&11 */ #define ICH_PCM_20BIT 0x00400000 /* 20-bit samples (ICH4) */ -#define ICH_PCM_246_MASK 0x00300000 /* 6 channels (not all chips) */ +#define ICH_PCM_246_MASK 0x00300000 /* chan mask (not all chips) */ +#define ICH_PCM_8 0x00300000 /* 8 channels (not all chips) */ #define ICH_PCM_6 0x00200000 /* 6 channels (not all chips) */ #define ICH_PCM_4 0x00100000 /* 4 channels (not all chips) */ #define ICH_PCM_2 0x00000000 /* 2 channels (stereo) */ @@ -382,6 +383,7 @@ struct intel8x0 { unsigned multi4: 1, multi6: 1, + multi8 :1, dra: 1, smp20bit: 1; unsigned in_ac97_init: 1, @@ -997,6 +999,8 @@ static void snd_intel8x0_setup_pcm_out(struct intel8x0 *chip, cnt |= ICH_PCM_4; else if (runtime->channels == 6) cnt |= ICH_PCM_6; + else if (runtime->channels == 8) + cnt |= ICH_PCM_8; if (chip->device_type == DEVICE_NFORCE) { /* reset to 2ch once to keep the 6 channel data in alignment, * to start from Front Left always @@ -1106,6 +1110,16 @@ static struct snd_pcm_hw_constraint_list hw_constraints_channels6 = { .mask = 0, }; +static unsigned int channels8[] = { + 2, 4, 6, 8, +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_channels8 = { + .count = ARRAY_SIZE(channels8), + .list = channels8, + .mask = 0, +}; + static int snd_intel8x0_pcm_open(struct snd_pcm_substream *substream, struct ichdev *ichdev) { struct intel8x0 *chip = snd_pcm_substream_chip(substream); @@ -1136,7 +1150,12 @@ static int snd_intel8x0_playback_open(struct snd_pcm_substream *substream) if (err < 0) return err; - if (chip->multi6) { + if (chip->multi8) { + runtime->hw.channels_max = 8; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_constraints_channels8); + } else if (chip->multi6) { runtime->hw.channels_max = 6; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels6); @@ -2203,8 +2222,11 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, } if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_PCM_SLEFT)) { chip->multi4 = 1; - if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_LFE)) + if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_LFE)) { chip->multi6 = 1; + if (chip->ac97[0]->flags & AC97_HAS_8CH) + chip->multi8 = 1; + } } if (pbus->pcms[0].r[1].rslots[0]) { chip->dra = 1; From e922b0028fad87de0d262f9fa51f98595d2df258 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Feb 2008 13:03:13 +0100 Subject: [PATCH 039/250] [ALSA] Move vmaster code to sound core Move the codes for virtual master controls to sound core part so that not only hda-intel drivers can use it. Signed-off-by: Takashi Iwai --- include/sound/control.h | 7 +++++++ sound/core/Kconfig | 4 ++++ sound/core/Makefile | 1 + sound/{pci/hda => core}/vmaster.c | 4 ++++ sound/pci/Kconfig | 1 + sound/pci/hda/hda_local.h | 7 ------- 6 files changed, 17 insertions(+), 7 deletions(-) rename sound/{pci/hda => core}/vmaster.c (99%) diff --git a/include/sound/control.h b/include/sound/control.h index e79baa63912f..3dc1291f52db 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -169,4 +169,11 @@ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +/* + * virtual master control + */ +struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, + const unsigned int *tlv); +int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave); + #endif /* __SOUND_CONTROL_H */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 829ca38b595e..a8d71c6c8e75 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -181,3 +181,7 @@ config SND_PCM_XRUN_DEBUG It is usually not required, but if you have trouble with sound clicking when system is loaded, it may help to determine the process or driver which causes the scheduling gaps. + +config SND_VMASTER + bool + depends on SND diff --git a/sound/core/Makefile b/sound/core/Makefile index 267039a97bd5..da8e685eef9c 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -6,6 +6,7 @@ snd-y := sound.o init.o memory.o info.o control.o misc.o device.o snd-$(CONFIG_ISA_DMA_API) += isadma.o snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o +snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o diff --git a/sound/pci/hda/vmaster.c b/sound/core/vmaster.c similarity index 99% rename from sound/pci/hda/vmaster.c rename to sound/core/vmaster.c index 2da49d20a1fc..7cfd8b8fb4e7 100644 --- a/sound/pci/hda/vmaster.c +++ b/sound/core/vmaster.c @@ -253,6 +253,8 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) return 0; } +EXPORT_SYMBOL(snd_ctl_add_slave); + /* * ctl callbacks for master controls */ @@ -362,3 +364,5 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, } return kctl; } + +EXPORT_SYMBOL(snd_ctl_make_virtual_master); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 812085d521f1..48296d97bf33 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -517,6 +517,7 @@ config SND_HDA_INTEL tristate "Intel HD Audio" depends on SND select SND_PCM + select SND_VMASTER help Say Y here to include support for Intel "High Definition Audio" (Azalia) motherboard devices. diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ce2ad42a8a8a..5c9e578f7f2d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -418,11 +418,4 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, hda_nid_t nid); #endif /* CONFIG_SND_HDA_POWER_SAVE */ -/* - * virtual master control - */ -struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, - const unsigned int *tlv); -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave); - #endif /* __SOUND_HDA_LOCAL_H */ From 1c82ed1bc531746a8fa9b46c593ddce546f28026 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Feb 2008 13:05:50 +0100 Subject: [PATCH 040/250] [ALSA] Keep private TLV entry in vmaster itself Use a private array for TLV entries of virtual master controls instead of (supposed) static array. This cleans up the existing codes. Also, now vmaster assumes the simple dB-range TLV that is the only type it can handle. Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 9 ++++++--- sound/pci/hda/Makefile | 2 +- sound/pci/hda/patch_analog.c | 6 +++--- sound/pci/hda/patch_realtek.c | 6 +++--- sound/pci/hda/patch_sigmatel.c | 8 +++----- 5 files changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 7cfd8b8fb4e7..4cc57f902e2c 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -12,6 +12,7 @@ #include #include #include +#include /* * a subset of information returned via ctl info callback @@ -34,6 +35,7 @@ struct link_master { struct list_head slaves; struct link_ctl_info info; int val; /* the master value */ + unsigned int tlv[4]; }; /* @@ -357,11 +359,12 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, kctl->private_free = master_free; /* additional (constant) TLV read */ - if (tlv) { - /* FIXME: this assumes that the max volume is 0 dB */ + if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - kctl->tlv.p = tlv; + memcpy(master->tlv, tlv, sizeof(master->tlv)); + kctl->tlv.p = master->tlv; } + return kctl; } diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 9e0d8a1268aa..ab0c726d648e 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -2,7 +2,7 @@ snd-hda-intel-y := hda_intel.o # since snd-hda-intel is the only driver using hda-codec, # merge it into a single module although it was originally # designed to be individual modules -snd-hda-intel-y += hda_codec.o vmaster.o +snd-hda-intel-y += hda_codec.o snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3f3905cc4e01..e0f3559f8b13 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -80,7 +80,6 @@ struct ad198x_spec { #endif /* for virtual master */ hda_nid_t vmaster_nid; - u32 vmaster_tlv[4]; const char **slave_vols; const char **slave_sws; }; @@ -185,10 +184,11 @@ static int ad198x_build_controls(struct hda_codec *codec) /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); + HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, + vmaster_tlv, (spec->slave_vols ? spec->slave_vols : ad_slave_vols)); if (err < 0) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eea18b3336d3..e8ce525d297b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -272,7 +272,6 @@ struct alc_spec { /* for virtual master */ hda_nid_t vmaster_nid; - u32 vmaster_tlv[4]; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -1534,10 +1533,11 @@ static int alc_build_controls(struct hda_codec *codec) /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); + HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, alc_slave_vols); + vmaster_tlv, alc_slave_vols); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7901e76f2690..132d1e3eafa5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -186,9 +186,6 @@ struct sigmatel_spec { struct hda_input_mux private_dimux; struct hda_input_mux private_imux; struct hda_input_mux private_mono_mux; - - /* virtual master */ - unsigned int vmaster_tlv[4]; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -930,10 +927,11 @@ static int stac92xx_build_controls(struct hda_codec *codec) /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], - HDA_OUTPUT, spec->vmaster_tlv); + HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, slave_vols); + vmaster_tlv, slave_vols); if (err < 0) return err; } From 49c88b85b53767f97eb8c9171cb0b976c62a0114 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Feb 2008 13:06:49 +0100 Subject: [PATCH 041/250] [ALSA] ca0106 - Add master volume controls Added master volume and switch controls for ca0106 using vmaster. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 + sound/pci/ca0106/ca0106_mixer.c | 51 +++++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 48296d97bf33..b05435cfee1e 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -162,6 +162,7 @@ config SND_CA0106 depends on SND select SND_AC97_CODEC select SND_RAWMIDI + select SND_VMASTER help Say Y here to include support for the Sound Blaster Audigy LS and Live 24bit. diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index af736869d9b1..94618ecaab65 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -658,11 +658,47 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch } \ } while (0) +static __devinitdata +DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 50, 1); + +static char *slave_vols[] __devinitdata = { + "Analog Front Playback Volume", + "Analog Rear Playback Volume", + "Analog Center/LFE Playback Volume", + "Analog Side Playback Volume", + "IEC958 Front Playback Volume", + "IEC958 Rear Playback Volume", + "IEC958 Center/LFE Playback Volume", + "IEC958 Unknown Playback Volume", + "CAPTURE feedback Playback Volume", + NULL +}; + +static char *slave_sws[] __devinitdata = { + "Analog Front Playback Switch", + "Analog Rear Playback Switch", + "Analog Center/LFE Playback Switch", + "Analog Side Playback Switch", + "IEC958 Playback Switch", + NULL +}; + +static void __devinit add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + if (slave) + snd_ctl_add_slave(master, slave); + } +} + int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) { int err; struct snd_card *card = emu->card; char **c; + struct snd_kcontrol *vmaster; static char *ca0106_remove_ctls[] = { "Master Mono Playback Switch", "Master Mono Playback Volume", @@ -719,6 +755,21 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) } if (emu->details->spi_dac == 1) ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls); + + /* Create virtual master controls */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + snd_ca0106_master_db_scale); + if (!vmaster) + return -ENOMEM; + add_slaves(card, vmaster, slave_vols); + + if (emu->details->spi_dac == 1) { + vmaster = snd_ctl_make_virtual_master("Master Playback Switch", + NULL); + if (!vmaster) + return -ENOMEM; + add_slaves(card, vmaster, slave_sws); + } return 0; } From 8b6ed8e70d9a7c39748a9902d64138e070d4064b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Feb 2008 11:36:35 +0100 Subject: [PATCH 042/250] [ALSA] hda-intel - Clean up stream definitions Clean up the code to define playback/capture streams. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 29 ++++++++--------------------- 1 file changed, 8 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 66dceffa1217..cf1a1d0124fe 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -185,21 +185,15 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* max number of SDs */ /* ICH, ATI and VIA have 4 playback and 4 capture */ -#define ICH6_CAPTURE_INDEX 0 #define ICH6_NUM_CAPTURE 4 -#define ICH6_PLAYBACK_INDEX 4 #define ICH6_NUM_PLAYBACK 4 /* ULI has 6 playback and 5 capture */ -#define ULI_CAPTURE_INDEX 0 #define ULI_NUM_CAPTURE 5 -#define ULI_PLAYBACK_INDEX 5 #define ULI_NUM_PLAYBACK 6 /* ATI HDMI has 1 playback and 0 capture */ -#define ATIHDMI_CAPTURE_INDEX 0 #define ATIHDMI_NUM_CAPTURE 0 -#define ATIHDMI_PLAYBACK_INDEX 0 #define ATIHDMI_NUM_PLAYBACK 1 /* this number is statically defined for simplicity */ @@ -1846,38 +1840,31 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK)) pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK); - if (gcap) { - /* read number of streams from GCAP register instead of using - * hardcoded value - */ - chip->playback_streams = (gcap & (0xF << 12)) >> 12; - chip->capture_streams = (gcap & (0xF << 8)) >> 8; - chip->playback_index_offset = chip->capture_streams; - chip->capture_index_offset = 0; - } else { + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { /* gcap didn't give any info, switching to old method */ switch (chip->driver_type) { case AZX_DRIVER_ULI: chip->playback_streams = ULI_NUM_PLAYBACK; chip->capture_streams = ULI_NUM_CAPTURE; - chip->playback_index_offset = ULI_PLAYBACK_INDEX; - chip->capture_index_offset = ULI_CAPTURE_INDEX; break; case AZX_DRIVER_ATIHDMI: chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; - chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX; - chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX; break; default: chip->playback_streams = ICH6_NUM_PLAYBACK; chip->capture_streams = ICH6_NUM_CAPTURE; - chip->playback_index_offset = ICH6_PLAYBACK_INDEX; - chip->capture_index_offset = ICH6_CAPTURE_INDEX; break; } } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); From 77a261b75521564dcc5f22355cce4830f6b1376a Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 19 Feb 2008 11:38:05 +0100 Subject: [PATCH 043/250] [ALSA] hda-codec - Fix ALC662 recording Fixed ALC662 recording issue. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e8ce525d297b..62e1bd882a57 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12752,7 +12752,7 @@ static hda_nid_t alc662_adc_nids[1] = { 0x09, }; -static hda_nid_t alc662_capsrc_nids[1] = { 0x23 }; +static hda_nid_t alc662_capsrc_nids[1] = { 0x22 }; /* input MUX */ /* FIXME: should be a matrix-type input source selection */ From aef9d318b1d741d80486ff7ea3507a8321dedf6b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Feb 2008 13:16:41 +0100 Subject: [PATCH 044/250] [ALSA] hda-codec - Add beep volume control to ALC268 Added the beep volume control to ALC268 codec support code. Since the codec doesn't return the correct AMP caps, we need to override the value. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 53 +++++++++++++++++++++++++++++------ 1 file changed, 45 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62e1bd882a57..6c8423dbace1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9507,6 +9507,22 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { { } }; +/* bind Beep switches of both NID 0x0f and 0x10 */ +static struct hda_bind_ctls alc268_bind_beep_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc268_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT), + HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw), + { } +}; + static struct hda_verb alc268_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, @@ -9703,7 +9719,11 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + + /* set PCBEEP vol = 0, mute connections */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Unmute Selector 23h,24h and set the default input to mic-in */ @@ -9742,8 +9762,10 @@ static struct hda_verb alc268_volume_init_verbs[] = { {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* set PCBEEP vol = 0 */ - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))}, + /* set PCBEEP vol = 0, mute connections */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, { } }; @@ -10032,6 +10054,9 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->autocfg.speaker_pins[0] != 0x1d) + spec->mixers[spec->num_mixers++] = alc268_beep_mixer; + spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs; spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; @@ -10091,7 +10116,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC268_3ST] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, @@ -10105,7 +10131,8 @@ static struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -10121,7 +10148,8 @@ static struct alc_config_preset alc268_presets[] = { .init_hook = alc268_toshiba_automute, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -10137,7 +10165,7 @@ static struct alc_config_preset alc268_presets[] = { .init_hook = alc268_acer_init_hook, }, [ALC268_DELL] = { - .mixers = { alc268_dell_mixer }, + .mixers = { alc268_dell_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_dell_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -10150,7 +10178,8 @@ static struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_ZEPTO] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -10232,6 +10261,14 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; + if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) + /* override the amp caps for beep generator */ + snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, + (0x0c << AC_AMPCAP_OFFSET_SHIFT) | + (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); From 85860c06aba5e145805ad840553a2388e60a7e23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Feb 2008 15:00:15 +0100 Subject: [PATCH 045/250] [ALSA] hda-codec - Fix ALC268 capture source Initialize the capture source properly for auto model. It's especially important for cases that only mic is detected. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6c8423dbace1..630c7b22542a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10272,6 +10272,7 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); + int i; /* get type */ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; @@ -10289,6 +10290,11 @@ static int patch_alc268(struct hda_codec *codec) spec->num_mixers++; } spec->capsrc_nids = alc268_capsrc_nids; + /* set default input source */ + for (i = 0; i < spec->num_adc_nids; i++) + snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], + 0, AC_VERB_SET_CONNECT_SEL, + spec->input_mux->items[0].index); } spec->vmaster_nid = 0x02; From 67ebcb0311110dc7268bb5b135bf437d8033337e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Feb 2008 15:03:57 +0100 Subject: [PATCH 046/250] [ALSA] hda-codec - Don't create multiple capture streams for single inputs When the device has only one input source, it makes no sense to have multiple capture streams. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 630c7b22542a..2100ee480809 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4950,7 +4950,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) /* check whether NID 0x04 is valid */ wcap = get_wcaps(codec, 0x04); wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ - if (wcap != AC_WID_AUD_IN) { + if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc260_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt); spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer; @@ -10276,7 +10276,7 @@ static int patch_alc268(struct hda_codec *codec) /* get type */ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { + if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); spec->mixers[spec->num_mixers] = From 98f2a97f207a776603173ee96327d977e592579d Mon Sep 17 00:00:00 2001 From: Cedric Bregardis Date: Wed, 20 Feb 2008 12:05:13 +0100 Subject: [PATCH 047/250] [ALSA] Emagic Audiowerk 2 ALSA driver. Signed-off-by: Cedric Bregardis Signed-off-by: Jean-Christian Hassler Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 15 + sound/pci/Makefile | 1 + sound/pci/aw2/Makefile | 3 + sound/pci/aw2/aw2-alsa.c | 787 ++++++++++++++++++++++++++++++++++++ sound/pci/aw2/aw2-saa7146.c | 464 +++++++++++++++++++++ sound/pci/aw2/aw2-saa7146.h | 105 +++++ sound/pci/aw2/aw2-tsl.h | 116 ++++++ sound/pci/aw2/saa7146.h | 168 ++++++++ 8 files changed, 1659 insertions(+) create mode 100644 sound/pci/aw2/Makefile create mode 100644 sound/pci/aw2/aw2-alsa.c create mode 100644 sound/pci/aw2/aw2-saa7146.c create mode 100644 sound/pci/aw2/aw2-saa7146.h create mode 100644 sound/pci/aw2/aw2-tsl.h create mode 100644 sound/pci/aw2/saa7146.h diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index b05435cfee1e..868183bef243 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -122,6 +122,21 @@ config SND_AU8830 To compile this driver as a module, choose M here: the module will be called snd-au8830. +config SND_AW2 + tristate "Emagic Audiowerk 2" + depends on SND + help + Say Y here to include support for Emagic Audiowerk 2 soundcards. + + Supported features: Analog and SPDIF output. Analog or SPDIF input. + Note: Switch between analog and digital input does not always work. + It can produce continuous noise. The workaround is to switch again + (and again) between digital and analog input until it works. + + To compile this driver as a module, choose M here: the module + will be called snd-aw2. + + config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" depends on SND && EXPERIMENTAL diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 2d42fd28f4e7..85ef14bc8056 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -58,6 +58,7 @@ obj-$(CONFIG_SND) += \ ac97/ \ ali5451/ \ au88x0/ \ + aw2/ \ ca0106/ \ cs46xx/ \ cs5535audio/ \ diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile new file mode 100644 index 000000000000..842335d3b735 --- /dev/null +++ b/sound/pci/aw2/Makefile @@ -0,0 +1,3 @@ +snd-aw2-objs := aw2-alsa.o aw2-saa7146.o + +obj-$(CONFIG_SND_AW2) += snd-aw2.o diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c new file mode 100644 index 000000000000..74af639b9132 --- /dev/null +++ b/sound/pci/aw2/aw2-alsa.c @@ -0,0 +1,787 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "saa7146.h" +#include "aw2-saa7146.h" + +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Cedric Bregardis , " + "Jean-Christian Hassler "); +MODULE_DESCRIPTION("Emagic Audiowerk 2 sound driver"); +MODULE_LICENSE("GPL"); + +/********************************* + * DEFINES + ********************************/ +#define PCI_VENDOR_ID_SAA7146 0x1131 +#define PCI_DEVICE_ID_SAA7146 0x7146 + +#define CTL_ROUTE_ANALOG 0 +#define CTL_ROUTE_DIGITAL 1 + +/********************************* + * TYPEDEFS + ********************************/ + /* hardware definition */ +static struct snd_pcm_hardware snd_aw2_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 4, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_aw2_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +struct aw2_pcm_device { + struct snd_pcm *pcm; + unsigned int stream_number; + struct aw2 *chip; +}; + +struct aw2 { + struct snd_aw2_saa7146 saa7146; + + struct pci_dev *pci; + int irq; + spinlock_t reg_lock; + struct mutex mtx; + + unsigned long iobase_phys; + void __iomem *iobase_virt; + + struct snd_card *card; + + struct aw2_pcm_device device_playback[NB_STREAM_PLAYBACK]; + struct aw2_pcm_device device_capture[NB_STREAM_CAPTURE]; +}; + +/********************************* + * FUNCTION DECLARATIONS + ********************************/ +static int __init alsa_card_aw2_init(void); +static void __exit alsa_card_aw2_exit(void); +static int snd_aw2_dev_free(struct snd_device *device); +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2 **rchip); +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id); +static void __devexit snd_aw2_remove(struct pci_dev *pci); +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd); +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream); +static int __devinit snd_aw2_new_pcm(struct aw2 *chip); + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); + +/********************************* + * VARIABLES + ********************************/ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static struct pci_device_id snd_aw2_ids[] = { + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + 0, 0, 0}, + {0} +}; + +MODULE_DEVICE_TABLE(pci, snd_aw2_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = "Emagic Audiowerk 2", + .id_table = snd_aw2_ids, + .probe = snd_aw2_probe, + .remove = __devexit_p(snd_aw2_remove), +}; + +/* operators for playback PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_playback_ops = { + .open = snd_aw2_pcm_playback_open, + .close = snd_aw2_pcm_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_playback, + .trigger = snd_aw2_pcm_trigger_playback, + .pointer = snd_aw2_pcm_pointer_playback, +}; + +/* operators for capture PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_capture_ops = { + .open = snd_aw2_pcm_capture_open, + .close = snd_aw2_pcm_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_capture, + .trigger = snd_aw2_pcm_trigger_capture, + .pointer = snd_aw2_pcm_pointer_capture, +}; + +static struct snd_kcontrol_new aw2_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Capture Route", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = snd_aw2_control_switch_capture_info, + .get = snd_aw2_control_switch_capture_get, + .put = snd_aw2_control_switch_capture_put +}; + +/********************************* + * FUNCTION IMPLEMENTATIONS + ********************************/ + +/* initialization of the module */ +static int __init alsa_card_aw2_init(void) +{ + snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n"); + return pci_register_driver(&driver); +} + +/* clean up the module */ +static void __exit alsa_card_aw2_exit(void) +{ + snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n"); + pci_unregister_driver(&driver); +} + +module_init(alsa_card_aw2_init); +module_exit(alsa_card_aw2_exit); + +/* component-destructor */ +static int snd_aw2_dev_free(struct snd_device *device) +{ + struct aw2 *chip = device->device_data; + + /* Free hardware */ + snd_aw2_saa7146_free(&chip->saa7146); + + /* release the irq */ + if (chip->irq >= 0) + free_irq(chip->irq, (void *)chip); + /* release the i/o ports & memory */ + if (chip->iobase_virt) + iounmap(chip->iobase_virt); + + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + + return 0; +} + +/* chip-specific constructor */ +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2 **rchip) +{ + struct aw2 *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_aw2_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + pci_set_master(pci); + + /* check PCI availability (32bit DMA) */ + if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) || + (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) { + printk(KERN_ERR "aw2: Impossible to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "Audiowerk2"); + if (err < 0) { + pci_disable_device(pci); + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = + ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); + + if (chip->iobase_virt == NULL) { + printk(KERN_ERR "aw2: unable to remap memory region"); + pci_release_regions(pci); + pci_disable_device(pci); + kfree(chip); + return -ENOMEM; + } + + + if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, + IRQF_SHARED, "Audiowerk2", chip)) { + printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq); + + iounmap(chip->iobase_virt); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return -EBUSY; + } + chip->irq = pci->irq; + + /* (2) initialization of the chip hardware */ + snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + free_irq(chip->irq, (void *)chip); + iounmap(chip->iobase_virt); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + + printk(KERN_INFO + "Audiowerk 2 sound card (saa7146 chipset) detected and " + "managed\n"); + return 0; +} + +/* constructor */ +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct aw2 *chip; + int err; + + /* (1) Continue if device is not enabled, else inc dev */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) Create card instance */ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + /* (3) Create main component */ + err = snd_aw2_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* initialize mutex */ + mutex_init(&chip->mtx); + /* init spinlock */ + spin_lock_init(&chip->reg_lock); + /* (4) Define driver ID and name string */ + strcpy(card->driver, "aw2"); + strcpy(card->shortname, "Audiowerk2"); + + sprintf(card->longname, "%s with SAA7146 irq %i", + card->shortname, chip->irq); + + /* (5) Create other components */ + snd_aw2_new_pcm(chip); + + /* (6) Register card instance */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) Set PCI driver data */ + pci_set_drvdata(pci, card); + + dev++; + return 0; +} + +/* destructor */ +static void __devexit snd_aw2_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* open callback */ +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_printdd(KERN_DEBUG "aw2: Playback_open \n"); + runtime->hw = snd_aw2_playback_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream) +{ + return 0; + +} + +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_printdd(KERN_DEBUG "aw2: Capture_open \n"); + runtime->hw = snd_aw2_capture_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream) +{ + /* TODO: something to do ? */ + return 0; +} + + /* hw_params callback */ +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare callback for playback */ +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + mutex_lock(&chip->mtx); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_playback_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + mutex_unlock(&chip->mtx); + + return 0; +} + +/* prepare callback for capture */ +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + mutex_lock(&chip->mtx); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_capture_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + mutex_unlock(&chip->mtx); + + return 0; +} + +/* playback trigger callback */ +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + spin_lock(&chip->reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->reg_lock); + return status; +} + +/* capture trigger callback */ +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + spin_lock(&chip->reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->reg_lock); + return status; +} + +/* playback pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_ptr_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* capture pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_ptr_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* create a pcm device */ +static int __devinit snd_aw2_new_pcm(struct aw2 *chip) +{ + struct snd_pcm *pcm_playback_ana; + struct snd_pcm *pcm_playback_num; + struct snd_pcm *pcm_capture; + struct aw2_pcm_device *pcm_device; + int err = 0; + + /* Create new Alsa PCM device */ + + err = snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0, + &pcm_playback_ana); + if (err < 0) { + printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err); + return err; + } + + /* Creation ok */ + pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_ANA]; + + /* Set PCM device name */ + strcpy(pcm_playback_ana->name, "Analog playback"); + /* Associate private data to PCM device */ + pcm_playback_ana->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_ana, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_ana; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->pci), + 64 * 1024, 64 * 1024); + if (err) + printk(KERN_ERR "aw2: snd_pcm_lib_preallocate_pages_for_all " + "error (0x%X)\n", err); + + err = snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0, + &pcm_playback_num); + + if (err < 0) { + printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err); + return err; + } + /* Creation ok */ + pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_DIG]; + + /* Set PCM device name */ + strcpy(pcm_playback_num->name, "Digital playback"); + /* Associate private data to PCM device */ + pcm_playback_num->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_num, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_num; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_DIG; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->pci), + 64 * 1024, 64 * 1024); + if (err) + printk(KERN_ERR + "aw2: snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + + + + err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1, + &pcm_capture); + + if (err < 0) { + printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err); + return err; + } + + /* Creation ok */ + pcm_device = &chip->device_capture[NUM_STREAM_CAPTURE_ANA]; + + /* Set PCM device name */ + strcpy(pcm_capture->name, "Capture"); + /* Associate private data to PCM device */ + pcm_capture->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_capture, SNDRV_PCM_STREAM_CAPTURE, + &snd_aw2_capture_ops); + /* store PCM device */ + pcm_device->pcm = pcm_capture; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_CAPTURE_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = snd_pcm_lib_preallocate_pages_for_all(pcm_capture, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->pci), + 64 * 1024, 64 * 1024); + if (err) + printk(KERN_ERR + "aw2: snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + + + /* Create control */ + err = snd_ctl_add(chip->card, snd_ctl_new1(&aw2_control, chip)); + if (err < 0) { + printk(KERN_ERR "aw2: snd_ctl_add error (0x%X)\n", err); + return err; + } + + return 0; +} + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { + "Analog", "Digital" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) { + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + } + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2 *chip = snd_kcontrol_chip(kcontrol); + if (snd_aw2_saa7146_is_using_digital_input(&chip->saa7146)) + ucontrol->value.enumerated.item[0] = CTL_ROUTE_DIGITAL; + else + ucontrol->value.enumerated.item[0] = CTL_ROUTE_ANALOG; + return 0; +} + +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2 *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int is_disgital = + snd_aw2_saa7146_is_using_digital_input(&chip->saa7146); + + if (((ucontrol->value.integer.value[0] == CTL_ROUTE_DIGITAL) + && !is_disgital) + || ((ucontrol->value.integer.value[0] == CTL_ROUTE_ANALOG) + && is_disgital)) { + snd_aw2_saa7146_use_digital_input(&chip->saa7146, !is_disgital); + changed = 1; + } + return changed; +} diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c new file mode 100644 index 000000000000..f20f213489a3 --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.c @@ -0,0 +1,464 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define AW2_SAA7146_M + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "aw2-tsl.h" +#include "saa7146.h" +#include "aw2-saa7146.h" + +#define WRITEREG(value, addr) writel((value), chip->base_addr + (addr)) +#define READREG(addr) readl(chip->base_addr + (addr)) + +static struct snd_aw2_saa7146_cb_param + arr_substream_it_playback_cb[NB_STREAM_PLAYBACK]; +static struct snd_aw2_saa7146_cb_param + arr_substream_it_capture_cb[NB_STREAM_CAPTURE]; + +static int snd_aw2_saa7146_get_limit(int size); + +/* chip-specific destructor */ +int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip) +{ + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* Unset base addr */ + chip->base_addr = NULL; + + return 0; +} + +void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip, + void __iomem *pci_base_addr) +{ + /* set PCI burst/threshold + + Burst length definition + VALUE BURST LENGTH + 000 1 Dword + 001 2 Dwords + 010 4 Dwords + 011 8 Dwords + 100 16 Dwords + 101 32 Dwords + 110 64 Dwords + 111 128 Dwords + + Threshold definition + VALUE WRITE MODE READ MODE + 00 1 Dword of valid data 1 empty Dword + 01 4 Dwords of valid data 4 empty Dwords + 10 8 Dwords of valid data 8 empty Dwords + 11 16 Dwords of valid data 16 empty Dwords */ + + unsigned int acon2; + unsigned int acon1 = 0; + int i; + + /* Set base addr */ + chip->base_addr = pci_base_addr; + + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* enable audio interface */ +#ifdef __BIG_ENDIAN + acon1 |= A1_SWAP; + acon1 |= A2_SWAP; +#endif + /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */ + + /* At initialization WS1 and WS2 are disbaled (configured as input */ + acon1 |= 0 * WS1_CTRL; + acon1 |= 0 * WS2_CTRL; + + /* WS4 is not used. So it must not restart A2. + This is why it is configured as output (force to low) */ + acon1 |= 3 * WS4_CTRL; + + /* WS3_CTRL, WS3_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS3_CTRL; + + /* A1 and A2 are active and asynchronous */ + acon1 |= 3 * AUDIO_MODE; + WRITEREG(acon1, ACON1); + + /* The following comes from original windows driver. + It is needed to have a correct behavior of input and output + simultenously, but I don't know why ! */ + WRITEREG(3 * (BurstA1_in) + 3 * (ThreshA1_in) + + 3 * (BurstA1_out) + 3 * (ThreshA1_out) + + 3 * (BurstA2_out) + 3 * (ThreshA2_out), PCI_BT_A); + + /* enable audio port pins */ + WRITEREG((EAP << 16) | EAP, MC1); + + /* enable I2C */ + WRITEREG((EI2C << 16) | EI2C, MC1); + /* enable interrupts */ + WRITEREG(A1_out | A2_out | A1_in | IIC_S | IIC_E, IER); + + /* audio configuration */ + acon2 = A2_CLKSRC | BCLK1_OEN; + WRITEREG(acon2, ACON2); + + /* By default use analog input */ + snd_aw2_saa7146_use_digital_input(chip, 0); + + /* TSL setup */ + for (i = 0; i < 8; ++i) { + WRITEREG(tsl1[i], TSL1 + (i * 4)); + WRITEREG(tsl2[i], TSL2 + (i * 4)); + } + +} + +void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA2_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA2_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA2_out); + + } else if (stream_number == 1) { + WRITEREG(dw_page, PageA1_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_out); + } else { + printk(KERN_ERR + "aw2: snd_aw2_saa7146_pcm_init_playback: " + "Substream number is not 0 or 1 -> not managed\n"); + } +} + +void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip, + int stream_number, unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA1_in); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_in); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_in); + } else { + printk(KERN_ERR + "aw2: snd_aw2_saa7146_pcm_init_capture: " + "Substream number is not 0 -> not managed\n"); + } +} + +void snd_aw2_saa7146_define_it_playback_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_PLAYBACK) { + arr_substream_it_playback_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_playback_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_define_it_capture_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_CAPTURE) { + arr_substream_it_capture_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_capture_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + unsigned int acon1 = 0; + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + acon1 = READREG(ACON1); + if (stream_number == 0) { + WRITEREG((TR_E_A2_OUT << 16) | TR_E_A2_OUT, MC1); + + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS2_CTRL; + WRITEREG(acon1, ACON1); + + } else if (stream_number == 1) { + WRITEREG((TR_E_A1_OUT << 16) | TR_E_A1_OUT, MC1); + + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 |= 1 * WS1_CTRL; + WRITEREG(acon1, ACON1); + } +} + +void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + unsigned int acon1 = 0; + acon1 = READREG(ACON1); + if (stream_number == 0) { + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 &= ~(3 * WS2_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A2_OUT << 16), MC1); + } else if (stream_number == 1) { + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 &= ~(3 * WS1_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A1_OUT << 16), MC1); + } +} + +void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + if (stream_number == 0) + WRITEREG((TR_E_A1_IN << 16) | TR_E_A1_IN, MC1); +} + +void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + if (stream_number == 0) + WRITEREG((TR_E_A1_IN << 16), MC1); +} + +irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id) +{ + unsigned int isr; + unsigned int iicsta; + struct snd_aw2_saa7146 *chip = dev_id; + + isr = READREG(ISR); + if (!isr) + return IRQ_NONE; + + WRITEREG(isr, ISR); + + if (isr & (IIC_S | IIC_E)) { + iicsta = READREG(IICSTA); + WRITEREG(0x100, IICSTA); + } + + if (isr & A1_out) { + if (arr_substream_it_playback_cb[1].p_it_callback != NULL) { + arr_substream_it_playback_cb[1]. + p_it_callback(arr_substream_it_playback_cb[1]. + p_callback_param); + } + } + if (isr & A2_out) { + if (arr_substream_it_playback_cb[0].p_it_callback != NULL) { + arr_substream_it_playback_cb[0]. + p_it_callback(arr_substream_it_playback_cb[0]. + p_callback_param); + } + + } + if (isr & A1_in) { + if (arr_substream_it_capture_cb[0].p_it_callback != NULL) { + arr_substream_it_capture_cb[0]. + p_it_callback(arr_substream_it_capture_cb[0]. + p_callback_param); + } + } + return IRQ_HANDLED; +} + +unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + long pci_adp = 0; + size_t ptr = 0; + + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP3); + ptr = pci_adp - (long)start_addr; + + if (ptr == buffer_size) + ptr = 0; + } + if (stream_number == 1) { + pci_adp = READREG(PCI_ADP1); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) + ptr = 0; + } + return ptr; +} + +unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + size_t pci_adp = 0; + size_t ptr = 0; + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP2); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) + ptr = 0; + } + return ptr; +} + +void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip, + int use_digital) +{ + /* FIXME: switch between analog and digital input does not always work. + It can produce a kind of white noise. It seams that received data + are inverted sometime (endian inversion). Why ? I don't know, maybe + a problem of synchronization... However for the time being I have + not found the problem. Workaround: switch again (and again) between + digital and analog input until it works. */ + if (use_digital) + WRITEREG(0x40, GPIO_CTRL); + else + WRITEREG(0x50, GPIO_CTRL); +} + +int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146 *chip) +{ + unsigned int reg_val = READREG(GPIO_CTRL); + if ((reg_val & 0xFF) == 0x40) + return 1; + else + return 0; +} + + +static int snd_aw2_saa7146_get_limit(int size) +{ + int limitsize = 32; + int limit = 0; + while (limitsize < size) { + limitsize *= 2; + limit++; + } + return limit; +} diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h new file mode 100644 index 000000000000..5b35e358937f --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.h @@ -0,0 +1,105 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#ifndef AW2_SAA7146_H +#define AW2_SAA7146_H + +#define NB_STREAM_PLAYBACK 2 +#define NB_STREAM_CAPTURE 1 + +#define NUM_STREAM_PLAYBACK_ANA 0 +#define NUM_STREAM_PLAYBACK_DIG 1 + +#define NUM_STREAM_CAPTURE_ANA 0 + +typedef void (*snd_aw2_saa7146_it_cb) (void *); + +struct snd_aw2_saa7146_cb_param { + snd_aw2_saa7146_it_cb p_it_callback; + void *p_callback_param; +}; + +/* definition of the chip-specific record */ + +struct snd_aw2_saa7146 { + void __iomem *base_addr; +}; + +extern void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip, + void __iomem *pci_base_addr); +extern int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip); + +extern void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +extern void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +extern void snd_aw2_saa7146_define_it_playback_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +extern void snd_aw2_saa7146_define_it_capture_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +extern void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146 + *chip, int stream_number); +extern void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 + *chip, int stream_number); + +extern void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146 + *chip, + int stream_number); +extern void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146 + *chip, int stream_number); + +extern irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id); +extern unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146 + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); +extern unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146 + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); + +extern void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip, + int use_digital); + +extern int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146 + *chip); + +#endif diff --git a/sound/pci/aw2/aw2-tsl.h b/sound/pci/aw2/aw2-tsl.h new file mode 100644 index 000000000000..e8afaa0a468a --- /dev/null +++ b/sound/pci/aw2/aw2-tsl.h @@ -0,0 +1,116 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * Copyright 1998 Emagic Soft- und Hardware GmbH + * Copyright 2002 Martijn Sipkema + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define TSL_WS0 (1UL << 31) +#define TSL_WS1 (1UL << 30) +#define TSL_WS2 (1UL << 29) +#define TSL_WS3 (1UL << 28) +#define TSL_WS4 (1UL << 27) +#define TSL_DIS_A1 (1UL << 24) +#define TSL_SDW_A1 (1UL << 23) +#define TSL_SIB_A1 (1UL << 22) +#define TSL_SF_A1 (1UL << 21) +#define TSL_LF_A1 (1UL << 20) +#define TSL_BSEL_A1 (1UL << 17) +#define TSL_DOD_A1 (1UL << 15) +#define TSL_LOW_A1 (1UL << 14) +#define TSL_DIS_A2 (1UL << 11) +#define TSL_SDW_A2 (1UL << 10) +#define TSL_SIB_A2 (1UL << 9) +#define TSL_SF_A2 (1UL << 8) +#define TSL_LF_A2 (1UL << 7) +#define TSL_BSEL_A2 (1UL << 4) +#define TSL_DOD_A2 (1UL << 2) +#define TSL_LOW_A2 (1UL << 1) +#define TSL_EOS (1UL << 0) + + /* Audiowerk8 hardware setup: */ + /* WS0, SD4, TSL1 - Analog/ digital in */ + /* WS1, SD0, TSL1 - Analog out #1, digital out */ + /* WS2, SD2, TSL1 - Analog out #2 */ + /* WS3, SD1, TSL2 - Analog out #3 */ + /* WS4, SD3, TSL2 - Analog out #4 */ + + /* Audiowerk8 timing: */ + /* Timeslot: | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ... */ + + /* A1_INPUT: */ + /* SD4: <_ADC-L_>-------<_ADC-R_>-------< */ + /* WS0: _______________/---------------\_ */ + + /* A1_OUTPUT: */ + /* SD0: <_1-L___>-------<_1-R___>-------< */ + /* WS1: _______________/---------------\_ */ + /* SD2: >-------<_2-L___>-------<_2-R___> */ + /* WS2: -------\_______________/--------- */ + + /* A2_OUTPUT: */ + /* SD1: <_3-L___>-------<_3-R___>-------< */ + /* WS3: _______________/---------------\_ */ + /* SD3: >-------<_4-L___>-------<_4-R___> */ + /* WS4: -------\_______________/--------- */ + +#ifdef __BIG_ENDIAN + /* TODO: not yet implemented */ +#else /* */ + +static int tsl1[8] = { + 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1, + + 1 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 1 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 1 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 | + 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0 | TSL_SF_A1 | TSL_EOS, +}; + +static int tsl2[8] = { + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_LF_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS +}; + +#endif /* */ diff --git a/sound/pci/aw2/saa7146.h b/sound/pci/aw2/saa7146.h new file mode 100644 index 000000000000..ce0ab5f9ee9c --- /dev/null +++ b/sound/pci/aw2/saa7146.h @@ -0,0 +1,168 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +/* SAA7146 registers */ +#define PCI_BT_A 0x4C +#define IICTFR 0x8C +#define IICSTA 0x90 +#define BaseA1_in 0x94 +#define ProtA1_in 0x98 +#define PageA1_in 0x9C +#define BaseA1_out 0xA0 +#define ProtA1_out 0xA4 +#define PageA1_out 0xA8 +#define BaseA2_in 0xAC +#define ProtA2_in 0xB0 +#define PageA2_in 0xB4 +#define BaseA2_out 0xB8 +#define ProtA2_out 0xBC +#define PageA2_out 0xC0 +#define IER 0xDC +#define GPIO_CTRL 0xE0 +#define ACON1 0xF4 +#define ACON2 0xF8 +#define MC1 0xFC +#define MC2 0x100 +#define ISR 0x10C +#define PSR 0x110 +#define SSR 0x114 +#define PCI_ADP1 0x12C +#define PCI_ADP2 0x130 +#define PCI_ADP3 0x134 +#define PCI_ADP4 0x138 +#define LEVEL_REP 0x140 +#define FB_BUFFER1 0x144 +#define FB_BUFFER2 0x148 +#define TSL1 0x180 +#define TSL2 0x1C0 + +#define ME (1UL << 11) +#define LIMIT (1UL << 4) +#define PV (1UL << 3) + +/* PSR/ISR/IER */ +#define PPEF (1UL << 31) +#define PABO (1UL << 30) +#define IIC_S (1UL << 17) +#define IIC_E (1UL << 16) +#define A2_in (1UL << 15) +#define A2_out (1UL << 14) +#define A1_in (1UL << 13) +#define A1_out (1UL << 12) +#define AFOU (1UL << 11) +#define PIN3 (1UL << 6) +#define PIN2 (1UL << 5) +#define PIN1 (1UL << 4) +#define PIN0 (1UL << 3) +#define ECS (1UL << 2) +#define EC3S (1UL << 1) +#define EC0S (1UL << 0) + +/* SSR */ +#define PRQ (1UL << 31) +#define PMA (1UL << 30) +#define IIC_EA (1UL << 21) +#define IIC_EW (1UL << 20) +#define IIC_ER (1UL << 19) +#define IIC_EL (1UL << 18) +#define IIC_EF (1UL << 17) +#define AF2_in (1UL << 10) +#define AF2_out (1UL << 9) +#define AF1_in (1UL << 8) +#define AF1_out (1UL << 7) +#define EC5S (1UL << 3) +#define EC4S (1UL << 2) +#define EC2S (1UL << 1) +#define EC1S (1UL << 0) + +/* PCI_BT_A */ +#define BurstA1_in (1UL << 26) +#define ThreshA1_in (1UL << 24) +#define BurstA1_out (1UL << 18) +#define ThreshA1_out (1UL << 16) +#define BurstA2_in (1UL << 10) +#define ThreshA2_in (1UL << 8) +#define BurstA2_out (1UL << 2) +#define ThreshA2_out (1UL << 0) + +/* MC1 */ +#define MRST_N (1UL << 15) +#define EAP (1UL << 9) +#define EI2C (1UL << 8) +#define TR_E_A2_OUT (1UL << 3) +#define TR_E_A2_IN (1UL << 2) +#define TR_E_A1_OUT (1UL << 1) +#define TR_E_A1_IN (1UL << 0) + +/* MC2 */ +#define UPLD_IIC (1UL << 0) + +/* ACON1 */ +#define AUDIO_MODE (1UL << 29) +#define MAXLEVEL (1UL << 22) +#define A1_SWAP (1UL << 21) +#define A2_SWAP (1UL << 20) +#define WS0_CTRL (1UL << 18) +#define WS0_SYNC (1UL << 16) +#define WS1_CTRL (1UL << 14) +#define WS1_SYNC (1UL << 12) +#define WS2_CTRL (1UL << 10) +#define WS2_SYNC (1UL << 8) +#define WS3_CTRL (1UL << 6) +#define WS3_SYNC (1UL << 4) +#define WS4_CTRL (1UL << 2) +#define WS4_SYNC (1UL << 0) + +/* ACON2 */ +#define A1_CLKSRC (1UL << 27) +#define A2_CLKSRC (1UL << 22) +#define INVERT_BCLK1 (1UL << 21) +#define INVERT_BCLK2 (1UL << 20) +#define BCLK1_OEN (1UL << 19) +#define BCLK2_OEN (1UL << 18) + +/* IICSTA */ +#define IICCC (1UL << 8) +#define ABORT (1UL << 7) +#define SPERR (1UL << 6) +#define APERR (1UL << 5) +#define DTERR (1UL << 4) +#define DRERR (1UL << 3) +#define AL (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +/* IICTFR */ +#define BYTE2 (1UL << 24) +#define BYTE1 (1UL << 16) +#define BYTE0 (1UL << 8) +#define ATRR2 (1UL << 6) +#define ATRR1 (1UL << 4) +#define ATRR0 (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +#define START 3 +#define CONT 2 +#define STOP 1 +#define NOP 0 From 34b6757dc7ce0e9d5d3930b29d53a7bcb0fde047 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Feb 2008 12:12:58 +0100 Subject: [PATCH 048/250] [ALSA] aw2 - Add missing module parameters Added the missing declarations for module parameters. Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-alsa.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 74af639b9132..24e3e4961d9f 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -157,6 +157,13 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Audiowerk2 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); + static struct pci_device_id snd_aw2_ids[] = { {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, From 9e235323db4689e8b7e123252b998a4904806c38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Feb 2008 12:13:29 +0100 Subject: [PATCH 049/250] [ALSA] Add description of aw2 driver Added a brief description of aw2 driver to ALSA-Configuration.txt. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 2cfb8b469c53..8a645a4555f2 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -284,6 +284,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. control correctly. If you have problems regarding this, try another ALSA compliant mixer (alsamixer works). + Module snd-aw2 + -------------- + + Module for Audiowerk2 sound card + + This module supports multiple cards. + Module snd-azt2320 ------------------ From 6876a5323f6169f9321354a398f7364b41ca82fa Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 20 Feb 2008 17:13:44 +0100 Subject: [PATCH 050/250] [ALSA] ASoC: Add support for 12 MHz MCLK in TLV320AIC3X Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 889a897d41ac..e6a730b0dd29 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -660,33 +660,43 @@ struct aic3x_rate_divs { /* AIC3X codec mclk clock divider coefficients */ static const struct aic3x_rate_divs aic3x_divs[] = { /* 8k */ + {12000000, 8000, 48000, 0xa, 16, 3840}, {22579200, 8000, 48000, 0xa, 8, 7075}, {33868800, 8000, 48000, 0xa, 5, 8049}, /* 11.025k */ + {12000000, 11025, 44100, 0x6, 15, 528}, {22579200, 11025, 44100, 0x6, 8, 0}, {33868800, 11025, 44100, 0x6, 5, 3333}, /* 16k */ + {12000000, 16000, 48000, 0x4, 16, 3840}, {22579200, 16000, 48000, 0x4, 8, 7075}, {33868800, 16000, 48000, 0x4, 5, 8049}, /* 22.05k */ + {12000000, 22050, 44100, 0x2, 15, 528}, {22579200, 22050, 44100, 0x2, 8, 0}, {33868800, 22050, 44100, 0x2, 5, 3333}, /* 32k */ + {12000000, 32000, 48000, 0x1, 16, 3840}, {22579200, 32000, 48000, 0x1, 8, 7075}, {33868800, 32000, 48000, 0x1, 5, 8049}, /* 44.1k */ + {12000000, 44100, 44100, 0x0, 15, 528}, {22579200, 44100, 44100, 0x0, 8, 0}, {33868800, 44100, 44100, 0x0, 5, 3333}, /* 48k */ + {12000000, 48000, 48000, 0x0, 16, 3840}, {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ + {12000000, 64000, 96000, 0x1, 16, 3840}, {22579200, 64000, 96000, 0x1, 8, 7075}, {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ + {12000000, 88200, 88200, 0x0, 15, 528}, {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, /* 96k */ + {12000000, 96000, 96000, 0x0, 16, 3840}, {22579200, 96000, 96000, 0x0, 8, 7075}, {33868800, 96000, 96000, 0x0, 5, 8049}, }; @@ -807,6 +817,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, struct aic3x_priv *aic3x = codec->private_data; switch (freq) { + case 12000000: case 22579200: case 33868800: aic3x->sysclk = freq; From 4451089e2aafba87d7574e27c839895131a80293 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 21 Feb 2008 07:49:31 +0100 Subject: [PATCH 051/250] [ALSA] hda: fix STAC927x power management Fix issue on STAC927x codecs that first DAC was getting powered down even if was being used. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 132d1e3eafa5..a31155d41405 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2893,7 +2893,8 @@ static void stac92xx_power_down(struct hda_codec *codec) /* power down inactive DACs */ hda_nid_t *dac; for (dac = spec->dac_list; *dac; dac++) - if (!is_in_dac_nids(spec, *dac)) + if (!is_in_dac_nids(spec, *dac) && + spec->multiout.hp_nid != *dac) snd_hda_codec_write_cache(codec, *dac, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } From 7989fba979daea99285079dc0099ab658e4db759 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 21 Feb 2008 07:50:12 +0100 Subject: [PATCH 052/250] [ALSA] hda: STAC927x invalid association value STAC_DELL_BIOS quirks were setting the association value wrong for port 0x0f, which prevented it from being included in hp_outs[]. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a31155d41405..70c569459758 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3631,7 +3631,7 @@ static int patch_stac927x(struct hda_codec *codec) break; case STAC_DELL_BIOS: /* correct the front output jack as a hp out */ - stac92xx_set_config_reg(codec, 0x0f, 0x02270110); + stac92xx_set_config_reg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); /* fallthru */ From a7662640104599249e15cda7839e9050f92e6a0e Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 21 Feb 2008 07:51:14 +0100 Subject: [PATCH 053/250] [ALSA] hda: 92HDxxxx PCI Quirks Added PCI_QUIRKS for laptop that have the 92HDxxx family of codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 103 +++++++++++++++++++++++++++++++-- 1 file changed, 97 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 70c569459758..a0c290cef763 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -64,11 +64,14 @@ enum { enum { STAC_92HD73XX_REF, + STAC_DELL_M6, STAC_92HD73XX_MODELS }; enum { STAC_92HD71BXX_REF, + STAC_DELL_M4_1, + STAC_DELL_M4_2, STAC_92HD71BXX_MODELS }; @@ -334,10 +337,10 @@ static hda_nid_t stac922x_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x15, 0x1b, }; -static hda_nid_t stac92hd73xx_pin_nids[12] = { +static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x22 + 0x14, 0x1e, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -1220,24 +1223,48 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd73xx_pin_configs[12] = { +static unsigned int ref92hd73xx_pin_configs[13] = { 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, 0x0181302e, 0x01014010, 0x01014020, 0x01014030, 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, + 0x01452050, +}; + +static unsigned int dell_m6_pin_configs[13] = { + 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, + 0x03a11020, 0x03011050, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, }; static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { - [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_DELL_M6] = dell_m6_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = "ref", + [STAC_DELL_M6] = "dell-m6", }; static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD73XX_REF), + "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0256, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0257, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025e, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025f, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271, + "unknown Dell", STAC_DELL_M6), {} /* terminator */ }; @@ -1247,18 +1274,56 @@ static unsigned int ref92hd71bxx_pin_configs[10] = { 0x90a000f0, 0x01452050, }; +static unsigned int dell_m4_1_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, +}; + +static unsigned int dell_m4_2_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, + 0x40f000f0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, + [STAC_DELL_M4_1] = dell_m4_1_pin_configs, + [STAC_DELL_M4_2] = dell_m4_2_pin_configs, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", + [STAC_DELL_M4_1] = "dell-m4-1", + [STAC_DELL_M4_2] = "dell-m4-2", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0250, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024f, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024d, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0251, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0277, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0263, + "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0265, + "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0262, + "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, + "unknown Dell", STAC_DELL_M4_2), {} /* terminator */ }; @@ -3356,13 +3421,39 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); - spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; + switch (spec->board_config) { + case STAC_DELL_M6: + switch (codec->subsystem_id) { + case 0x1028025e: /* Analog Mics */ + case 0x1028025f: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + spec->num_dmics = 0; + break; + case 0x10280254: /* Digital Mics */ + case 0x10280255: + case 0x10280271: + case 0x10280272: + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + case 0x10280256: /* Both */ + case 0x10280057: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + } + break; + default: + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + } + spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; From 03d7ca177fd2ecac8eb22f482f327ecaae4ac8cb Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 21 Feb 2008 07:51:46 +0100 Subject: [PATCH 054/250] [ALSA] hda: STAC927x analog mic Some laptops have a internal analog microphone that is not setup by the BIOS. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a0c290cef763..314ea51538b7 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3721,6 +3721,8 @@ static int patch_stac927x(struct hda_codec *codec) spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + /* configure the analog microphone on some laptops */ + stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ stac92xx_set_config_reg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ From 53463a8302d0c3148c4c64c034312215e76429c2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Ahmet=20=C4=B0nan?= Date: Thu, 21 Feb 2008 07:55:30 +0100 Subject: [PATCH 055/250] [ALSA] snd-dummy - improved timing, silence on prepare MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Ahmet İnan mathematik.uni-freiburg.de> Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index a240eaeb5c62..f67f65829f3e 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -181,10 +181,10 @@ struct snd_dummy_pcm { struct snd_dummy *dummy; spinlock_t lock; struct timer_list timer; - unsigned int pcm_size; - unsigned int pcm_count; + unsigned int pcm_buffer_size; + unsigned int pcm_period_size; unsigned int pcm_bps; /* bytes per second */ - unsigned int pcm_jiffie; /* bytes per one jiffie */ + unsigned int pcm_hz; /* HZ */ unsigned int pcm_irq_pos; /* IRQ position */ unsigned int pcm_buf_pos; /* position in buffer */ struct snd_pcm_substream *substream; @@ -238,11 +238,15 @@ static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream) if (bps <= 0) return -EINVAL; dpcm->pcm_bps = bps; - dpcm->pcm_jiffie = bps / HZ; - dpcm->pcm_size = snd_pcm_lib_buffer_bytes(substream); - dpcm->pcm_count = snd_pcm_lib_period_bytes(substream); + dpcm->pcm_hz = HZ; + dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream); + dpcm->pcm_period_size = snd_pcm_lib_period_bytes(substream); dpcm->pcm_irq_pos = 0; dpcm->pcm_buf_pos = 0; + + snd_pcm_format_set_silence(runtime->format, runtime->dma_area, + bytes_to_samples(runtime, runtime->dma_bytes)); + return 0; } @@ -254,11 +258,11 @@ static void snd_card_dummy_pcm_timer_function(unsigned long data) spin_lock_irqsave(&dpcm->lock, flags); dpcm->timer.expires = 1 + jiffies; add_timer(&dpcm->timer); - dpcm->pcm_irq_pos += dpcm->pcm_jiffie; - dpcm->pcm_buf_pos += dpcm->pcm_jiffie; - dpcm->pcm_buf_pos %= dpcm->pcm_size; - if (dpcm->pcm_irq_pos >= dpcm->pcm_count) { - dpcm->pcm_irq_pos %= dpcm->pcm_count; + dpcm->pcm_irq_pos += dpcm->pcm_bps; + if (dpcm->pcm_irq_pos >= dpcm->pcm_period_size * dpcm->pcm_hz) { + dpcm->pcm_irq_pos %= dpcm->pcm_period_size * dpcm->pcm_hz; + dpcm->pcm_buf_pos += dpcm->pcm_period_size; + dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size; spin_unlock_irqrestore(&dpcm->lock, flags); snd_pcm_period_elapsed(dpcm->substream); } else From 87218e9c6e7f7908baf98030b6d724e14aa8b5cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Feb 2008 08:13:11 +0100 Subject: [PATCH 056/250] [ALSA] hda-intel - Use PCI_DEVICE() macro Clean up the pci id table using PCI_DEVICE() macro. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 98 +++++++++++++++++++++------------------ 1 file changed, 53 insertions(+), 45 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cf1a1d0124fe..c495ca012941 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2017,51 +2017,59 @@ static void __devexit azx_remove(struct pci_dev *pci) /* PCI IDs */ static struct pci_device_id azx_ids[] = { - { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH6 */ - { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH7 */ - { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ESB2 */ - { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */ - { 0x8086, 0x293e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ - { 0x8086, 0x293f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ - { 0x8086, 0x3a3e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ - { 0x8086, 0x3a6e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ - { 0x8086, 0x811b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SCH }, /* SCH*/ - { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ - { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ - { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ - { 0x1002, 0x7919, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS690 HDMI */ - { 0x1002, 0x960f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */ - { 0x1002, 0xaa00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI R600 HDMI */ - { 0x1002, 0xaa08, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV630 HDMI */ - { 0x1002, 0xaa10, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV610 HDMI */ - { 0x1002, 0xaa18, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV670 HDMI */ - { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */ - { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */ - { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */ - { 0x1002, 0xaa38, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV730 HDMI */ - { 0x1002, 0xaa40, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV710 HDMI */ - { 0x1002, 0xaa48, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV740 HDMI */ - { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ - { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ - { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ - { 0x10de, 0x026c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP51 */ - { 0x10de, 0x0371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP55 */ - { 0x10de, 0x03e4, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP61 */ - { 0x10de, 0x03f0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP61 */ - { 0x10de, 0x044a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ - { 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ - { 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ - { 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ - { 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ - { 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ - { 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0ac0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ - { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ - { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ - { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + /* ICH 6..10 */ + { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, + /* SCH */ + { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, + /* ATI SB 450/600 */ + { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI }, + { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI }, + /* ATI HDMI */ + { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa18), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa20), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa28), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa30), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa38), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa40), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI }, + /* VIA VT8251/VT8237A */ + { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, + /* SIS966 */ + { PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS }, + /* ULI M5461 */ + { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI }, + /* NVIDIA MCP */ + { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); From c354cd7d9627930dcfbcff8355d422fa1bca948a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Feb 2008 12:40:00 +0100 Subject: [PATCH 057/250] [ALSA] seq-oss - Remove invalid BUG() Removed invalid BUG() - the driver should handle the error case properly rather than issuing BUG(). Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_synth.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index ab570a0a6183..558dadbf45f1 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -245,8 +245,13 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) info->nr_voices = rec->nr_voices; if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); - if (!info->ch) - BUG(); + if (!info->ch) { + snd_printk(KERN_ERR "Cannot malloc\n"); + rec->oper.close(&info->arg); + module_put(rec->oper.owner); + snd_use_lock_free(&rec->use_lock); + continue; + } reset_channels(info); } debug_printk(("synth %d assigned\n", i)); From 88d18ea2c2b40496b56efcb354e9eae1f09ef126 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Feb 2008 14:11:09 +0100 Subject: [PATCH 058/250] [ALSA] hda-codec - Add missing descriptions for STAC codec models Added the missing descriptions for STAC codec models. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8a645a4555f2..999b7048162a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1030,6 +1030,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack D965 3stack 5stack D965 5stack + SPDIF dell-3stack Dell Dimension E520 + dell-bios Fixes with Dell BIOS setup + + STAC92HD71B* + ref Reference board + dell-m4-1 Dell desktops + dell-m4-2 Dell desktops + + STAC92HD73* + ref Reference board + dell-m6 Dell desktops STAC9872 vaio Setup for VAIO FE550G/SZ110 From ae0afd81b34ce287ffda7dd4e33b5144de2ad39d Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 22 Feb 2008 17:55:05 +0100 Subject: [PATCH 059/250] [ALSA] hda: Mic as output fix Added logic to check if AUTO_PIN_FRONT_MIC is available for output switch, if AUTO_PIN_MIC isn't. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 58 +++++++++++++++++++--------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 314ea51538b7..ef86402d7e67 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2307,6 +2307,29 @@ static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_ return 0; } +static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else if (spec->multiout.num_dacs > 4) { + printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); + return 1; + } else { + spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; + spec->multiout.num_dacs++; + } + return 0; +} + +static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (is_in_dac_nids(spec, nid)) + return 1; + if (spec->multiout.hp_nid == nid) + return 1; + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -2369,10 +2392,11 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, if (spec->mic_switch) { unsigned int def_conf; - nid = cfg->input_pins[AUTO_PIN_MIC]; + unsigned int mic_pin = AUTO_PIN_MIC; +again: + nid = cfg->input_pins[mic_pin]; def_conf = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { @@ -2382,38 +2406,22 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (nid << 8) | 1); + nid = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (!check_in_dac_nids(spec, nid)) + add_spec_dacs(spec, nid); if (err < 0) return err; } + } else if (mic_pin == AUTO_PIN_MIC) { + mic_pin = AUTO_PIN_FRONT_MIC; + goto again; } } return 0; } -static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (is_in_dac_nids(spec, nid)) - return 1; - if (spec->multiout.hp_nid == nid) - return 1; - return 0; -} - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else if (spec->multiout.num_dacs > 4) { - printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); - return 1; - } else { - spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; - spec->multiout.num_dacs++; - } - return 0; -} - /* add playback controls for Speaker and HP outputs */ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) From 47ba97f8fa01bb5b48e73b4b1271fbf1436a2d4b Mon Sep 17 00:00:00 2001 From: Remy Bruno Date: Fri, 22 Feb 2008 17:57:02 +0100 Subject: [PATCH 060/250] [ALSA] hdsp - RME 9632 fix at 192kHz The bits indicating SPDIF frequency in the status register are not the same for the 9632 than for the other cards, because it also supports 192kHz. A specific bitmask has thus been added (used in hdsp_spdif_sample_rate()). The 9632 does not seem to report external sample rates greater than 96kHz. In this case, the best seems to report spdif rate when autosync reference is spdif. This also required to move function hdsp_spdif_sample_rate(). Signed-off-by: Remy Bruno Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 54 ++++++++++++++++++++++++++-------------- 1 file changed, 36 insertions(+), 18 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 1be84f22d0de..4d6fbb36ab8a 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -318,6 +318,10 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) +#define HDSP_spdifFrequencyMask_9632 (HDSP_spdifFrequency0|\ + HDSP_spdifFrequency1|\ + HDSP_spdifFrequency2|\ + HDSP_spdifFrequency3) #define HDSP_spdifFrequency32KHz (HDSP_spdifFrequency0) #define HDSP_spdifFrequency44_1KHz (HDSP_spdifFrequency1) @@ -328,7 +332,9 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_spdifFrequency96KHz (HDSP_spdifFrequency2|HDSP_spdifFrequency1) /* This is for H9632 cards */ -#define HDSP_spdifFrequency128KHz HDSP_spdifFrequencyMask +#define HDSP_spdifFrequency128KHz (HDSP_spdifFrequency0|\ + HDSP_spdifFrequency1|\ + HDSP_spdifFrequency2) #define HDSP_spdifFrequency176_4KHz HDSP_spdifFrequency3 #define HDSP_spdifFrequency192KHz (HDSP_spdifFrequency3|HDSP_spdifFrequency0) @@ -885,28 +891,15 @@ static int snd_hdsp_use_is_exclusive(struct hdsp *hdsp) return ret; } -static int hdsp_external_sample_rate (struct hdsp *hdsp) -{ - unsigned int status2 = hdsp_read(hdsp, HDSP_status2Register); - unsigned int rate_bits = status2 & HDSP_systemFrequencyMask; - - switch (rate_bits) { - case HDSP_systemFrequency32: return 32000; - case HDSP_systemFrequency44_1: return 44100; - case HDSP_systemFrequency48: return 48000; - case HDSP_systemFrequency64: return 64000; - case HDSP_systemFrequency88_2: return 88200; - case HDSP_systemFrequency96: return 96000; - default: - return 0; - } -} - static int hdsp_spdif_sample_rate(struct hdsp *hdsp) { unsigned int status = hdsp_read(hdsp, HDSP_statusRegister); unsigned int rate_bits = (status & HDSP_spdifFrequencyMask); + /* For the 9632, the mask is different */ + if (hdsp->io_type == H9632) + rate_bits = (status & HDSP_spdifFrequencyMask_9632); + if (status & HDSP_SPDIFErrorFlag) return 0; @@ -933,6 +926,31 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) return 0; } +static int hdsp_external_sample_rate(struct hdsp *hdsp) +{ + unsigned int status2 = hdsp_read(hdsp, HDSP_status2Register); + unsigned int rate_bits = status2 & HDSP_systemFrequencyMask; + + /* For the 9632 card, there seems to be no bit for indicating external + * sample rate greater than 96kHz. The card reports the corresponding + * single speed. So the best means seems to get spdif rate when + * autosync reference is spdif */ + if (hdsp->io_type == H9632 && + hdsp_autosync_ref(hdsp) == HDSP_AUTOSYNC_FROM_SPDIF) + return hdsp_spdif_sample_rate(hdsp); + + switch (rate_bits) { + case HDSP_systemFrequency32: return 32000; + case HDSP_systemFrequency44_1: return 44100; + case HDSP_systemFrequency48: return 48000; + case HDSP_systemFrequency64: return 64000; + case HDSP_systemFrequency88_2: return 88200; + case HDSP_systemFrequency96: return 96000; + default: + return 0; + } +} + static void hdsp_compute_period_size(struct hdsp *hdsp) { hdsp->period_bytes = 1 << ((hdsp_decode_latency(hdsp->control_register) + 8)); From ea6b5828cdbbedaf26b12ae64befbec18084ea3c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 Feb 2008 10:59:52 +0100 Subject: [PATCH 061/250] [ALSA] mpu401: reduce tx loop timeout Reduce the number of times to check for a non-empty Tx FIFO from 100 to 2 because there is no MPU-401 implementation that needs more than one or two reads to determine the actual FIFO status. Signed-off-by: Clemens Ladisch --- sound/drivers/mpu401/mpu401_uart.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 5993864acbd3..dd6ec4266732 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -425,16 +425,17 @@ static void snd_mpu401_uart_input_read(struct snd_mpu401 * mpu) static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu) { unsigned char byte; - int max = 256, timeout; + int max = 256; do { if (snd_rawmidi_transmit_peek(mpu->substream_output, &byte, 1) == 1) { - for (timeout = 100; timeout > 0; timeout--) { - if (snd_mpu401_output_ready(mpu)) - break; - } - if (timeout == 0) + /* + * Try twice because there is hardware that insists on + * setting the output busy bit after each write. + */ + if (!snd_mpu401_output_ready(mpu) && + !snd_mpu401_output_ready(mpu)) break; /* Tx FIFO full - try again later */ mpu->write(mpu, byte, MPU401D(mpu)); snd_rawmidi_transmit_ack(mpu->substream_output, 1); From 25a47b6b01314f027553d231c1a67dee27ff02b0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 Feb 2008 11:04:19 +0100 Subject: [PATCH 062/250] [ALSA] usb-audio: sort quirks list Move some entries to their proper position so that the list is again sorted by vendor/product ID. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 75 ++++++++++++++++++++++--------------------- 1 file changed, 38 insertions(+), 37 deletions(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 938dff5f9cef..82a8d14c26af 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -39,6 +39,30 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* Creative/E-Mu devices */ +{ + USB_DEVICE(0x041e, 0x3010), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Creative Labs", + .product_name = "Sound Blaster MP3+", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ + /* E-Mu 0202 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f02, + .bInterfaceClass = USB_CLASS_AUDIO, +}, +{ + /* E-Mu 0404 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f04, + .bInterfaceClass = USB_CLASS_AUDIO, +}, + /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. @@ -97,19 +121,7 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, -/* E-Mu devices */ -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x041e, - .idProduct = 0x3f02, - .bInterfaceClass = USB_CLASS_AUDIO, -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x041e, - .idProduct = 0x3f04, - .bInterfaceClass = USB_CLASS_AUDIO, -}, + /* * Yamaha devices */ @@ -1165,19 +1177,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -{ - USB_DEVICE(0x582, 0x00a6), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "Juno-G", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, { /* * This quirk is for the "Advanced" modes of the Edirol UA-25. * If the switch is not in an advanced setting, the UA-25 has @@ -1335,6 +1334,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, /* TODO: add Edirol MD-P1 support */ +{ + USB_DEVICE(0x582, 0x00a6), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "Juno-G", + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + } +}, { /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), @@ -1719,17 +1731,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -{ - /* Creative Sound Blaster MP3+ */ - USB_DEVICE(0x041e, 0x3010), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Creative Labs", - .product_name = "Sound Blaster MP3+", - .ifnum = QUIRK_NO_INTERFACE - } - -}, - /* Emagic devices */ { USB_DEVICE(0x086a, 0x0001), From aea7bb0a6ff5e751ef611ba9c1146c3c8489f25e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Feb 2008 18:26:41 +0100 Subject: [PATCH 063/250] [ALSA] hda-codec - Fix initial DAC numbers of 92HD71bxx codecs Fix the initial num_dacs of 92HD71bxx codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ef86402d7e67..4bc7d8646fab 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -246,7 +246,7 @@ static hda_nid_t stac92hd71bxx_dmux_nids[1] = { 0x1c, }; -static hda_nid_t stac92hd71bxx_dac_nids[2] = { +static hda_nid_t stac92hd71bxx_dac_nids[1] = { 0x10, /*0x11, */ }; @@ -3550,7 +3550,7 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 2; + spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; From b26451c059e741ec5e3389f7758627cb094b3766 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Feb 2008 11:56:35 +0100 Subject: [PATCH 064/250] [ALSA] hda-codec - Add docking-station mic input for Thinkpad X61 Added the docking-stationc mic input to the capture source list for Thinkpad X61. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e0f3559f8b13..20446e320b2c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3188,11 +3188,12 @@ static int patch_ad1884(struct hda_codec *codec) * Lenovo Thinkpad T61/X61 */ static struct hda_input_mux ad1984_thinkpad_capture_source = { - .num_items = 3, + .num_items = 4, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, { "Mix", 0x3 }, + { "Docking-Station", 0x4 }, }, }; From 964a788e0ba64aa4ce2e6488718f3ee28cc2e61e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Feb 2008 13:16:08 +0100 Subject: [PATCH 065/250] [ALSA] soc - Report errors from snd_soc_dapm_set_endpoint() Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 620d7ea3c15f..abac6847ecad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1334,10 +1334,11 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, list_for_each_entry(w, &codec->dapm_widgets, list) { if (!strcmp(w->name, endpoint)) { w->connected = status; + return 0; } } - return 0; + return -ENODEV; } EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); From 7dfa31ed5e1fc0ace7f1959b9564ad43d78fd7af Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:52:17 +0100 Subject: [PATCH 066/250] [ALSA] sound: cmipci.c fix shadowed variable warning A temporary variable for each mixer element is used in an initialization loop, use the name elem_id. sound/pci/cmipci.c:2747:26: warning: symbol 'id' shadows an earlier one sound/pci/cmipci.c:56:13: originally declared here [tiwai - fixed a coding style issue as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 135f30860753..4074584a7d90 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2744,12 +2744,13 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic } for (idx = 0; idx < CM_SAVED_MIXERS; idx++) { - struct snd_ctl_elem_id id; + struct snd_ctl_elem_id elem_id; struct snd_kcontrol *ctl; - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, cm_saved_mixer[idx].name); - if ((ctl = snd_ctl_find_id(cm->card, &id)) != NULL) + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, cm_saved_mixer[idx].name); + ctl = snd_ctl_find_id(cm->card, &elem_id); + if (ctl) cm->mixer_res_ctl[idx] = ctl; } From 405b0a377cfe3750f4af54b80d0402c3fe777b87 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:53:07 +0100 Subject: [PATCH 067/250] [ALSA] sound: ens1370.c fix shadowed variable warning index is incremented only when AC97_EI_SPDIF and then assigned to the index field. Change the temporary name to is_spdif. sound/pci/ens1370.c:1638:10: warning: symbol 'index' shadows an earlier one sound/pci/ens1370.c:84:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 72d85a5ae6a0..52fae4a7cfdd 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1635,20 +1635,20 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, if (has_spdif > 0 || (!has_spdif && es1371_quirk_lookup(ensoniq, es1371_spdif_present))) { struct snd_kcontrol *kctl; - int i, index = 0; + int i, is_spdif = 0; ensoniq->spdif_default = ensoniq->spdif_stream = SNDRV_PCM_DEFAULT_CON_SPDIF; outl(ensoniq->spdif_default, ES_REG(ensoniq, CHANNEL_STATUS)); if (ensoniq->u.es1371.ac97->ext_id & AC97_EI_SPDIF) - index++; + is_spdif++; for (i = 0; i < ARRAY_SIZE(snd_es1371_mixer_spdif); i++) { kctl = snd_ctl_new1(&snd_es1371_mixer_spdif[i], ensoniq); if (!kctl) return -ENOMEM; - kctl->id.index = index; + kctl->id.index = is_spdif; err = snd_ctl_add(card, kctl); if (err < 0) return err; From 3463d8fa14ba2e00ede9894efdaa65189eb04b36 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:53:41 +0100 Subject: [PATCH 068/250] [ALSA] sound: es1968.c fox shadowed variable warning id is used when initializing the mixer elements, use elem_id here instead. sound/pci/es1968.c:1963:25: warning: symbol 'id' shadows an earlier one sound/pci/es1968.c:129:13: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 7d911a18c082..67f03264f871 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1972,7 +1972,7 @@ snd_es1968_mixer(struct es1968 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; - struct snd_ctl_elem_id id; + struct snd_ctl_elem_id elem_id; int err; static struct snd_ac97_bus_ops ops = { .write = snd_es1968_ac97_write, @@ -1989,14 +1989,14 @@ snd_es1968_mixer(struct es1968 *chip) return err; /* attach master switch / volumes for h/w volume control */ - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, "Master Playback Switch"); - chip->master_switch = snd_ctl_find_id(chip->card, &id); - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, "Master Playback Volume"); - chip->master_volume = snd_ctl_find_id(chip->card, &id); + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, "Master Playback Switch"); + chip->master_switch = snd_ctl_find_id(chip->card, &elem_id); + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, "Master Playback Volume"); + chip->master_volume = snd_ctl_find_id(chip->card, &elem_id); return 0; } From 58e4334e82c0f4eb0147a905a127bd14f0ea0a2d Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:55:07 +0100 Subject: [PATCH 069/250] [ALSA] sound: fm801.c fix shadowed variable warning id was only used as a counter in a for loop, move the declaration to where it is used and change it to i. sound/pci/fm801.c:1288:6: warning: symbol 'id' shadows an earlier one sound/pci/fm801.c:51:13: originally declared here [tiwai - fixed a coding style issue as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4c300e6149fc..c129f9e2072c 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1285,7 +1285,6 @@ static int wait_for_codec(struct fm801 *chip, unsigned int codec_id, static int snd_fm801_chip_init(struct fm801 *chip, int resume) { - int id; unsigned short cmdw; if (chip->tea575x_tuner & 0x0010) @@ -1310,13 +1309,14 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) } else { /* my card has the secondary codec */ /* at address #3, so the loop is inverted */ - for (id = 3; id > 0; id--) { - if (! wait_for_codec(chip, id, AC97_VENDOR_ID1, + int i; + for (i = 3; i > 0; i--) { + if (!wait_for_codec(chip, i, AC97_VENDOR_ID1, msecs_to_jiffies(50))) { cmdw = inw(FM801_REG(chip, AC97_DATA)); if (cmdw != 0xffff && cmdw != 0) { chip->secondary = 1; - chip->secondary_addr = id; + chip->secondary_addr = i; break; } } From e37273d3484e241063fefb2611a0c93eb0d9ddbd Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:56:37 +0100 Subject: [PATCH 070/250] [ALSA] sound: maestro3.c fix shadowed variable warnings change id to elem_id as it is used to initialize each mixer element sound/pci/maestro3.c:2071:25: warning: symbol 'id' shadows an earlier one sound/pci/maestro3.c:67:13: originally declared here index is used in each of these places to count over the dsp's memory, change to the name dsp_index sound/pci/maestro3.c:2572:9: warning: symbol 'index' shadows an earlier one sound/pci/maestro3.c:66:12: originally declared here sound/pci/maestro3.c:2604:9: warning: symbol 'index' shadows an earlier one sound/pci/maestro3.c:66:12: originally declared here [tiwai - fixed coding style issues as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 04fa0a68416c..a753dae65ab6 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2068,7 +2068,7 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; - struct snd_ctl_elem_id id; + struct snd_ctl_elem_id elem_id; int err; static struct snd_ac97_bus_ops ops = { .write = snd_m3_ac97_write, @@ -2088,14 +2088,14 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) schedule_timeout_uninterruptible(msecs_to_jiffies(100)); snd_ac97_write(chip->ac97, AC97_PCM, 0); - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, "Master Playback Switch"); - chip->master_switch = snd_ctl_find_id(chip->card, &id); - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, "Master Playback Volume"); - chip->master_volume = snd_ctl_find_id(chip->card, &id); + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, "Master Playback Switch"); + chip->master_switch = snd_ctl_find_id(chip->card, &elem_id); + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, "Master Playback Volume"); + chip->master_volume = snd_ctl_find_id(chip->card, &elem_id); return 0; } @@ -2569,7 +2569,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_m3 *chip = card->private_data; - int i, index; + int i, dsp_index; if (chip->suspend_mem == NULL) return 0; @@ -2583,12 +2583,12 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) snd_m3_assp_halt(chip); /* save dsp image */ - index = 0; + dsp_index = 0; for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++) - chip->suspend_mem[index++] = + chip->suspend_mem[dsp_index++] = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_CODE, i); for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++) - chip->suspend_mem[index++] = + chip->suspend_mem[dsp_index++] = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA, i); pci_disable_device(pci); @@ -2601,7 +2601,7 @@ static int m3_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_m3 *chip = card->private_data; - int i, index; + int i, dsp_index; if (chip->suspend_mem == NULL) return 0; @@ -2625,13 +2625,13 @@ static int m3_resume(struct pci_dev *pci) snd_m3_ac97_reset(chip); /* restore dsp image */ - index = 0; + dsp_index = 0; for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++) snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, i, - chip->suspend_mem[index++]); + chip->suspend_mem[dsp_index++]); for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++) snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, i, - chip->suspend_mem[index++]); + chip->suspend_mem[dsp_index++]); /* tell the dma engine to restart itself */ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, From 4db9e4f2b5278338ff9487eefdc8e32109aa0552 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:57:23 +0100 Subject: [PATCH 071/250] [ALSA] sound: rme32.c fix integer as NULL pointer warning kernel style does assignment outside of if() statements. sound/pci/rme32.c:1353:71: warning: Using plain integer as NULL pointer Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/rme32.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index df184aabce84..e7ef3a1a25a8 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1350,7 +1350,8 @@ static int __devinit snd_rme32_create(struct rme32 * rme32) return err; rme32->port = pci_resource_start(rme32->pci, 0); - if ((rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE)) == 0) { + rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE); + if (!rme32->iobase) { snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme32->port, rme32->port + RME32_IO_SIZE - 1); return -ENOMEM; From 44977b719f7425ddb1cb67d647a4f588a9718163 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:57:47 +0100 Subject: [PATCH 072/250] [ALSA] sound: rme96.c fix integer as NULL pointer warning kernel style does assignment outside of if() block sound/pci/rme96.c:1562:71: warning: Using plain integer as NULL pointer Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/rme96.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index fb0a4ee8bc02..3fdd488d0975 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1559,7 +1559,8 @@ snd_rme96_create(struct rme96 *rme96) return err; rme96->port = pci_resource_start(rme96->pci, 0); - if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) { + rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE); + if (!rme96->iobase) { snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1); return -ENOMEM; } From 608b10bad3563e2349393136ce421d9f67329170 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 11:58:18 +0100 Subject: [PATCH 073/250] [ALSA] sound: ac97_pcm.c fix shadowed variable warning err is always assigned before it is used, no need to declare another inside the if statement. sound/pci/ac97/ac97_pcm.c:577:7: warning: symbol 'err' shadows an earlier one sound/pci/ac97/ac97_pcm.c:572:6: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 3674f35c4a79..48cbda9378c5 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -574,7 +574,6 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, r = rate > 48000; bus = pcm->bus; if (cfg == AC97_PCM_CFG_SPDIF) { - int err; for (cidx = 0; cidx < 4; cidx++) if (bus->codec[cidx] && (bus->codec[cidx]->ext_id & AC97_EI_SPDIF)) { err = set_spdif_rate(bus->codec[cidx], rate); From c74056d437401dc7d43970cd845c34a7e28723c0 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 12:00:48 +0100 Subject: [PATCH 074/250] [ALSA] sound: ali5451.c fix shadowed variable warnings enable is used to test for whether or not spdif should be enabled, change to spdif_enable. sound/pci/ali5451/ali5451.c:1812:15: warning: symbol 'enable' shadows an earlier one sound/pci/ali5451/ali5451.c:63:12: originally declared here sound/pci/ali5451/ali5451.c:1840:27: warning: symbol 'enable' shadows an earlier one sound/pci/ali5451/ali5451.c:63:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 6a905ed9cbd6..fc04d3da1af5 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1809,26 +1809,26 @@ static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ali *codec = kcontrol->private_data; - unsigned int enable; + unsigned int spdif_enable; - enable = ucontrol->value.integer.value[0] ? 1 : 0; + spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0; spin_lock_irq(&codec->reg_lock); switch (kcontrol->private_value) { case 0: - enable = (codec->spdif_mask & 0x02) ? 1 : 0; + spdif_enable = (codec->spdif_mask & 0x02) ? 1 : 0; break; case 1: - enable = ((codec->spdif_mask & 0x02) && + spdif_enable = ((codec->spdif_mask & 0x02) && (codec->spdif_mask & 0x04)) ? 1 : 0; break; case 2: - enable = (codec->spdif_mask & 0x01) ? 1 : 0; + spdif_enable = (codec->spdif_mask & 0x01) ? 1 : 0; break; default: break; } - ucontrol->value.integer.value[0] = enable; + ucontrol->value.integer.value[0] = spdif_enable; spin_unlock_irq(&codec->reg_lock); return 0; } @@ -1837,17 +1837,17 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ali *codec = kcontrol->private_data; - unsigned int change = 0, enable = 0; + unsigned int change = 0, spdif_enable = 0; - enable = ucontrol->value.integer.value[0] ? 1 : 0; + spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0; spin_lock_irq(&codec->reg_lock); switch (kcontrol->private_value) { case 0: change = (codec->spdif_mask & 0x02) ? 1 : 0; - change = change ^ enable; + change = change ^ spdif_enable; if (change) { - if (enable) { + if (spdif_enable) { codec->spdif_mask |= 0x02; snd_ali_enable_spdif_out(codec); } else { @@ -1859,9 +1859,9 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, break; case 1: change = (codec->spdif_mask & 0x04) ? 1 : 0; - change = change ^ enable; + change = change ^ spdif_enable; if (change && (codec->spdif_mask & 0x02)) { - if (enable) { + if (spdif_enable) { codec->spdif_mask |= 0x04; snd_ali_enable_spdif_chnout(codec); } else { @@ -1872,9 +1872,9 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, break; case 2: change = (codec->spdif_mask & 0x01) ? 1 : 0; - change = change ^ enable; + change = change ^ spdif_enable; if (change) { - if (enable) { + if (spdif_enable) { codec->spdif_mask |= 0x01; snd_ali_enable_spdif_in(codec); } else { From d967a02712f09265b3c357f35f125715f5dffd2f Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 12:02:22 +0100 Subject: [PATCH 075/250] [ALSA] sound: ca0106_main.c fix shadowed variable warnings change to intr_enable as per the two functions it is defined in. sound/pci/ca0106/ca0106_main.c:438:15: warning: symbol 'enable' shadows an earlier one sound/pci/ca0106/ca0106_main.c:159:12: originally declared here sound/pci/ca0106/ca0106_main.c:449:15: warning: symbol 'enable' shadows an earlier one sound/pci/ca0106/ca0106_main.c:159:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 176e0f0e8058..3818249fcc81 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -435,22 +435,22 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) | intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) | intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) & ~intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) & ~intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } From bed515b0dfdcf8f440c7e6c5bad8ce3eb96fb625 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 28 Feb 2008 12:02:56 +0100 Subject: [PATCH 076/250] [ALSA] sound: ca0106_mixer.c fix shadowed variable warnings Change the variable err to _err within the ADD_CTLS macro to avoid shadowing the local variable. sound/pci/ca0106/ca0106_mixer.c:710:2: warning: symbol 'err' shadows an earlier one sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here sound/pci/ca0106/ca0106_mixer.c:712:3: warning: symbol 'err' shadows an earlier one sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here sound/pci/ca0106/ca0106_mixer.c:721:3: warning: symbol 'err' shadows an earlier one sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_mixer.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 94618ecaab65..3025ed1b6e1e 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -650,11 +650,11 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch #define ADD_CTLS(emu, ctls) \ do { \ - int i, err; \ + int i, _err; \ for (i = 0; i < ARRAY_SIZE(ctls); i++) { \ - err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \ - if (err < 0) \ - return err; \ + _err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \ + if (_err < 0) \ + return _err; \ } \ } while (0) From 470f23b873679b045908551302fec6b1edf05a5a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Ahmet=20=C4=B0nan?= Date: Thu, 28 Feb 2008 12:46:32 +0100 Subject: [PATCH 077/250] [ALSA] snd-dummy - better realtime app support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit when the time interval for a period is smaller than kernel HZ, then snd-aloop and snd-dummy cannot call snd_pcm_period_elapsed as fast enough annymore. this happens for example with games. but the app still needs to see, that the buffer actually did go further, which is provided by these patches. Signed-off-by: Ahmet İnan mathematik.uni-freiburg.de> Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index f67f65829f3e..83ac4980c5fb 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -259,10 +259,10 @@ static void snd_card_dummy_pcm_timer_function(unsigned long data) dpcm->timer.expires = 1 + jiffies; add_timer(&dpcm->timer); dpcm->pcm_irq_pos += dpcm->pcm_bps; + dpcm->pcm_buf_pos += dpcm->pcm_bps; + dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size * dpcm->pcm_hz; if (dpcm->pcm_irq_pos >= dpcm->pcm_period_size * dpcm->pcm_hz) { dpcm->pcm_irq_pos %= dpcm->pcm_period_size * dpcm->pcm_hz; - dpcm->pcm_buf_pos += dpcm->pcm_period_size; - dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size; spin_unlock_irqrestore(&dpcm->lock, flags); snd_pcm_period_elapsed(dpcm->substream); } else @@ -274,7 +274,7 @@ static snd_pcm_uframes_t snd_card_dummy_pcm_pointer(struct snd_pcm_substream *su struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dummy_pcm *dpcm = runtime->private_data; - return bytes_to_frames(runtime, dpcm->pcm_buf_pos); + return bytes_to_frames(runtime, dpcm->pcm_buf_pos / dpcm->pcm_hz); } static struct snd_pcm_hardware snd_card_dummy_playback = From 3fa4a9073886a1031400c19e8b09fca3eebb645f Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:41:56 +0100 Subject: [PATCH 078/250] [ALSA] sound: au88x0_pcm.c fix integer as NULL pointer warning sound/pci/au88x0/au88x0_pcm.c:508:15: warning: Using plain integer as NULL pointer Also some small codingstyle fixes. Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 526c6c5ecf7b..f9a58b4a30eb 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -498,14 +498,14 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = { }; /* create a pcm device */ -static int __devinit snd_vortex_new_pcm(vortex_t * chip, int idx, int nr) +static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) { struct snd_pcm *pcm; struct snd_kcontrol *kctl; int i; int err, nr_capt; - if ((chip == 0) || (idx < 0) || (idx >= VORTEX_PCM_LAST)) + if (!chip || idx < 0 || idx >= VORTEX_PCM_LAST) return -ENODEV; /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the @@ -514,9 +514,9 @@ static int __devinit snd_vortex_new_pcm(vortex_t * chip, int idx, int nr) nr_capt = nr; else nr_capt = 0; - if ((err = - snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr, - nr_capt, &pcm)) < 0) + err = snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr, + nr_capt, &pcm); + if (err < 0) return err; strcpy(pcm->name, vortex_pcm_name[idx]); chip->pcm[idx] = pcm; From 4677df07e551d64167f64eba5e3563b3df7f4ca8 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:44:26 +0100 Subject: [PATCH 079/250] [ALSA] sound: emuproc.c fix signedness warning Reading regs from the fpga into an int instead of a u32, trivial fix. sound/pci/emu10k1/emuproc.c:422:34: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emuproc.c:422:34: expected unsigned int [usertype] *value sound/pci/emu10k1/emuproc.c:422:34: got int * Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emuproc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index f3caa3f890c6..216f9748aff5 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -412,7 +412,7 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; - int value; + u32 value; unsigned long flags; int i; snd_iprintf(buffer, "EMU1010 Registers:\n\n"); From f2948fc2f0e1c19b8bea77a14338d338e941ac9a Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:44:57 +0100 Subject: [PATCH 080/250] [ALSA] sound: emu10k1x.c fix shadowed variable warnings enable in these contexts refers specifically to intr enable, as per the two functions it is found in. Use intr_enable instead. sound/pci/emu10k1/emu10k1x.c:330:15: warning: symbol 'enable' shadows an earlier one sound/pci/emu10k1/emu10k1x.c:53:12: originally declared here sound/pci/emu10k1/emu10k1x.c:341:15: warning: symbol 'enable' shadows an earlier one sound/pci/emu10k1/emu10k1x.c:53:12: originally declared here instead of shadowing, use cap_voice as we test for the capture voice in this statement. sound/pci/emu10k1/emu10k1x.c:798:25: warning: symbol 'pvoice' shadows an earlier one sound/pci/emu10k1/emu10k1x.c:787:24: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1x.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 5512abd98bd9..341f34e19f3c 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -327,22 +327,22 @@ static void snd_emu10k1x_ptr_write(struct emu10k1x *emu, static void snd_emu10k1x_intr_enable(struct emu10k1x *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) | intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) | intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } static void snd_emu10k1x_intr_disable(struct emu10k1x *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) & ~intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) & ~intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -795,9 +795,9 @@ static irqreturn_t snd_emu10k1x_interrupt(int irq, void *dev_id) // capture interrupt if (status & (IPR_CAP_0_LOOP | IPR_CAP_0_HALF_LOOP)) { - struct emu10k1x_voice *pvoice = &chip->capture_voice; - if (pvoice->use) - snd_emu10k1x_pcm_interrupt(chip, pvoice); + struct emu10k1x_voice *cap_voice = &chip->capture_voice; + if (cap_voice->use) + snd_emu10k1x_pcm_interrupt(chip, cap_voice); else snd_emu10k1x_intr_disable(chip, INTE_CAP_0_LOOP | From c3daa92d60552891057b65f278d882348b76fffe Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:52:50 +0100 Subject: [PATCH 081/250] [ALSA] sound: ice1712.c fix shadowed variable warnings In all four case, adding a private value to the iooff index, call it priv_idx. sound/pci/ice1712/ice1712.c:1300:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here sound/pci/ice1712/ice1712.c:1312:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here sound/pci/ice1712/ice1712.c:1338:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here sound/pci/ice1712/ice1712.c:1350:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here [tiwai - fixed coding issues as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 40 ++++++++++++++++++++++--------------- 1 file changed, 24 insertions(+), 16 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index df292af67381..38e93ca12e27 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1297,11 +1297,14 @@ static void snd_ice1712_update_volume(struct snd_ice1712 *ice, int index) static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; spin_lock_irq(&ice->reg_lock); - ucontrol->value.integer.value[0] = !((ice->pro_volumes[index] >> 15) & 1); - ucontrol->value.integer.value[1] = !((ice->pro_volumes[index] >> 31) & 1); + ucontrol->value.integer.value[0] = + !((ice->pro_volumes[priv_idx] >> 15) & 1); + ucontrol->value.integer.value[1] = + !((ice->pro_volumes[priv_idx] >> 31) & 1); spin_unlock_irq(&ice->reg_lock); return 0; } @@ -1309,16 +1312,17 @@ static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struc static int snd_ice1712_pro_mixer_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; unsigned int nval, change; nval = (ucontrol->value.integer.value[0] ? 0 : 0x00008000) | (ucontrol->value.integer.value[1] ? 0 : 0x80000000); spin_lock_irq(&ice->reg_lock); - nval |= ice->pro_volumes[index] & ~0x80008000; - change = nval != ice->pro_volumes[index]; - ice->pro_volumes[index] = nval; - snd_ice1712_update_volume(ice, index); + nval |= ice->pro_volumes[priv_idx] & ~0x80008000; + change = nval != ice->pro_volumes[priv_idx]; + ice->pro_volumes[priv_idx] = nval; + snd_ice1712_update_volume(ice, priv_idx); spin_unlock_irq(&ice->reg_lock); return change; } @@ -1335,11 +1339,14 @@ static int snd_ice1712_pro_mixer_volume_info(struct snd_kcontrol *kcontrol, stru static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; spin_lock_irq(&ice->reg_lock); - ucontrol->value.integer.value[0] = (ice->pro_volumes[index] >> 0) & 127; - ucontrol->value.integer.value[1] = (ice->pro_volumes[index] >> 16) & 127; + ucontrol->value.integer.value[0] = + (ice->pro_volumes[priv_idx] >> 0) & 127; + ucontrol->value.integer.value[1] = + (ice->pro_volumes[priv_idx] >> 16) & 127; spin_unlock_irq(&ice->reg_lock); return 0; } @@ -1347,16 +1354,17 @@ static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struc static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; unsigned int nval, change; nval = (ucontrol->value.integer.value[0] & 127) | ((ucontrol->value.integer.value[1] & 127) << 16); spin_lock_irq(&ice->reg_lock); - nval |= ice->pro_volumes[index] & ~0x007f007f; - change = nval != ice->pro_volumes[index]; - ice->pro_volumes[index] = nval; - snd_ice1712_update_volume(ice, index); + nval |= ice->pro_volumes[priv_idx] & ~0x007f007f; + change = nval != ice->pro_volumes[priv_idx]; + ice->pro_volumes[priv_idx] = nval; + snd_ice1712_update_volume(ice, priv_idx); spin_unlock_irq(&ice->reg_lock); return change; } From ff143874d09a5850e7bf6c68d141243cb12a7b58 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:46:57 +0100 Subject: [PATCH 082/250] [ALSA] sound: virtuoso.c fix shadowed variable warning Use priv_idx as an identifier. sound/pci/oxygen/virtuoso.c:277:15: warning: symbol 'index' shadows an earlier one sound/pci/oxygen/virtuoso.c:56:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index d163397b85cc..e4e23789080a 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -274,12 +274,12 @@ static void xonar_gpio_changed(struct oxygen *chip) static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) { - unsigned int index = chip->controls[control]->private_value & 0xff; + unsigned int priv_idx = chip->controls[control]->private_value & 0xff; u16 value; - value = oxygen_read_ac97(chip, 0, index); + value = oxygen_read_ac97(chip, 0, priv_idx); if (!(value & 0x8000)) { - oxygen_write_ac97(chip, 0, index, value | 0x8000); + oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000); snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->controls[control]->id); } From caba7f70fce924dc5da2019f7678189086d0acd4 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:53:59 +0100 Subject: [PATCH 083/250] [ALSA] sound: pcxhr_core.c fix shadowed variable warning Inner loop redeclares err with u32 rather than int, stupid fix here is to change the inner err to err2. sound/pci/pcxhr/pcxhr_core.c:1008:8: warning: symbol 'err' shadows an earlier one sound/pci/pcxhr/pcxhr_core.c:983:6: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_core.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index c4e415d07380..846afbd30682 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1005,30 +1005,37 @@ void pcxhr_msg_tasklet(unsigned long arg) int nb_stream = (prmh->stat[i] >> (2*FIELD_SIZE)) & MASK_FIRST_FIELD; int pipe = prmh->stat[i] & MASK_FIRST_FIELD; int is_capture = prmh->stat[i] & 0x400000; - u32 err; + u32 err2; if (prmh->stat[i] & 0x800000) { /* if BIT_END */ snd_printdd("TASKLET : End%sPipe %d\n", is_capture ? "Record" : "Play", pipe); } i++; - err = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1]; - if (err) - pcxhr_handle_async_err(mgr, err, PCXHR_ERR_PIPE, + err2 = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1]; + if (err2) + pcxhr_handle_async_err(mgr, err2, + PCXHR_ERR_PIPE, pipe, is_capture); i += 2; for (j = 0; j < nb_stream; j++) { - err = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1]; - if (err) - pcxhr_handle_async_err(mgr, err, PCXHR_ERR_STREAM, - pipe, is_capture); + err2 = prmh->stat[i] ? + prmh->stat[i] : prmh->stat[i+1]; + if (err2) + pcxhr_handle_async_err(mgr, err2, + PCXHR_ERR_STREAM, + pipe, + is_capture); i += 2; } for (j = 0; j < nb_audio; j++) { - err = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1]; - if (err) - pcxhr_handle_async_err(mgr, err, PCXHR_ERR_AUDIO, - pipe, is_capture); + err2 = prmh->stat[i] ? + prmh->stat[i] : prmh->stat[i+1]; + if (err2) + pcxhr_handle_async_err(mgr, err2, + PCXHR_ERR_AUDIO, + pipe, + is_capture); i += 2; } } From 0cd87b10ca29a351c61c8c63761ab8fb48e47b2f Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:54:26 +0100 Subject: [PATCH 084/250] [ALSA] sound: riptide.c fix shadowed variable warnings In both cases we are passing around the substream number, use sub_num for this. sound/pci/riptide/riptide.c:1633:6: warning: symbol 'index' shadows an earlier one sound/pci/riptide/riptide.c:121:12: originally declared here sound/pci/riptide/riptide.c:1673:6: warning: symbol 'index' shadows an earlier one sound/pci/riptide/riptide.c:121:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 9408b1eeec40..979f7da641ce 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1630,14 +1630,14 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream) struct snd_riptide *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct pcmhw *data; - int index = substream->number; + int sub_num = substream->number; - chip->playback_substream[index] = substream; + chip->playback_substream[sub_num] = substream; runtime->hw = snd_riptide_playback; data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); - data->paths = lbus_play_paths[index]; - data->id = play_ids[index]; - data->source = play_sources[index]; + data->paths = lbus_play_paths[sub_num]; + data->id = play_ids[sub_num]; + data->source = play_sources[sub_num]; data->intdec[0] = 0xff; data->intdec[1] = 0xff; data->state = ST_STOP; @@ -1670,10 +1670,10 @@ static int snd_riptide_playback_close(struct snd_pcm_substream *substream) { struct snd_riptide *chip = snd_pcm_substream_chip(substream); struct pcmhw *data = get_pcmhwdev(substream); - int index = substream->number; + int sub_num = substream->number; substream->runtime->private_data = NULL; - chip->playback_substream[index] = NULL; + chip->playback_substream[sub_num] = NULL; kfree(data); return 0; } From 0b76b51e5807951995a39ea791b39971a7ae945f Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:54:49 +0100 Subject: [PATCH 085/250] [ALSA] sound: hdspm.c fix returning void expression warnings Just drop the returns. sound/pci/rme9652/hdspm.c:1031:3: warning: returning void-valued expression sound/pci/rme9652/hdspm.c:1033:3: warning: returning void-valued expression Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9a19ae6a64d9..38c931c480d9 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1028,9 +1028,9 @@ static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id, { /* the hardware already does the relevant bit-mask with 0xff */ if (id) - return hdspm_write(hdspm, HDSPM_midiDataOut1, val); + hdspm_write(hdspm, HDSPM_midiDataOut1, val); else - return hdspm_write(hdspm, HDSPM_midiDataOut0, val); + hdspm_write(hdspm, HDSPM_midiDataOut0, val); } static inline int snd_hdspm_midi_input_available (struct hdspm *hdspm, int id) From 8b55178515e8872670dc830203dad0e9e51e16be Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:56:48 +0100 Subject: [PATCH 086/250] [ALSA] sound: patch_sigmatel.c fix shadowed variable warning Temp variable in the loop shadows the second argument (which is otherwise unused in this function). Change this to defcfg as it is used to hold the default config. sound/pci/hda/patch_sigmatel.c:2759:18: warning: symbol 'cfg' shadows an earlier one sound/pci/hda/patch_sigmatel.c:2734:26: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4bc7d8646fab..3bf528d8fc22 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2849,11 +2849,11 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) { for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; - unsigned long cfg; - cfg = snd_hda_codec_read(codec, pin, 0, + unsigned long defcfg; + defcfg = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONFIG_DEFAULT, 0x00); - if (get_defcfg_device(cfg) == AC_JACK_SPEAKER) { + if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { unsigned long wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); if (wcaps == AC_WCAP_OUT_AMP) From 64ed0dfd1f42edb15f4d18c13d7696edbc2f7e4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Feb 2008 11:57:53 +0100 Subject: [PATCH 087/250] [ALSA] hda-codec - Use int instead of long in patch_sigmatel.c The HD-audio parameters are at most 32bit int. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3bf528d8fc22..e1d61a035bab 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2838,7 +2838,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, */ for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.speaker_pins[i]; - unsigned long wcaps = get_wcaps(codec, pin); + unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); if (wcaps == AC_WCAP_OUT_AMP) /* found a mono speaker with an amp, must be lfe */ @@ -2849,12 +2849,12 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) { for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; - unsigned long defcfg; + unsigned int defcfg; defcfg = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONFIG_DEFAULT, 0x00); if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { - unsigned long wcaps = get_wcaps(codec, pin); + unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); if (wcaps == AC_WCAP_OUT_AMP) /* found a mono speaker with an amp, From 3c9a3203ff9863fbe798030928f496347c2ed3bd Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:59:26 +0100 Subject: [PATCH 088/250] [ALSA] sound: hda: missing includes of hda_patch.h Move the array declaration to hda_codec.c where it is used and add includes where the individual presets are declared. Fixes the following sparse warnings: sound/pci/hda/patch_realtek.c:13744:25: warning: symbol 'snd_hda_preset_realtek' was not declared. Should it be static? sound/pci/hda/patch_cmedia.c:729:25: warning: symbol 'snd_hda_preset_cmedia' was not declared. Should it be static? sound/pci/hda/patch_analog.c:3656:25: warning: symbol 'snd_hda_preset_analog' was not declared. Should it be static? sound/pci/hda/patch_sigmatel.c:3995:25: warning: symbol 'snd_hda_preset_sigmatel' was not declared. Should it be static? sound/pci/hda/patch_si3054.c:286:25: warning: symbol 'snd_hda_preset_si3054' was not declared. Should it be static? sound/pci/hda/patch_atihdmi.c:156:25: warning: symbol 'snd_hda_preset_atihdmi' was not declared. Should it be static? sound/pci/hda/patch_conexant.c:1721:25: warning: symbol 'snd_hda_preset_conexant' was not declared. Should it be static? sound/pci/hda/patch_via.c:1962:25: warning: symbol 'snd_hda_preset_via' was not declared. Should it be static? Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 31 ++++++++++++++++++++++++++++--- sound/pci/hda/hda_patch.h | 28 ---------------------------- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_atihdmi.c | 1 + sound/pci/hda/patch_cmedia.c | 1 + sound/pci/hda/patch_conexant.c | 1 + sound/pci/hda/patch_realtek.c | 1 + sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + sound/pci/hda/patch_via.c | 2 +- 10 files changed, 36 insertions(+), 33 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8ab88d9ba3b5..e6bace83e7cf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -31,6 +31,7 @@ #include #include "hda_local.h" #include +#include "hda_patch.h" /* codec presets */ #ifdef CONFIG_SND_HDA_POWER_SAVE /* define this option here to hide as static */ @@ -68,9 +69,33 @@ static struct hda_vendor_id hda_vendor_ids[] = { {} /* terminator */ }; -/* codec presets */ -#include "hda_patch.h" - +static const struct hda_codec_preset *hda_preset_tables[] = { +#ifdef CONFIG_SND_HDA_CODEC_REALTEK + snd_hda_preset_realtek, +#endif +#ifdef CONFIG_SND_HDA_CODEC_CMEDIA + snd_hda_preset_cmedia, +#endif +#ifdef CONFIG_SND_HDA_CODEC_ANALOG + snd_hda_preset_analog, +#endif +#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL + snd_hda_preset_sigmatel, +#endif +#ifdef CONFIG_SND_HDA_CODEC_SI3054 + snd_hda_preset_si3054, +#endif +#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI + snd_hda_preset_atihdmi, +#endif +#ifdef CONFIG_SND_HDA_CODEC_CONEXANT + snd_hda_preset_conexant, +#endif +#ifdef CONFIG_SND_HDA_CODEC_VIA + snd_hda_preset_via, +#endif + NULL +}; #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_power_work(struct work_struct *work); diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h index f5c23bb16d7e..2fdf2358dbc2 100644 --- a/sound/pci/hda/hda_patch.h +++ b/sound/pci/hda/hda_patch.h @@ -18,31 +18,3 @@ extern struct hda_codec_preset snd_hda_preset_atihdmi[]; extern struct hda_codec_preset snd_hda_preset_conexant[]; /* VIA codecs */ extern struct hda_codec_preset snd_hda_preset_via[]; - -static const struct hda_codec_preset *hda_preset_tables[] = { -#ifdef CONFIG_SND_HDA_CODEC_REALTEK - snd_hda_preset_realtek, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CMEDIA - snd_hda_preset_cmedia, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ANALOG - snd_hda_preset_analog, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL - snd_hda_preset_sigmatel, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SI3054 - snd_hda_preset_si3054, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI - snd_hda_preset_atihdmi, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CONEXANT - snd_hda_preset_conexant, -#endif -#ifdef CONFIG_SND_HDA_CODEC_VIA - snd_hda_preset_via, -#endif - NULL -}; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 20446e320b2c..87db3c410a10 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,6 +28,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 45a2e30cbf42..12272508b112 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -27,6 +27,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" struct atihdmi_spec { struct hda_multi_out multiout; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 9794d4166ae4..1892c81f1d11 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -28,6 +28,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define NUM_PINS 11 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bb9a58db9fa..e4fa9a35848f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,6 +27,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2100ee480809..33c9505adbae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 598ee2119bbe..9332b63e406c 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -28,7 +28,7 @@ #include #include "hda_codec.h" #include "hda_local.h" - +#include "hda_patch.h" /* si3054 verbs */ #define SI3054_VERB_READ_NODE 0x900 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e1d61a035bab..47d3536a6576 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,6 +32,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define NUM_CONTROL_ALLOC 32 #define STAC_PWR_EVENT 0x20 diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3515a3fb5d9d..09f1c25eb7e5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -39,7 +39,7 @@ #include #include "hda_codec.h" #include "hda_local.h" - +#include "hda_patch.h" /* amp values */ #define AMP_VAL_IDX_SHIFT 19 From bce6c2b5b4dbe8cd97c48c633b62adeb535954ad Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 29 Feb 2008 12:07:43 +0100 Subject: [PATCH 089/250] [ALSA] hda: disable power management on fixed ports Power management can't be enabled on fixed ports, since the presence will always return false and prevent output. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 47d3536a6576..9b242a263637 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3016,12 +3016,16 @@ static int stac92xx_init(struct hda_codec *codec) ? STAC_HP_EVENT : STAC_PWR_EVENT; int pinctl = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); } From 52fe0f9d59cf4f5842bd319e4f212f907abd2e5d Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 29 Feb 2008 12:08:20 +0100 Subject: [PATCH 090/250] [ALSA] hda: add verbs for 92hd73xxx laptops Added core_init[] for several 92hd73xxx laptops. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9b242a263637..f6c02c0b1f8d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -534,6 +534,24 @@ static struct hda_verb stac92hd73xx_6ch_core_init[] = { {} }; +static struct hda_verb dell_m6_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + static struct hda_verb stac92hd73xx_8ch_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -3442,6 +3460,7 @@ again: switch (spec->board_config) { case STAC_DELL_M6: + spec->init = dell_m6_core_init; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: From fd6640fa2d8b5f5f471aad5abd8ce5d6995df563 Mon Sep 17 00:00:00 2001 From: Pawel MOLL Date: Fri, 29 Feb 2008 12:41:31 +0100 Subject: [PATCH 091/250] [ALSA] IEC958 definitions for consumer status channel, byte 4 Added definition for byte 4 of SPDIF channel status, according to second edition of IEC 60958-3 (consumer) spec. Signed-off-by: Pawel MOLL Signed-off-by: Takashi Iwai --- include/sound/asoundef.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index 024ce62f7d16..a6e0facf8a37 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -112,6 +112,14 @@ #define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /* 1000 ppm */ #define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /* 50 ppm */ #define IEC958_AES3_CON_CLOCK_VARIABLE (2<<4) /* variable pitch */ +#define IEC958_AES4_CON_MAX_WORDLEN_24 (1<<0) /* 0 = 20-bit, 1 = 24-bit */ +#define IEC958_AES4_CON_WORDLEN (7<<1) /* mask - sample word length */ +#define IEC958_AES4_CON_WORDLEN_NOTID (0<<1) /* not indicated */ +#define IEC958_AES4_CON_WORDLEN_20_16 (1<<1) /* 20-bit or 16-bit */ +#define IEC958_AES4_CON_WORDLEN_22_18 (2<<1) /* 22-bit or 18-bit */ +#define IEC958_AES4_CON_WORDLEN_23_19 (4<<1) /* 23-bit or 19-bit */ +#define IEC958_AES4_CON_WORDLEN_24_20 (5<<1) /* 24-bit or 20-bit */ +#define IEC958_AES4_CON_WORDLEN_21_17 (6<<1) /* 21-bit or 17-bit */ /***************************************************************************** * * From 40ac8c4f208111cdc1542ccc9feb21b98a6b0219 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Feb 2008 14:16:17 +0100 Subject: [PATCH 092/250] [ALSA] hda-codec - Fix the array over-range access with stac92hd71bxx codec Add the check of the array range for dac_nids to prevent the over-range access. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f6c02c0b1f8d..6c85e7e81034 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2363,7 +2363,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, unsigned int wid_caps, pincap; - for (i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs && i < spec->multiout.num_dacs; i++) { if (!spec->multiout.dac_nids[i]) continue; From 9ab4d072ad67793d70b8707e14fb9261749c4e07 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Mon, 3 Mar 2008 10:53:54 +0100 Subject: [PATCH 093/250] [ALSA] Add PC-speaker sound driver Added PC-speaker sound driver (snd-pcsp). Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/Kconfig | 17 ++ sound/drivers/Makefile | 2 +- sound/drivers/pcsp/Makefile | 2 + sound/drivers/pcsp/pcsp.c | 241 ++++++++++++++++++++++ sound/drivers/pcsp/pcsp.h | 82 ++++++++ sound/drivers/pcsp/pcsp_input.c | 116 +++++++++++ sound/drivers/pcsp/pcsp_input.h | 14 ++ sound/drivers/pcsp/pcsp_lib.c | 347 ++++++++++++++++++++++++++++++++ sound/drivers/pcsp/pcsp_mixer.c | 143 +++++++++++++ 9 files changed, 963 insertions(+), 1 deletion(-) create mode 100644 sound/drivers/pcsp/Makefile create mode 100644 sound/drivers/pcsp/pcsp.c create mode 100644 sound/drivers/pcsp/pcsp.h create mode 100644 sound/drivers/pcsp/pcsp_input.c create mode 100644 sound/drivers/pcsp/pcsp_input.h create mode 100644 sound/drivers/pcsp/pcsp_lib.c create mode 100644 sound/drivers/pcsp/pcsp_mixer.c diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 75d4fe09fdf3..78648c4e9e73 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -4,6 +4,23 @@ menu "Generic devices" depends on SND!=n +config SND_PCSP + tristate "Internal PC speaker support" + depends on X86_PC && HIGH_RES_TIMERS + help + If you don't have a sound card in your computer, you can include a + driver for the PC speaker which allows it to act like a primitive + sound card. + This driver also replaces the pcspkr driver for beeps. + + You can compile this as a module which will be called snd-pcsp. + + You don't need this driver if you only want your pc-speaker to beep. + You don't need this driver if you have a tablet piezo beeper + in your PC instead of the real speaker. + + It should not hurt to say Y or M here in all other cases. + config SND_MPU401_UART tristate select SND_RAWMIDI diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index 8e5530006e1f..d4a07f9ff2c7 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -20,4 +20,4 @@ obj-$(CONFIG_SND_MTS64) += snd-mts64.o obj-$(CONFIG_SND_PORTMAN2X4) += snd-portman2x4.o obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403-ac97cr.o -obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ +obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ pcsp/ diff --git a/sound/drivers/pcsp/Makefile b/sound/drivers/pcsp/Makefile new file mode 100644 index 000000000000..b19555b440da --- /dev/null +++ b/sound/drivers/pcsp/Makefile @@ -0,0 +1,2 @@ +snd-pcsp-objs := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o +obj-$(CONFIG_SND_PCSP) += snd-pcsp.o diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c new file mode 100644 index 000000000000..34477286b394 --- /dev/null +++ b/sound/drivers/pcsp/pcsp.c @@ -0,0 +1,241 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 1997-2001 David Woodhouse + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include "pcsp_input.h" +#include "pcsp.h" + +MODULE_AUTHOR("Stas Sergeev "); +MODULE_DESCRIPTION("PC-Speaker driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{PC-Speaker, pcsp}}"); +MODULE_ALIAS("platform:pcspkr"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "dummy"); + +struct snd_pcsp pcsp_chip; + +static int __devinit snd_pcsp_create(struct snd_card *card) +{ + static struct snd_device_ops ops = { }; + struct timespec tp; + int err; + int div, min_div, order; + + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { + printk(KERN_ERR "PCSP: Timer resolution is not sufficient " + "(%linS)\n", tp.tv_nsec); + printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " + "enabled.\n"); + return -EIO; + } + + if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS) + min_div = MIN_DIV; + else + min_div = MAX_DIV; +#if PCSP_DEBUG + printk("PCSP: lpj=%li, min_div=%i, res=%li\n", + loops_per_jiffy, min_div, tp.tv_nsec); +#endif + + div = MAX_DIV / min_div; + order = fls(div) - 1; + + pcsp_chip.max_treble = min(order, PCSP_MAX_TREBLE); + pcsp_chip.treble = min(pcsp_chip.max_treble, PCSP_DEFAULT_TREBLE); + pcsp_chip.playback_ptr = 0; + pcsp_chip.period_ptr = 0; + atomic_set(&pcsp_chip.timer_active, 0); + pcsp_chip.enable = 1; + pcsp_chip.pcspkr = 1; + + spin_lock_init(&pcsp_chip.substream_lock); + + pcsp_chip.card = card; + pcsp_chip.port = 0x61; + pcsp_chip.irq = -1; + pcsp_chip.dma = -1; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, &pcsp_chip, &ops); + if (err < 0) + return err; + + return 0; +} + +static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) +{ + struct snd_card *card; + int err; + + if (devnum != 0) + return -EINVAL; + + hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE; + pcsp_chip.timer.function = pcsp_do_timer; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (!card) + return -ENOMEM; + + err = snd_pcsp_create(card); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_pcsp_new_pcm(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_pcsp_new_mixer(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + snd_card_set_dev(pcsp_chip.card, dev); + + strcpy(card->driver, "PC-Speaker"); + strcpy(card->shortname, "pcsp"); + sprintf(card->longname, "Internal PC-Speaker at port 0x%x", + pcsp_chip.port); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + return 0; +} + +static int __devinit alsa_card_pcsp_init(struct device *dev) +{ + int devnum = 0, cards = 0; + +#ifdef CONFIG_DEBUG_PAGEALLOC + /* Well, CONFIG_DEBUG_PAGEALLOC makes the sound horrible. Lets alert */ + printk(KERN_WARNING + "PCSP: Warning, CONFIG_DEBUG_PAGEALLOC is enabled!\n" + "You have to disable it if you want to use the PC-Speaker " + "driver.\n" + "Unless it is disabled, enjoy the horrible, distorted " + "and crackling noise.\n"); +#endif + + if (enable) { + if (snd_card_pcsp_probe(devnum, dev) >= 0) + cards++; + if (!cards) { + printk(KERN_ERR "PC-Speaker initialization failed.\n"); + return -ENODEV; + } + } + + return 0; +} + +static void __devexit alsa_card_pcsp_exit(struct snd_pcsp *chip) +{ + snd_card_free(chip->card); +} + +static int __devinit pcsp_probe(struct platform_device *dev) +{ + int err; + err = pcspkr_input_init(&pcsp_chip.input_dev, &dev->dev); + if (err < 0) + return err; + + err = alsa_card_pcsp_init(&dev->dev); + if (err < 0) { + pcspkr_input_remove(pcsp_chip.input_dev); + return err; + } + + platform_set_drvdata(dev, &pcsp_chip); + return 0; +} + +static int __devexit pcsp_remove(struct platform_device *dev) +{ + struct snd_pcsp *chip = platform_get_drvdata(dev); + alsa_card_pcsp_exit(chip); + pcspkr_input_remove(chip->input_dev); + platform_set_drvdata(dev, NULL); + return 0; +} + +static void pcsp_stop_beep(struct snd_pcsp *chip) +{ + unsigned long flags; + spin_lock_irqsave(&chip->substream_lock, flags); + if (!chip->playback_substream) + pcspkr_stop_sound(); + spin_unlock_irqrestore(&chip->substream_lock, flags); +} + +static int pcsp_suspend(struct platform_device *dev, pm_message_t state) +{ + struct snd_pcsp *chip = platform_get_drvdata(dev); + pcsp_stop_beep(chip); + snd_pcm_suspend_all(chip->pcm); + return 0; +} + +static void pcsp_shutdown(struct platform_device *dev) +{ + struct snd_pcsp *chip = platform_get_drvdata(dev); + pcsp_stop_beep(chip); +} + +static struct platform_driver pcsp_platform_driver = { + .driver = { + .name = "pcspkr", + .owner = THIS_MODULE, + }, + .probe = pcsp_probe, + .remove = __devexit_p(pcsp_remove), + .suspend = pcsp_suspend, + .shutdown = pcsp_shutdown, +}; + +static int __init pcsp_init(void) +{ + return platform_driver_register(&pcsp_platform_driver); +} + +static void __exit pcsp_exit(void) +{ + platform_driver_unregister(&pcsp_platform_driver); +} + +module_init(pcsp_init); +module_exit(pcsp_exit); diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h new file mode 100644 index 000000000000..f07cc1ee1fe7 --- /dev/null +++ b/sound/drivers/pcsp/pcsp.h @@ -0,0 +1,82 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 1993-1997 Michael Beck + * Copyright (C) 1997-2001 David Woodhouse + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#ifndef __PCSP_H__ +#define __PCSP_H__ + +#include +#if defined(CONFIG_MIPS) || defined(CONFIG_X86) +/* Use the global PIT lock ! */ +#include +#else +#include +static DEFINE_SPINLOCK(i8253_lock); +#endif + +#define PCSP_SOUND_VERSION 0x400 /* read 4.00 */ +#define PCSP_DEBUG 0 + +/* default timer freq for PC-Speaker: 18643 Hz */ +#define DIV_18KHZ 64 +#define MAX_DIV DIV_18KHZ +#define CUR_DIV() (MAX_DIV >> chip->treble) +#define PCSP_MAX_TREBLE 1 + +/* unfortunately, with hrtimers 37KHz does not work very well :( */ +#define PCSP_DEFAULT_TREBLE 0 +#define MIN_DIV (MAX_DIV >> PCSP_MAX_TREBLE) + +/* wild guess */ +#define PCSP_MIN_LPJ 1000000 +#define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1) +#define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV) +#define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble)) +#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV()) +#define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE +#define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE +#define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1) +#define PCSP_MIN_PERIOD_NS (1000000000ULL * PCSP_MAX_RATE__1) +#define PCSP_CALC_NS(div) ({ \ + u64 __val = 1000000000ULL * (div); \ + do_div(__val, PIT_TICK_RATE); \ + __val; \ +}) +#define PCSP_PERIOD_NS() PCSP_CALC_NS(CUR_DIV()) + +#define PCSP_MAX_PERIOD_SIZE (64*1024) +#define PCSP_MAX_PERIODS 512 +#define PCSP_BUFFER_SIZE (128*1024) + +struct snd_pcsp { + struct snd_card *card; + struct snd_pcm *pcm; + struct input_dev *input_dev; + struct hrtimer timer; + unsigned short port, irq, dma; + spinlock_t substream_lock; + struct snd_pcm_substream *playback_substream; + size_t playback_ptr; + size_t period_ptr; + atomic_t timer_active; + int thalf; + u64 ns_rem; + unsigned char val61; + int enable; + int max_treble; + int treble; + int pcspkr; +}; + +extern struct snd_pcsp pcsp_chip; + +extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); + +extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); +extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); + +#endif diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c new file mode 100644 index 000000000000..cd9b83e7f7d1 --- /dev/null +++ b/sound/drivers/pcsp/pcsp_input.c @@ -0,0 +1,116 @@ +/* + * PC Speaker beeper driver for Linux + * + * Copyright (c) 2002 Vojtech Pavlik + * Copyright (c) 1992 Orest Zborowski + * + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation + */ + +#include +#include +#include +#include "pcsp.h" + +static void pcspkr_do_sound(unsigned int count) +{ + unsigned long flags; + + spin_lock_irqsave(&i8253_lock, flags); + + if (count) { + /* enable counter 2 */ + outb_p(inb_p(0x61) | 3, 0x61); + /* set command for counter 2, 2 byte write */ + outb_p(0xB6, 0x43); + /* select desired HZ */ + outb_p(count & 0xff, 0x42); + outb((count >> 8) & 0xff, 0x42); + } else { + /* disable counter 2 */ + outb(inb_p(0x61) & 0xFC, 0x61); + } + + spin_unlock_irqrestore(&i8253_lock, flags); +} + +void pcspkr_stop_sound(void) +{ + pcspkr_do_sound(0); +} + +static int pcspkr_input_event(struct input_dev *dev, unsigned int type, + unsigned int code, int value) +{ + unsigned int count = 0; + + if (atomic_read(&pcsp_chip.timer_active) || !pcsp_chip.pcspkr) + return 0; + + switch (type) { + case EV_SND: + switch (code) { + case SND_BELL: + if (value) + value = 1000; + case SND_TONE: + break; + default: + return -1; + } + break; + + default: + return -1; + } + + if (value > 20 && value < 32767) + count = PIT_TICK_RATE / value; + + pcspkr_do_sound(count); + + return 0; +} + +int __devinit pcspkr_input_init(struct input_dev **rdev, struct device *dev) +{ + int err; + + struct input_dev *input_dev = input_allocate_device(); + if (!input_dev) + return -ENOMEM; + + input_dev->name = "PC Speaker"; + input_dev->phys = "isa0061/input0"; + input_dev->id.bustype = BUS_ISA; + input_dev->id.vendor = 0x001f; + input_dev->id.product = 0x0001; + input_dev->id.version = 0x0100; + input_dev->dev.parent = dev; + + input_dev->evbit[0] = BIT(EV_SND); + input_dev->sndbit[0] = BIT(SND_BELL) | BIT(SND_TONE); + input_dev->event = pcspkr_input_event; + + err = input_register_device(input_dev); + if (err) { + input_free_device(input_dev); + return err; + } + + *rdev = input_dev; + return 0; +} + +int pcspkr_input_remove(struct input_dev *dev) +{ + pcspkr_stop_sound(); + input_unregister_device(dev); /* this also does kfree() */ + + return 0; +} diff --git a/sound/drivers/pcsp/pcsp_input.h b/sound/drivers/pcsp/pcsp_input.h new file mode 100644 index 000000000000..e66738c78333 --- /dev/null +++ b/sound/drivers/pcsp/pcsp_input.h @@ -0,0 +1,14 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#ifndef __PCSP_INPUT_H__ +#define __PCSP_INPUT_H__ + +int __devinit pcspkr_input_init(struct input_dev **rdev, struct device *dev); +int pcspkr_input_remove(struct input_dev *dev); +void pcspkr_stop_sound(void); + +#endif diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c new file mode 100644 index 000000000000..6bdcb89129d8 --- /dev/null +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -0,0 +1,347 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 1993-1997 Michael Beck + * Copyright (C) 1997-2001 David Woodhouse + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#include +#include +#include +#include +#include +#include +#include +#include "pcsp.h" + +static int nforce_wa; +module_param(nforce_wa, bool, 0444); +MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " + "(expect bad sound)"); + +#define DMIX_WANTS_S16 1 + +static void pcsp_start_timer(unsigned long dummy) +{ + hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); +} + +/* + * We need the hrtimer_start as a tasklet to avoid + * the nasty locking problem. :( + * The problem: + * - The timer handler is called with the cpu_base->lock + * already held by hrtimer code. + * - snd_pcm_period_elapsed() takes the + * substream->self_group.lock. + * So far so good. + * But the snd_pcsp_trigger() is called with the + * substream->self_group.lock held, and it calls + * hrtimer_start(), which takes the cpu_base->lock. + * You see the problem. We have the code pathes + * which take two locks in a reverse order. This + * can deadlock and the lock validator complains. + * The only solution I could find was to move the + * hrtimer_start() into a tasklet. -stsp + */ +DECLARE_TASKLET(pcsp_start_timer_tasklet, pcsp_start_timer, 0); + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + unsigned long flags; + unsigned char timer_cnt, val; + int fmt_size, periods_elapsed; + u64 ns; + size_t period_bytes, buffer_bytes; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + + if (chip->thalf) { + outb(chip->val61, 0x61); + chip->thalf = 0; + if (!atomic_read(&chip->timer_active)) + return HRTIMER_NORESTART; + hrtimer_forward(&chip->timer, chip->timer.expires, + ktime_set(0, chip->ns_rem)); + return HRTIMER_RESTART; + } + + /* hrtimer calls us from both hardirq and softirq contexts, + * so irqsave :( */ + spin_lock_irqsave(&chip->substream_lock, flags); + /* Takashi Iwai says regarding this extra lock: + + If the irq handler handles some data on the DMA buffer, it should + do snd_pcm_stream_lock(). + That protects basically against all races among PCM callbacks, yes. + However, there are two remaining issues: + 1. The substream pointer you try to lock isn't protected _before_ + this lock yet. + 2. snd_pcm_period_elapsed() itself acquires the lock. + The requirement of another lock is because of 1. When you get + chip->playback_substream, it's not protected. + Keeping this lock while snd_pcm_period_elapsed() assures the substream + is still protected (at least, not released). And the other status is + handled properly inside snd_pcm_stream_lock() in + snd_pcm_period_elapsed(). + + */ + if (!chip->playback_substream) + goto exit_nr_unlock1; + substream = chip->playback_substream; + snd_pcm_stream_lock(substream); + if (!atomic_read(&chip->timer_active)) + goto exit_nr_unlock2; + + runtime = substream->runtime; + fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3; + /* assume it is mono! */ + val = runtime->dma_area[chip->playback_ptr + fmt_size - 1]; + if (snd_pcm_format_signed(runtime->format)) + val ^= 0x80; + timer_cnt = val * CUR_DIV() / 256; + + if (timer_cnt && chip->enable) { + spin_lock(&i8253_lock); + if (!nforce_wa) { + outb_p(chip->val61, 0x61); + outb_p(timer_cnt, 0x42); + outb(chip->val61 ^ 1, 0x61); + } else { + outb(chip->val61 ^ 2, 0x61); + chip->thalf = 1; + } + spin_unlock(&i8253_lock); + } + + period_bytes = snd_pcm_lib_period_bytes(substream); + buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + chip->playback_ptr += PCSP_INDEX_INC() * fmt_size; + periods_elapsed = chip->playback_ptr - chip->period_ptr; + if (periods_elapsed < 0) { + printk(KERN_WARNING "PCSP: playback_ptr inconsistent " + "(%zi %zi %zi)\n", + chip->playback_ptr, period_bytes, buffer_bytes); + periods_elapsed += buffer_bytes; + } + periods_elapsed /= period_bytes; + /* wrap the pointer _before_ calling snd_pcm_period_elapsed(), + * or ALSA will BUG on us. */ + chip->playback_ptr %= buffer_bytes; + + snd_pcm_stream_unlock(substream); + + if (periods_elapsed) { + snd_pcm_period_elapsed(substream); + chip->period_ptr += periods_elapsed * period_bytes; + chip->period_ptr %= buffer_bytes; + } + + spin_unlock_irqrestore(&chip->substream_lock, flags); + + if (!atomic_read(&chip->timer_active)) + return HRTIMER_NORESTART; + + chip->ns_rem = PCSP_PERIOD_NS(); + ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); + chip->ns_rem -= ns; + hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns)); + return HRTIMER_RESTART; + +exit_nr_unlock2: + snd_pcm_stream_unlock(substream); +exit_nr_unlock1: + spin_unlock_irqrestore(&chip->substream_lock, flags); + return HRTIMER_NORESTART; +} + +static void pcsp_start_playing(struct snd_pcsp *chip) +{ +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: start_playing called\n"); +#endif + if (atomic_read(&chip->timer_active)) { + printk(KERN_ERR "PCSP: Timer already active\n"); + return; + } + + spin_lock(&i8253_lock); + chip->val61 = inb(0x61) | 0x03; + outb_p(0x92, 0x43); /* binary, mode 1, LSB only, ch 2 */ + spin_unlock(&i8253_lock); + atomic_set(&chip->timer_active, 1); + chip->thalf = 0; + + tasklet_schedule(&pcsp_start_timer_tasklet); +} + +static void pcsp_stop_playing(struct snd_pcsp *chip) +{ +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: stop_playing called\n"); +#endif + if (!atomic_read(&chip->timer_active)) + return; + + atomic_set(&chip->timer_active, 0); + spin_lock(&i8253_lock); + /* restore the timer */ + outb_p(0xb6, 0x43); /* binary, mode 3, LSB/MSB, ch 2 */ + outb(chip->val61 & 0xFC, 0x61); + spin_unlock(&i8253_lock); +} + +static int snd_pcsp_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: close called\n"); +#endif + if (atomic_read(&chip->timer_active)) { + printk(KERN_ERR "PCSP: timer still active\n"); + pcsp_stop_playing(chip); + } + spin_lock_irq(&chip->substream_lock); + chip->playback_substream = NULL; + spin_unlock_irq(&chip->substream_lock); + return 0; +} + +static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err; + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + return 0; +} + +static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) +{ +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: hw_free called\n"); +#endif + return snd_pcm_lib_free_pages(substream); +} + +static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: prepare called, " + "size=%zi psize=%zi f=%zi f1=%i\n", + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + snd_pcm_lib_buffer_bytes(substream) / + snd_pcm_lib_period_bytes(substream), + substream->runtime->periods); +#endif + chip->playback_ptr = 0; + chip->period_ptr = 0; + return 0; +} + +static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: trigger called\n"); +#endif + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + pcsp_start_playing(chip); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + pcsp_stop_playing(chip); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream + *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); + return bytes_to_frames(substream->runtime, chip->playback_ptr); +} + +static struct snd_pcm_hardware snd_pcsp_playback = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_U8 +#if DMIX_WANTS_S16 + | SNDRV_PCM_FMTBIT_S16_LE +#endif + ), + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = PCSP_DEFAULT_SRATE, + .rate_max = PCSP_DEFAULT_SRATE, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = PCSP_BUFFER_SIZE, + .period_bytes_min = 64, + .period_bytes_max = PCSP_MAX_PERIOD_SIZE, + .periods_min = 2, + .periods_max = PCSP_MAX_PERIODS, + .fifo_size = 0, +}; + +static int snd_pcsp_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: open called\n"); +#endif + if (atomic_read(&chip->timer_active)) { + printk(KERN_ERR "PCSP: still active!!\n"); + return -EBUSY; + } + runtime->hw = snd_pcsp_playback; + chip->playback_substream = substream; + return 0; +} + +static struct snd_pcm_ops snd_pcsp_playback_ops = { + .open = snd_pcsp_playback_open, + .close = snd_pcsp_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_pcsp_playback_hw_params, + .hw_free = snd_pcsp_playback_hw_free, + .prepare = snd_pcsp_playback_prepare, + .trigger = snd_pcsp_trigger, + .pointer = snd_pcsp_playback_pointer, +}; + +int __devinit snd_pcsp_new_pcm(struct snd_pcsp *chip) +{ + int err; + + err = snd_pcm_new(chip->card, "pcspeaker", 0, 1, 0, &chip->pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(chip->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_pcsp_playback_ops); + + chip->pcm->private_data = chip; + chip->pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(chip->pcm->name, "pcsp"); + + snd_pcm_lib_preallocate_pages_for_all(chip->pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), PCSP_BUFFER_SIZE, + PCSP_BUFFER_SIZE); + + return 0; +} diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c new file mode 100644 index 000000000000..64a695fef74e --- /dev/null +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -0,0 +1,143 @@ +/* + * PC-Speaker driver for Linux + * + * Mixer implementation. + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#include +#include +#include "pcsp.h" + + +static int pcsp_enable_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int pcsp_enable_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = chip->enable; + return 0; +} + +static int pcsp_enable_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int enab = ucontrol->value.integer.value[0]; + if (enab != chip->enable) { + chip->enable = enab; + changed = 1; + } + return changed; +} + +static int pcsp_treble_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = chip->max_treble + 1; + if (uinfo->value.enumerated.item > chip->max_treble) + uinfo->value.enumerated.item = chip->max_treble; + sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE()); + return 0; +} + +static int pcsp_treble_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.enumerated.item[0] = chip->treble; + return 0; +} + +static int pcsp_treble_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int treble = ucontrol->value.enumerated.item[0]; + if (treble != chip->treble) { + chip->treble = treble; +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE()); +#endif + changed = 1; + } + return changed; +} + +static int pcsp_pcspkr_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int pcsp_pcspkr_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = chip->pcspkr; + return 0; +} + +static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int spkr = ucontrol->value.integer.value[0]; + if (spkr != chip->pcspkr) { + chip->pcspkr = spkr; + changed = 1; + } + return changed; +} + +#define PCSP_MIXER_CONTROL(ctl_type, ctl_name) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = ctl_name, \ + .info = pcsp_##ctl_type##_info, \ + .get = pcsp_##ctl_type##_get, \ + .put = pcsp_##ctl_type##_put, \ +} + +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = { + PCSP_MIXER_CONTROL(enable, "Master Playback Switch"), + PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"), + PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), +}; + +int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip) +{ + struct snd_card *card = chip->card; + int i, err; + + for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) { + err = snd_ctl_add(card, + snd_ctl_new1(snd_pcsp_controls + i, + chip)); + if (err < 0) + return err; + } + + strcpy(card->mixername, "PC-Speaker"); + + return 0; +} From ae092c9ede515bd6864b44efc2d83135bd3c535b Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Mon, 3 Mar 2008 17:19:45 +0100 Subject: [PATCH 094/250] [ALSA] soc - Add Invert Switch for ROUT2 GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2 needs to be inverted. This patch adds a mixer control for this. Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8753.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 02882758415a..76a5c7b05dfb 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC", "Channel Swap"}; +static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"}; static const struct soc_enum wm8753_enum[] = { SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), @@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel), +SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase), }; @@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), SOC_ENUM("ADC Data Select", wm8753_enum[27]), +SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; /* add non dapm controls */ From 24982c5f7feca2f4d1b0b562a28b767d93a01ce0 Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Tue, 4 Mar 2008 10:08:58 +0100 Subject: [PATCH 095/250] [ALSA] hda_intel needs dma-mapping.h sparc32: sound/pci/hda/hda_intel.c: In function 'azx_create': sound/pci/hda/hda_intel.c:1838: error: 'DMA_64BIT_MASK' undeclared (first use in this function) sound/pci/hda/hda_intel.c:1838: error: (Each undeclared identifier is reported only once sound/pci/hda/hda_intel.c:1838: error: for each function it appears in.) Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c495ca012941..48677f36f5e4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include From 9bf8e7ddeaf57f1ec534014c447705ad31d5d721 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Mon, 3 Mar 2008 15:32:18 -0800 Subject: [PATCH 096/250] [ALSA] sound: replace remaining __FUNCTION__ occurences __FUNCTION__ is gcc-specific, use __func__ Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 8 ++++---- sound/core/init.c | 2 +- sound/isa/sb/sb16_csp.c | 28 ++++++++++++++-------------- sound/isa/sb/sb_common.c | 6 +++--- sound/oss/trident.h | 2 +- sound/oss/vwsnd.c | 6 +++--- sound/pci/ad1889.c | 4 ++-- sound/pci/als300.c | 4 ++-- sound/pci/azt3328.c | 4 ++-- sound/pci/intel8x0.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 8 ++++---- sound/soc/s3c24xx/s3c24xx-i2s.c | 20 ++++++++++---------- sound/soc/s3c24xx/s3c24xx-pcm.c | 28 ++++++++++++++-------------- sound/soc/soc-dapm.c | 2 +- 14 files changed, 62 insertions(+), 62 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 8704e2825b10..490729799e59 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -72,7 +72,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1) <= 0 && !((GSR | gsr_bits) & GSR_SDONE)) { printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); val = -1; goto out; } @@ -104,7 +104,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigne if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1) <= 0 && !((GSR | gsr_bits) & GSR_CDONE)) printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); mutex_unlock(&car_mutex); } @@ -131,7 +131,7 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); /* let's try warm reset */ gsr_bits = 0; @@ -150,7 +150,7 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); } GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); diff --git a/sound/core/init.c b/sound/core/init.c index e3338d6071ef..f045f7db3ab4 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -254,7 +254,7 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) if (likely(df)) return df->disconnected_f_op->release(inode, file); - panic("%s(%p, %p) failed!", __FUNCTION__, inode, file); + panic("%s(%p, %p) failed!", __func__, inode, file); } static unsigned int snd_disconnect_poll(struct file * file, poll_table * wait) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index bed29ca22239..f3fd7b4f4668 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -331,7 +331,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; if ((file_h.name != RIFF_HEADER) || (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) { - snd_printd("%s: Invalid RIFF header\n", __FUNCTION__); + snd_printd("%s: Invalid RIFF header\n", __func__); return -EINVAL; } data_ptr += sizeof(file_h); @@ -340,7 +340,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; if (item_type != CSP__HEADER) { - snd_printd("%s: Invalid RIFF file type\n", __FUNCTION__); + snd_printd("%s: Invalid RIFF file type\n", __func__); return -EINVAL; } data_ptr += sizeof (item_type); @@ -395,7 +395,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; if (code_h.name != MAIN_HEADER) { - snd_printd("%s: Missing 'main' microcode\n", __FUNCTION__); + snd_printd("%s: Missing 'main' microcode\n", __func__); return -EINVAL; } data_ptr += sizeof(code_h); @@ -439,7 +439,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, p->acc_format = p->acc_width = p->acc_rates = 0; p->mode = 0; snd_printd("%s: Unsupported CSP codec type: 0x%04x\n", - __FUNCTION__, + __func__, le16_to_cpu(funcdesc_h.VOC_type)); return -EINVAL; } @@ -458,7 +458,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return 0; } } - snd_printd("%s: Function #%d not found\n", __FUNCTION__, info.func_req); + snd_printd("%s: Function #%d not found\n", __func__, info.func_req); return -EINVAL; } @@ -612,7 +612,7 @@ static int get_version(struct snd_sb *chip) static int snd_sb_csp_check_version(struct snd_sb_csp * p) { if (p->version < 0x10 || p->version > 0x1f) { - snd_printd("%s: Invalid CSP version: 0x%x\n", __FUNCTION__, p->version); + snd_printd("%s: Invalid CSP version: 0x%x\n", __func__, p->version); return 1; } return 0; @@ -631,7 +631,7 @@ static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int spin_lock_irqsave(&p->chip->reg_lock, flags); snd_sbdsp_command(p->chip, 0x01); /* CSP download command */ if (snd_sbdsp_get_byte(p->chip)) { - snd_printd("%s: Download command failed\n", __FUNCTION__); + snd_printd("%s: Download command failed\n", __func__); goto __fail; } /* Send CSP low byte (size - 1) */ @@ -658,7 +658,7 @@ static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int udelay (10); } if (status != 0x55) { - snd_printd("%s: Microcode initialization failed\n", __FUNCTION__); + snd_printd("%s: Microcode initialization failed\n", __func__); goto __fail; } } else { @@ -824,19 +824,19 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel unsigned long flags; if (!(p->running & (SNDRV_SB_CSP_ST_LOADED | SNDRV_SB_CSP_ST_AUTO))) { - snd_printd("%s: Microcode not loaded\n", __FUNCTION__); + snd_printd("%s: Microcode not loaded\n", __func__); return -ENXIO; } if (p->running & SNDRV_SB_CSP_ST_RUNNING) { - snd_printd("%s: CSP already running\n", __FUNCTION__); + snd_printd("%s: CSP already running\n", __func__); return -EBUSY; } if (!(sample_width & p->acc_width)) { - snd_printd("%s: Unsupported PCM sample width\n", __FUNCTION__); + snd_printd("%s: Unsupported PCM sample width\n", __func__); return -EINVAL; } if (!(channels & p->acc_channels)) { - snd_printd("%s: Invalid number of channels\n", __FUNCTION__); + snd_printd("%s: Invalid number of channels\n", __func__); return -EINVAL; } @@ -858,11 +858,11 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel s_type |= 0x22; /* 00dX 00dX (d = 1 if 8 bit samples) */ if (set_codec_parameter(p->chip, 0x81, s_type)) { - snd_printd("%s: Set sample type command failed\n", __FUNCTION__); + snd_printd("%s: Set sample type command failed\n", __func__); goto __fail; } if (set_codec_parameter(p->chip, 0x80, 0x00)) { - snd_printd("%s: Codec start command failed\n", __FUNCTION__); + snd_printd("%s: Codec start command failed\n", __func__); goto __fail; } p->run_width = sample_width; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index d63c1af550de..b432d9ae874b 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -51,7 +51,7 @@ int snd_sbdsp_command(struct snd_sb *chip, unsigned char val) outb(val, SBP(chip, COMMAND)); return 1; } - snd_printd("%s [0x%lx]: timeout (0x%x)\n", __FUNCTION__, chip->port, val); + snd_printd("%s [0x%lx]: timeout (0x%x)\n", __func__, chip->port, val); return 0; } @@ -68,7 +68,7 @@ int snd_sbdsp_get_byte(struct snd_sb *chip) return val; } } - snd_printd("%s [0x%lx]: timeout\n", __FUNCTION__, chip->port); + snd_printd("%s [0x%lx]: timeout\n", __func__, chip->port); return -ENODEV; } @@ -87,7 +87,7 @@ int snd_sbdsp_reset(struct snd_sb *chip) else break; } - snd_printdd("%s [0x%lx] failed...\n", __FUNCTION__, chip->port); + snd_printdd("%s [0x%lx] failed...\n", __func__, chip->port); return -ENODEV; } diff --git a/sound/oss/trident.h b/sound/oss/trident.h index 4713b49fc91d..ff30a1d7c2f1 100644 --- a/sound/oss/trident.h +++ b/sound/oss/trident.h @@ -322,7 +322,7 @@ enum miscint_bits { #define VALIDATE_MAGIC(FOO,MAG) \ ({ \ if (!(FOO) || (FOO)->magic != MAG) { \ - printk(invalid_magic,__FUNCTION__); \ + printk(invalid_magic,__func__); \ return -ENXIO; \ } \ }) diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index d25249a932bf..2c5aaa58046d 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -194,11 +194,11 @@ static void dbgassert(const char *fcn, int line, const char *expr) * DBGRV - debug print function return when verbose */ -#define ASSERT(e) ((e) ? (void) 0 : dbgassert(__FUNCTION__, __LINE__, #e)) +#define ASSERT(e) ((e) ? (void) 0 : dbgassert(__func__, __LINE__, #e)) #define DBGDO(x) x #define DBGX(fmt, args...) (in_interrupt() ? 0 : printk(KERN_ERR fmt, ##args)) -#define DBGP(fmt, args...) (DBGX("%s: " fmt, __FUNCTION__ , ##args)) -#define DBGE(fmt, args...) (DBGX("%s" fmt, __FUNCTION__ , ##args)) +#define DBGP(fmt, args...) (DBGX("%s: " fmt, __func__ , ##args)) +#define DBGE(fmt, args...) (DBGX("%s" fmt, __func__ , ##args)) #define DBGC(rtn) (DBGP("calling %s\n", rtn)) #define DBGR() (DBGP("returning\n")) #define DBGXV(fmt, args...) (shut_up ? 0 : DBGX(fmt, ##args)) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index a66d5150bb7a..1edb6448946d 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -264,10 +264,10 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip) mdelay(1); if (!retry) { snd_printk(KERN_ERR PFX "[%s] Link is not ready.\n", - __FUNCTION__); + __func__); return -EIO; } - ad1889_debug("[%s] ready after %d ms\n", __FUNCTION__, 400 - retry); + ad1889_debug("[%s] ready after %d ms\n", __func__, 400 - retry); return 0; } diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 0e990a735821..8df6824b51cd 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -92,8 +92,8 @@ #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) -#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __FUNCTION__) -#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __FUNCTION__) +#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) +#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else #define snd_als300_dbgcalls(format, args...) #define snd_als300_dbgcallenter() diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 4e71a55120a0..be87d3113ee4 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -157,8 +157,8 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) -#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __FUNCTION__) -#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __FUNCTION__) +#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) +#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 07782ba9c74d..47485afcab5f 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2670,7 +2670,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip) t = stop_time.tv_sec - start_time.tv_sec; t *= 1000000; t += stop_time.tv_usec - start_time.tv_usec; - printk(KERN_INFO "%s: measured %lu usecs\n", __FUNCTION__, t); + printk(KERN_INFO "%s: measured %lu usecs\n", __func__, t); if (t == 0) { snd_printk(KERN_ERR "?? calculation error..\n"); return; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e17379998802..1092d58e8523 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -87,7 +87,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); if (!((GSR | gsr_bits) & GSR_SDONE)) { printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); val = -1; goto out; } @@ -127,7 +127,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1); if (!((GSR | gsr_bits) & GSR_CDONE)) printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); mutex_unlock(&car_mutex); } @@ -151,7 +151,7 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR |= GCR_SDONE_IE|GCR_CDONE_IE; @@ -178,7 +178,7 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR |= GCR_SDONE_IE|GCR_CDONE_IE; diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0a3c630951be..301002cd3fc8 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -89,7 +89,7 @@ static void s3c24xx_snd_txctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -134,7 +134,7 @@ static void s3c24xx_snd_rxctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -182,7 +182,7 @@ static int s3c24xx_snd_lrsync(void) u32 iiscon; unsigned long timeout = jiffies + msecs_to_jiffies(5); - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); while (1) { iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -201,7 +201,7 @@ static int s3c24xx_snd_lrsync(void) */ static inline int s3c24xx_snd_is_clkmaster(void) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; } @@ -214,7 +214,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); DBG("hw_params r: IISMOD: %lx \n", iismod); @@ -250,7 +250,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; @@ -278,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) { int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -320,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod &= ~S3C2440_IISMOD_MPLL; @@ -346,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, { u32 reg; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (div_id) { case S3C24XX_DIV_BCLK: @@ -381,7 +381,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); if (s3c24xx_i2s.regs == NULL) diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 29a6c82f873a..40112e2b1ecb 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -88,7 +88,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) dma_addr_t pos = prtd->dma_pos; int ret; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); while (prtd->dma_loaded < prtd->dma_limit) { unsigned long len = prtd->dma_period; @@ -98,7 +98,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; DBG(KERN_DEBUG "%s: corrected dma len %ld\n", - __FUNCTION__, len); + __func__, len); } ret = s3c2410_dma_enqueue(prtd->params->channel, @@ -123,7 +123,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, struct snd_pcm_substream *substream = dev_id; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) return; @@ -152,7 +152,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, unsigned long totbytes = params_buffer_bytes(params); int ret=0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -200,7 +200,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); /* TODO - do we need to ensure DMA flushed */ snd_pcm_set_runtime_buffer(substream, NULL); @@ -218,7 +218,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -263,7 +263,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); spin_lock(&prtd->lock); @@ -301,7 +301,7 @@ static snd_pcm_uframes_t unsigned long res; dma_addr_t src, dst; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); spin_lock(&prtd->lock); s3c2410_dma_getposition(prtd->params->channel, &src, &dst); @@ -334,7 +334,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); @@ -353,7 +353,7 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (prtd) kfree(prtd); @@ -368,7 +368,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); return dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, @@ -394,7 +394,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -413,7 +413,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_dma_buffer *buf; int stream; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -437,7 +437,7 @@ static int s3c24xx_pcm_new(struct snd_card *card, { int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (!card->dev->dma_mask) card->dev->dma_mask = &s3c24xx_pcm_dmamask; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index abac6847ecad..4c64560493f5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1288,7 +1288,7 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, mutex_unlock(&codec->mutex); dapm_power_widgets(codec, event); - dump_dapm(codec, __FUNCTION__); + dump_dapm(codec, __func__); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); From fd2499f0ed765de3ab11c7fd6f37f9fbfaf059ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 4 Mar 2008 11:06:26 +0100 Subject: [PATCH 097/250] [ALSA] aw2 - Remove endian dependency Removed unnecessary dependency on the little-endianess. Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-tsl.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/aw2/aw2-tsl.h b/sound/pci/aw2/aw2-tsl.h index e8afaa0a468a..459b0311ea31 100644 --- a/sound/pci/aw2/aw2-tsl.h +++ b/sound/pci/aw2/aw2-tsl.h @@ -72,10 +72,6 @@ /* SD3: >-------<_4-L___>-------<_4-R___> */ /* WS4: -------\_______________/--------- */ -#ifdef __BIG_ENDIAN - /* TODO: not yet implemented */ -#else /* */ - static int tsl1[8] = { 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1, @@ -112,5 +108,3 @@ static int tsl2[8] = { 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS }; - -#endif /* */ From 368c7a95ea324b3f9728ba1c901ac119d409bf4e Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Tue, 4 Mar 2008 11:20:33 +0100 Subject: [PATCH 098/250] [ALSA] hda-codec - model for alc883 to support M720R There is no suitable model for M720R (ALSA bug#3781). This patch is to support HP jack-sensing and mixer. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 76 +++++++++++++++++++++++++++++++++++ 1 file changed, 76 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33c9505adbae..f321abd91e0c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -201,6 +201,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, + ALC883_CLEVO_M720R, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6645,6 +6646,33 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7178,6 +7206,20 @@ static struct hda_verb alc883_mitac_verbs[] = { { } /* end */ }; +static struct hda_verb alc883_clevo_m720r_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Int speaker */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7361,6 +7403,26 @@ static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) alc883_tagra_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_clevo_m720r_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_clevo_m720r_automute(codec); +} + static void alc883_haier_w66_automute(struct hda_codec *codec) { unsigned int present; @@ -7598,6 +7660,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", + [ALC883_CLEVO_M720R] = "clevo-m720r", [ALC883_AUTO] = "auto", }; @@ -7638,6 +7701,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), @@ -7780,6 +7844,18 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_CLEVO_M720R] = { + .mixers = { alc883_clevo_m720r_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720r_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_clevo_m720r_unsol_event, + .init_hook = alc883_clevo_m720r_automute, + }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs}, From a91605b86a234b01d2f99b38411ac30a40bdf067 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Tue, 4 Mar 2008 11:28:43 +0100 Subject: [PATCH 099/250] [ALSA] pcsp - clean ups - make pcsp_start_timer_tasklet static - remove redundant includes. is not available on all platforms. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 2 -- sound/drivers/pcsp/pcsp_lib.c | 4 +--- 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 34477286b394..264d2a56dcd2 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -11,9 +11,7 @@ #include #include #include - #include -#include #include #include "pcsp_input.h" #include "pcsp.h" diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 6bdcb89129d8..a302756eac7c 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -9,10 +9,8 @@ #include #include #include -#include #include #include -#include #include "pcsp.h" static int nforce_wa; @@ -45,7 +43,7 @@ static void pcsp_start_timer(unsigned long dummy) * The only solution I could find was to move the * hrtimer_start() into a tasklet. -stsp */ -DECLARE_TASKLET(pcsp_start_timer_tasklet, pcsp_start_timer, 0); +static DECLARE_TASKLET(pcsp_start_timer_tasklet, pcsp_start_timer, 0); enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) { From 8280823668d42ed9695da759047b86074ad14ba4 Mon Sep 17 00:00:00 2001 From: Pascal Terjan Date: Tue, 4 Mar 2008 11:33:28 +0100 Subject: [PATCH 100/250] [ALSA] ALC288 - Add NEC S970 to the quirk table NEC S970 has no sound in the internal speakers when autodetection is used. With targa-dig model, there is sound in the speakers and it gets correctly muted when pluging headphones. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f321abd91e0c..d2e81d0fcd0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7685,6 +7685,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG), From 52337310af443278ee84ec4b9adaee0037cc0e30 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Thu, 6 Mar 2008 11:01:16 +0100 Subject: [PATCH 101/250] [ALSA] pcsp: improve "enable" option handling Simplify init code. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 264d2a56dcd2..547005cb0942 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -31,7 +31,7 @@ MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); -MODULE_PARM_DESC(enable, "dummy"); +MODULE_PARM_DESC(enable, "Enable PC-Speaker sound."); struct snd_pcsp pcsp_chip; @@ -136,7 +136,13 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) static int __devinit alsa_card_pcsp_init(struct device *dev) { - int devnum = 0, cards = 0; + int err; + + err = snd_card_pcsp_probe(0, dev); + if (err) { + printk(KERN_ERR "PC-Speaker initialization failed.\n"); + return err; + } #ifdef CONFIG_DEBUG_PAGEALLOC /* Well, CONFIG_DEBUG_PAGEALLOC makes the sound horrible. Lets alert */ @@ -148,15 +154,6 @@ static int __devinit alsa_card_pcsp_init(struct device *dev) "and crackling noise.\n"); #endif - if (enable) { - if (snd_card_pcsp_probe(devnum, dev) >= 0) - cards++; - if (!cards) { - printk(KERN_ERR "PC-Speaker initialization failed.\n"); - return -ENODEV; - } - } - return 0; } @@ -168,6 +165,7 @@ static void __devexit alsa_card_pcsp_exit(struct snd_pcsp *chip) static int __devinit pcsp_probe(struct platform_device *dev) { int err; + err = pcspkr_input_init(&pcsp_chip.input_dev, &dev->dev); if (err < 0) return err; @@ -227,6 +225,8 @@ static struct platform_driver pcsp_platform_driver = { static int __init pcsp_init(void) { + if (!enable) + return -ENODEV; return platform_driver_register(&pcsp_platform_driver); } From d08cd58db9f4078bb9237fc6ff7104ed50d7804b Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Thu, 6 Mar 2008 11:01:44 +0100 Subject: [PATCH 102/250] [ALSA] pcsp: add description update ALSA-Configuration.txt Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 999b7048162a..80a0629f4917 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1613,6 +1613,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Power management is _not_ supported. + Module snd-pcsp + ----------------- + + Module for internal PC-Speaker. + + nforce_wa - enable NForce chipset workaround. Expect bad sound. + + This module supports system beeps, some kind of PCM playback and + even a few mixer controls. + Module snd-pcxhr ---------------- From fb97dc67397c1ee63eb0094c28fc9a5dcc0a83a6 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Thu, 6 Mar 2008 11:07:11 +0100 Subject: [PATCH 103/250] [ALSA] hda-codec - model for alc883 to support FUJITSU Pi2515 There is no suitable model for Pi2515. This model is to support it. ALSA bug#3800. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 85 +++++++++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2e81d0fcd0f..41bcbfd10598 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -202,6 +202,7 @@ enum { ALC888_6ST_DELL, ALC883_MITAC, ALC883_CLEVO_M720R, + ALC883_FUJITSU_PI2515, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6484,6 +6485,14 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { }, }; +static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Int Mic", 0x1 }, + }, +}; + #define alc883_mux_enum_info alc_mux_enum_info #define alc883_mux_enum_get alc_mux_enum_get /* ALC883 has the ALC882-type input selection */ @@ -6673,6 +6682,33 @@ static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7220,6 +7256,20 @@ static struct hda_verb alc883_clevo_m720r_verbs[] = { { } /* end */ }; +static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { + /* HP */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7423,6 +7473,26 @@ static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, alc883_clevo_m720r_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_2ch_fujitsu_pi2515_automute(codec); +} + static void alc883_haier_w66_automute(struct hda_codec *codec) { unsigned int present; @@ -7661,6 +7731,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M720R] = "clevo-m720r", + [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC883_AUTO] = "auto", }; @@ -7706,6 +7777,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), @@ -7950,6 +8022,19 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_mitac_unsol_event, .init_hook = alc883_mitac_automute, }, + [ALC883_FUJITSU_PI2515] = { + .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, + .init_verbs = { alc883_init_verbs, + alc883_2ch_fujitsu_pi2515_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_fujitsu_pi2515_capture_source, + .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, + .init_hook = alc883_2ch_fujitsu_pi2515_automute, + }, }; From 2de3c232a0050ee247ae6e97b055f39e15a08ee3 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Thu, 6 Mar 2008 11:09:09 +0100 Subject: [PATCH 104/250] [ALSA] hda-codec - model for cx20549 to support laptop HP530 Currently the model laptop-hpsense use the 0x12 as ExtMic, and use 0x14 as Internal IntMic. But the hp530 only have one ExtMic, the Pin widget is 0x14. In this patch, I changed the mixer item for them. I still reserved the IntMic item, it will be helpful if other machine may use this model. ALSA bug#3821. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 51 +++++++++++++++++++++++++++++++++- 1 file changed, 50 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e4fa9a35848f..c67613ff842d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -519,6 +519,14 @@ static struct hda_input_mux cxt5045_capture_source_benq = { } }; +static struct hda_input_mux cxt5045_capture_source_hp530 = { + .num_items = 2, + .items = { + { "ExtMic", 0x1 }, + { "IntMic", 0x2 }, + } +}; + /* turn on/off EAPD (+ mute HP) as a master switch */ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -647,6 +655,37 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x1, HDA_INPUT), + HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5045_hp_master_sw_put, + .private_value = 0x10, + }, + + {} +}; + static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -841,6 +880,7 @@ enum { CXT5045_LAPTOP_MICSENSE, CXT5045_LAPTOP_HPMICSENSE, CXT5045_BENQ, + CXT5045_LAPTOP_HP530, #ifdef CONFIG_SND_DEBUG CXT5045_TEST, #endif @@ -852,6 +892,7 @@ static const char *cxt5045_models[CXT5045_MODELS] = { [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", [CXT5045_BENQ] = "benq", + [CXT5045_LAPTOP_HP530] = "laptop-hp530", #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif @@ -865,7 +906,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -949,6 +990,14 @@ static int patch_cxt5045(struct hda_codec *codec) spec->num_mixers = 2; codec->patch_ops.init = cxt5045_init; break; + case CXT5045_LAPTOP_HP530: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; + spec->input_mux = &cxt5045_capture_source_hp530; + spec->num_init_verbs = 2; + spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; + spec->mixers[0] = cxt5045_mixers_hp530; + codec->patch_ops.init = cxt5045_init; + break; #ifdef CONFIG_SND_DEBUG case CXT5045_TEST: spec->input_mux = &cxt5045_test_capture_source; From 487145a1984b78e9b194aa44f8079cc019779a58 Mon Sep 17 00:00:00 2001 From: Peer Chen Date: Thu, 6 Mar 2008 15:15:11 +0100 Subject: [PATCH 105/250] [ALSA] hda_intel: Add the DIDs of nvidia MCP79 HD audio controller to hda_intel.c Add the Device IDs of nvidia MCP79 HD audio controller to hda_intel.c Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 48677f36f5e4..3d9f0bd63ba6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2071,6 +2071,10 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); From 7194cae62e92c5db8b87df1120fbf24f83f488f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Mar 2008 16:58:17 +0100 Subject: [PATCH 106/250] [ALSA] hda-codec - Fix dmics on ALC268 in auto configuration Fixed the handling of dmics on ALC268 in the auto-configuration mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++--------- 1 file changed, 18 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 41bcbfd10598..c67c32faa903 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6258,16 +6258,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc882_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC882_PIN_CD_NID) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); + unsigned int vref; + if (!nid) + continue; + vref = PIN_IN; + if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { + if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) & + AC_PINCAP_VREF_80) + vref = PIN_VREF80; } + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, vref); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); } } @@ -10125,6 +10130,10 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, case 0x1c: idx1 = 3; /* CD */ break; + case 0x12: + case 0x13: + idx1 = 6; /* digital mics */ + break; default: continue; } From 0ccb541c96e6d40844d00ec88fae734568bdd0bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Mar 2008 16:58:35 +0100 Subject: [PATCH 107/250] [ALSA] hda-codec - Add internal mic item for ALC268 acer model Added the internal mic as a capture source item for ALC268 acer model. Signed-off-by: Takashi Iwai Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c67c32faa903..7937d97219d4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9775,8 +9775,12 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { }; static struct hda_verb alc268_acer_verbs[] = { + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } @@ -9990,6 +9994,15 @@ static struct hda_input_mux alc268_capture_source = { }, }; +static struct hda_input_mux alc268_acer_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x6 }, + { "Line", 0x2 }, + }, +}; + #ifdef CONFIG_SND_DEBUG static struct snd_kcontrol_new alc268_test_mixer[] = { /* Volume widgets */ @@ -10332,7 +10345,7 @@ static struct alc_config_preset alc268_presets[] = { .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, + .input_mux = &alc268_acer_capture_source, .unsol_event = alc268_acer_unsol_event, .init_hook = alc268_acer_init_hook, }, From 21ac1f9934d33ea2ffa71d13fa2f6286127d3caf Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Tue, 4 Mar 2008 15:07:24 -0800 Subject: [PATCH 108/250] sound: Use BUG_ON if (...) BUG(); should be replaced with BUG_ON(...) when the test has no side-effects to allow a definition of BUG_ON that drops the code completely. The semantic patch that makes this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // @ disable unlikely @ expression E,f; @@ ( if (<... f(...) ...>) { BUG(); } | - if (unlikely(E)) { BUG(); } + BUG_ON(E); ) @@ expression E,f; @@ ( if (<... f(...) ...>) { BUG(); } | - if (E) { BUG(); } + BUG_ON(E); ) // Signed-off-by: Julia Lawall Signed-off-by: Andrew Morton --- sound/oss/trident.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/oss/trident.c b/sound/oss/trident.c index d6af9065d1c0..f43f91ef86c7 100644 --- a/sound/oss/trident.c +++ b/sound/oss/trident.c @@ -3076,8 +3076,7 @@ ali_ac97_get(struct trident_card *card, int secondary, u8 reg) u16 wcontrol; unsigned long flags; - if (!card) - BUG(); + BUG_ON(!card); address = ALI_AC97_READ; if (card->revision == ALI_5451_V02) { @@ -3148,8 +3147,7 @@ ali_ac97_set(struct trident_card *card, int secondary, u8 reg, u16 val) data = ((u32) val) << 16; - if (!card) - BUG(); + BUG_ON(!card); address = ALI_AC97_WRITE; mask = ALI_AC97_WRITE_ACTION | ALI_AC97_AUDIO_BUSY; @@ -3213,8 +3211,7 @@ ali_ac97_read(struct ac97_codec *codec, u8 reg) struct trident_card *card = NULL; /* Added by Matt Wu */ - if (!codec) - BUG(); + BUG_ON(!codec); card = (struct trident_card *) codec->private_data; @@ -3240,8 +3237,7 @@ ali_ac97_write(struct ac97_codec *codec, u8 reg, u16 val) struct trident_card *card; /* Added by Matt Wu */ - if (!codec) - BUG(); + BUG_ON(!codec); card = (struct trident_card *) codec->private_data; From b419f346994d4ba082244fb1327754bc839a4d8a Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Fri, 7 Mar 2008 11:57:51 +0100 Subject: [PATCH 109/250] [ALSA] HDA Codecs: add support for Toshiba Equium L30 This patch adds support for the Toshiba Equium L30 laptop and renames the mixer controls to match Laptop usages. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++-------------- 1 file changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7937d97219d4..3c15bdf4d2cd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12054,11 +12054,10 @@ static struct hda_input_mux alc861vd_capture_source = { }; static struct hda_input_mux alc861vd_dallas_capture_source = { - .num_items = 3, + .num_items = 2, .items = { - { "Front Mic", 0x0 }, - { "ATAPI Mic", 0x1 }, - { "Line In", 0x5 }, + { "Ext Mic", 0x0 }, + { "Int Mic", 0x1 }, }, }; @@ -12228,20 +12227,22 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { { } /* end */ }; -/* Pin assignment: Front=0x14, HP = 0x15, - * Front Mic=0x18, ATAPI Mic = 0x19, Line In = 0x1d +/* Pin assignment: Speaker=0x14, HP = 0x15, + * Ext Mic=0x18, Int Mic = 0x19, CD = 0x1c, PC Beep = 0x1d */ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -12542,6 +12543,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/ SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), From c67582b195fb5deb24808ebbafb41045f1a61425 Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Sat, 8 Mar 2008 11:07:26 +0100 Subject: [PATCH 110/250] [ALSA] at73c213: fix error checking for clk API The clk_round_rate() and clk_set_rate() will return int, so not store thier return value to unsigned long variable. This bug hides real error on these API. Signed-off-by: Atsushi Nemoto Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index b8860b26fc62..c1b0b9dff819 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -133,7 +133,8 @@ static struct snd_pcm_hardware snd_at73c213_playback_hw = { static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) { unsigned long ssc_rate = clk_get_rate(chip->ssc->clk); - unsigned long dac_rate_new, ssc_div, status; + unsigned long dac_rate_new, ssc_div; + int status; unsigned long ssc_div_max, ssc_div_min; int max_tries; From 4a295ca47424b48c993d5cea7c3fbeca75ddb608 Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Sat, 8 Mar 2008 11:08:32 +0100 Subject: [PATCH 111/250] [ALSA] at73c213: monaural support Add support for monaural playback to at73c213 driver. The sound will be apear on L-channel. Tested on AT91SAM9260-EK. Signed-off-by: Atsushi Nemoto Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index c1b0b9dff819..1355fe0c6674 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -118,7 +118,7 @@ static struct snd_pcm_hardware snd_at73c213_playback_hw = { .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, /* Replaced by chip->bitrate later. */ .rate_max = 50000, /* Replaced by chip->bitrate later. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 64 * 1024 - 1, .period_bytes_min = 512, @@ -229,6 +229,14 @@ static int snd_at73c213_pcm_close(struct snd_pcm_substream *substream) static int snd_at73c213_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + int channels = params_channels(hw_params); + int val; + + val = ssc_readl(chip->ssc->regs, TFMR); + val = SSC_BFINS(TFMR_DATNB, channels - 1, val); + ssc_writel(chip->ssc->regs, TFMR, val); + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -250,10 +258,12 @@ static int snd_at73c213_pcm_prepare(struct snd_pcm_substream *substream) ssc_writel(chip->ssc->regs, PDC_TPR, (long)runtime->dma_addr); - ssc_writel(chip->ssc->regs, PDC_TCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TCR, + runtime->period_size * runtime->channels); ssc_writel(chip->ssc->regs, PDC_TNPR, (long)runtime->dma_addr + block_size); - ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TNCR, + runtime->period_size * runtime->channels); return 0; } @@ -376,7 +386,8 @@ static irqreturn_t snd_at73c213_interrupt(int irq, void *dev_id) ssc_writel(chip->ssc->regs, PDC_TNPR, (long)runtime->dma_addr + offset); - ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TNCR, + runtime->period_size * runtime->channels); retval = IRQ_HANDLED; } From 85bfb8fe5d26b316ccd4892d1834778ec5fc17c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Mar 2008 11:21:30 +0100 Subject: [PATCH 112/250] [ALSA] release 1.0.16 Signed-off-by: Takashi Iwai --- include/sound/version.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/version.h b/include/sound/version.h index fac66c49445a..ed6fb2eb1eac 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by alsa/ksync script. */ -#define CONFIG_SND_VERSION "1.0.16rc2" -#define CONFIG_SND_DATE " (Thu Jan 31 16:40:16 2008 UTC)" +#define CONFIG_SND_VERSION "1.0.16" +#define CONFIG_SND_DATE "" From 7747ecceb523d7b00c8dfe568154d0b4e70e0800 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 10 Mar 2008 11:30:04 +0100 Subject: [PATCH 113/250] [ALSA] hda: Reorganized DAC outputs Changed so that internal speakers point to the Front mixer instead of Surround. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6c85e7e81034..bf6142f54539 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -538,8 +538,8 @@ static struct hda_verb dell_m6_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* setup audio connections */ - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, From ee7a9c7c2eb5cd09c15824323eac4cd95e2d18a8 Mon Sep 17 00:00:00 2001 From: Michael Gruber Date: Mon, 10 Mar 2008 11:30:59 +0100 Subject: [PATCH 114/250] [ALSA] hda-intel - Fix microphone capture with ALC880 F1734 model The default capture source should be the mic which is 0x01 on this model. In addition to that the change to VREF50 allows for higher capture volume. Signed-off-by: Michael Gruber Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c15bdf4d2cd..bfb336de1dcb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1927,6 +1927,7 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, * HP = 0x14, speaker-out = 0x15, mic = 0x18 */ static struct hda_verb alc880_pin_f1734_init_verbs[] = { + {0x07, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -1939,7 +1940,7 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = { {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, From 91662577979cadf945cd3ffc4c470e5b91378370 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Mar 2008 12:19:12 +0100 Subject: [PATCH 115/250] [ALSA] aw2 - Rename aw2-tsl.h to aw2-tsl.c aw2-tsl.h should be rather a C file to be included since it's referred only in aw2-saa6146.c and includes a table data. Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-saa7146.c | 3 ++- sound/pci/aw2/{aw2-tsl.h => aw2-tsl.c} | 0 2 files changed, 2 insertions(+), 1 deletion(-) rename sound/pci/aw2/{aw2-tsl.h => aw2-tsl.c} (100%) diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index f20f213489a3..6a3891ab69dd 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -35,10 +35,11 @@ #include #include -#include "aw2-tsl.h" #include "saa7146.h" #include "aw2-saa7146.h" +#include "aw2-tsl.c" + #define WRITEREG(value, addr) writel((value), chip->base_addr + (addr)) #define READREG(addr) readl(chip->base_addr + (addr)) diff --git a/sound/pci/aw2/aw2-tsl.h b/sound/pci/aw2/aw2-tsl.c similarity index 100% rename from sound/pci/aw2/aw2-tsl.h rename to sound/pci/aw2/aw2-tsl.c From 9f2f0f7c4e997a74ff9fb8e2e2ed0daa21962e97 Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Tue, 11 Mar 2008 08:15:30 +0100 Subject: [PATCH 116/250] [ALSA] at73c213: remove redundant private_free routine snd_pcm_lib_preallocate_free_for_all() is called from snd_pcm_free() just after calling the private_free routine. So there should be no need to call it in driver's private_free routine. Signed-off-by: Atsushi Nemoto Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 1355fe0c6674..aa571152f034 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -325,15 +325,6 @@ static struct snd_pcm_ops at73c213_playback_ops = { .pointer = snd_at73c213_pcm_pointer, }; -static void snd_at73c213_pcm_free(struct snd_pcm *pcm) -{ - struct snd_at73c213 *chip = snd_pcm_chip(pcm); - if (chip->pcm) { - snd_pcm_lib_preallocate_free_for_all(chip->pcm); - chip->pcm = NULL; - } -} - static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device) { struct snd_pcm *pcm; @@ -345,7 +336,6 @@ static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device) goto out; pcm->private_data = chip; - pcm->private_free = snd_at73c213_pcm_free; pcm->info_flags = SNDRV_PCM_INFO_BLOCK_TRANSFER; strcpy(pcm->name, "at73c213"); chip->pcm = pcm; From bb9f76cd5909b9da6b4d31b55a4086cc35614fe0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Mar 2008 12:51:09 +0100 Subject: [PATCH 117/250] [ALSA] hda-codec - Improve ALC262 ultra model Improved ALC262 ultra model for Samsung Q1 Ultra series. - clean up mixers - support of input from HP jack as a mic - add quirk for Q1 EL Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 132 ++++++++++++++++++++++++---------- 1 file changed, 93 insertions(+), 39 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bfb336de1dcb..3965b7644ad5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8870,59 +8870,72 @@ static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { /* Samsung Q1 Ultra Vista model setup */ static struct snd_kcontrol_new alc262_ultra_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Mic Boost", 0x15, 0, HDA_INPUT), { } /* end */ }; static struct hda_verb alc262_ultra_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + /* output mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* speaker */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - /* Mic is on Node 0x19 */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x24, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + /* internal mic */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* ADC, choose mic */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)}, {} }; -static struct hda_input_mux alc262_ultra_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x1 }, - }, -}; - /* mute/unmute internal speaker according to the hp jack and mute state */ static void alc262_ultra_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int mute; - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); + mute = 0; + /* auto-mute only when HP is used as HP */ + if (!spec->cur_mux[0]) { + unsigned int present; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + if (spec->jack_present) + mute = HDA_AMP_MUTE; } + /* mute/unmute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + /* mute/unmute HP */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE); } /* unsolicited event for HP jack sensing */ @@ -8934,6 +8947,45 @@ static void alc262_ultra_unsol_event(struct hda_codec *codec, alc262_ultra_automute(codec); } +static struct hda_input_mux alc262_ultra_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "Headphone", 0x7 }, + }, +}; + +static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int ret; + + ret = alc882_mux_enum_put(kcontrol, ucontrol); + if (!ret) + return 0; + /* reprogram the HP pin as mic or HP according to the input source */ + snd_hda_codec_write_cache(codec, 0x15, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->cur_mux[0] ? PIN_VREF80 : PIN_HP); + alc262_ultra_automute(codec); /* mute/unmute HP */ + return ret; +} + +static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = alc882_mux_enum_info, + .get = alc882_mux_enum_get, + .put = alc262_ultra_mux_enum_put, + }, + { } /* end */ +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -9384,6 +9436,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), @@ -9533,15 +9586,16 @@ static struct alc_config_preset alc262_presets[] = { .init_hook = alc262_hippo_automute, }, [ALC262_ULTRA] = { - .mixers = { alc262_ultra_mixer }, - .init_verbs = { alc262_init_verbs, alc262_ultra_verbs }, + .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer }, + .init_verbs = { alc262_ultra_verbs }, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC262_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_ultra_capture_source, + .adc_nids = alc262_adc_nids, /* ADC0 */ + .capsrc_nids = alc262_capsrc_nids, + .num_adc_nids = 1, /* single ADC */ .unsol_event = alc262_ultra_unsol_event, .init_hook = alc262_ultra_automute, }, From 1bc1f30565b561bafc51725fce336aec59029437 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Wed, 12 Mar 2008 13:12:15 +0100 Subject: [PATCH 118/250] [ALSA] pcsp: locking fix pcsp: locking fix. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 5 ++--- sound/drivers/pcsp/pcsp_lib.c | 2 ++ 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 547005cb0942..ac57e87d01bc 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -191,11 +191,10 @@ static int __devexit pcsp_remove(struct platform_device *dev) static void pcsp_stop_beep(struct snd_pcsp *chip) { - unsigned long flags; - spin_lock_irqsave(&chip->substream_lock, flags); + spin_lock_irq(&chip->substream_lock); if (!chip->playback_substream) pcspkr_stop_sound(); - spin_unlock_irqrestore(&chip->substream_lock, flags); + spin_unlock_irq(&chip->substream_lock); } static int pcsp_suspend(struct platform_device *dev, pm_message_t state) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index a302756eac7c..54253e9b4b02 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -305,7 +305,9 @@ static int snd_pcsp_playback_open(struct snd_pcm_substream *substream) return -EBUSY; } runtime->hw = snd_pcsp_playback; + spin_lock_irq(&chip->substream_lock); chip->playback_substream = substream; + spin_unlock_irq(&chip->substream_lock); return 0; } From d654a660355f9dc30d3a6bf1493d32363bde8570 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 14 Mar 2008 08:46:51 +0100 Subject: [PATCH 119/250] [ALSA] hda: 92HD73xxx distortion fix Fixed issue on some laptops that if the Master mixer and DAC mixers are turned all the way up that will cause distortion. This is fixed by limiting the max volume with the volume knob nid. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bf6142f54539..b75bf3475653 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -534,6 +534,25 @@ static struct hda_verb stac92hd73xx_6ch_core_init[] = { {} }; +static struct hda_verb dell_eq_core_init[] = { + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, + /* setup audio connections */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + static struct hda_verb dell_m6_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -3460,17 +3479,19 @@ again: switch (spec->board_config) { case STAC_DELL_M6: - spec->init = dell_m6_core_init; + spec->init = dell_eq_core_init; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; break; - case 0x10280254: /* Digital Mics */ - case 0x10280255: - case 0x10280271: + case 0x10280271: /* Digital Mics */ case 0x10280272: + spec->init = dell_m6_core_init; + /* fall-through */ + case 0x10280254: + case 0x10280255: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; break; From 2626a263ffc2369499442933b1c313de0a066ede Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Mar 2008 09:18:32 +0100 Subject: [PATCH 120/250] [ALSA] hda-codec - Fix orphan Headphone controls in STAC codecs Currently, the headphone controls are created as Master wrongly in some cases, and this prevents the virtual master controls. The patch fixes the problem by simply using "Headphone" always for headphone controls. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b75bf3475653..b8e69a1b93f0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2509,12 +2509,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, return err; } if (spec->multiout.hp_nid) { - const char *pfx; - if (old_num_dacs == spec->multiout.num_dacs) - pfx = "Master"; - else - pfx = "Headphone"; - err = create_controls(spec, pfx, spec->multiout.hp_nid, 3); + err = create_controls(spec, "Headphone", + spec->multiout.hp_nid, 3); if (err < 0) return err; } From 7055ad8a996b2b77f12242109c0b5bacc237e824 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Fri, 14 Mar 2008 12:52:20 +0100 Subject: [PATCH 121/250] [ALSA] hda-codec - Fix ALC662 DAC mixer mutes Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03, 0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to mute?). The result is that currently mute for "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch" and "LFE Playback Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard with 3stack-6ch model). The solution I found for this is to mute the proper inputs in 0x0c, 0x0d, 0x0e audio mixers. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3965b7644ad5..8eb64dbfc78f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13139,13 +13139,13 @@ static struct hda_channel_mode alc662_5stack_modes[2] = { static struct snd_kcontrol_new alc662_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), /*Input mixer control */ @@ -13162,7 +13162,7 @@ static struct snd_kcontrol_new alc662_base_mixer[] = { static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -13179,13 +13179,13 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), From 3da23cac3d6b93803b8c381a755870cbafcd3212 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Fri, 14 Mar 2008 12:52:59 +0100 Subject: [PATCH 122/250] [ALSA] hda-codec - Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8eb64dbfc78f..c85da5766d39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13534,6 +13534,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), From ee9d6b9a30ae83f15fe8c8d2337ebc0a38151d38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Mar 2008 15:52:20 +0100 Subject: [PATCH 123/250] [ALSA] hda-intel - Fix power-off hang on ASUS P5AD2 The hda-intel driver has a problem at power-off on ASUS P5AD2. It's caused when the position-buffer is enabled -- most likely a hardware-specific problem. This patch adds a quirk to avoid the unnecessary enablement of position-buffer. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3d9f0bd63ba6..9a20fb1bef4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1059,9 +1059,12 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr)); /* enable the position buffer */ - if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) - azx_writel(chip, DPLBASE, - (u32)chip->posbuf.addr |ICH6_DPLBASE_ENABLE); + if (chip->position_fix == POS_FIX_POSBUF || + chip->position_fix == POS_FIX_AUTO) { + if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) + azx_writel(chip, DPLBASE, + (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE); + } /* set the interrupt enable bits in the descriptor control register */ azx_sd_writel(azx_dev, SD_CTL, @@ -1707,6 +1710,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), + SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), {} }; From 5d9fab2d84963ec598810c54a67332decdd922a8 Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Fri, 14 Mar 2008 17:09:18 +0100 Subject: [PATCH 124/250] [ALSA] hda-codec - Fujitsu Lifebook port replicator/dock headphone jack sense The docking station headphone output had no audio and jack sense was not considered. Jack information from the laptop itself and the dock are combined, as the dock does not obscure the connector. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 +++++++++++++++++++----------- 1 file changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c85da5766d39..bf729f518f87 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8725,7 +8725,8 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, /* * fujitsu model - * 0x14 = headphone/spdif-out, 0x15 = internal speaker + * 0x14 = headphone/spdif-out, 0x15 = internal speaker, + * 0x1b = port replicator headphone out */ #define ALC_HP_EVENT 0x37 @@ -8733,6 +8734,8 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, static struct hda_verb alc262_fujitsu_unsol_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {} }; @@ -8773,12 +8776,16 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present; + unsigned int present_int_hp, present_dock_hp; /* need to execute and sync at first */ snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + present_int_hp = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + snd_hda_codec_read(codec, 0x1B, 0, AC_VERB_SET_PIN_SENSE, 0); + present_dock_hp = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present_int_hp & 0x80000000) != 0; + spec->jack_present |= (present_dock_hp & 0x80000000) != 0; spec->sense_updated = 1; } if (spec->jack_present) { @@ -8820,12 +8827,13 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); + change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp ? 0 : HDA_AMP_MUTE); + change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp ? 0 : HDA_AMP_MUTE); + if (change) alc262_fujitsu_automute(codec, 0); return change; From c93f5a1eca1f6d662d791c14c469b6962e05a08f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Mar 2008 17:17:09 +0100 Subject: [PATCH 125/250] [ALSA] ice1724 - Fix the SPDIF input sample-rate on Juli@ AK4114 on Juli@ has the SPDIF input sample rate detection and causes errors when an incompatible sample rate is chosen. The patch adds the open hook to check the current rate and limit the hw constraints. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 8 ++++++++ sound/pci/ice1712/juli.c | 17 +++++++++++++++++ 2 files changed, 25 insertions(+) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f533850ec6e7..3bfd70577d7a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -970,6 +970,8 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->spdif.ops.open) + ice->spdif.ops.open(ice, substream); return 0; } @@ -980,6 +982,8 @@ static int snd_vt1724_playback_spdif_close(struct snd_pcm_substream *substream) if (PRO_RATE_RESET) snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); ice->playback_con_substream = NULL; + if (ice->spdif.ops.close) + ice->spdif.ops.close(ice, substream); return 0; } @@ -1002,6 +1006,8 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->spdif.ops.open) + ice->spdif.ops.open(ice, substream); return 0; } @@ -1012,6 +1018,8 @@ static int snd_vt1724_capture_spdif_close(struct snd_pcm_substream *substream) if (PRO_RATE_RESET) snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); ice->capture_con_substream = NULL; + if (ice->spdif.ops.close) + ice->spdif.ops.close(ice, substream); return 0; } diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index e8038c0ceb72..4550609b4d47 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -77,6 +77,22 @@ static unsigned char juli_ak4114_read(void *private_data, unsigned char reg) return snd_vt1724_read_i2c((struct snd_ice1712 *)private_data, AK4114_ADDR, reg); } +static void juli_spdif_in_open(struct snd_ice1712 *ice, + struct snd_pcm_substream *substream) +{ + struct juli_spec *spec = ice->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int rate; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return; + rate = snd_ak4114_external_rate(spec->ak4114); + if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) { + runtime->hw.rate_min = rate; + runtime->hw.rate_max = rate; + } +} + /* * AK4358 section */ @@ -210,6 +226,7 @@ static int __devinit juli_init(struct snd_ice1712 *ice) return err; } + ice->spdif.ops.open = juli_spdif_in_open; return 0; } From 5949d2443d96f054d9a32d31edddb0be836968c6 Mon Sep 17 00:00:00 2001 From: Joachim Foerster Date: Mon, 17 Mar 2008 08:40:12 +0100 Subject: [PATCH 126/250] [ALSA] [ML403-AC97CR] Remove duplicate snd_card_set_dev() We want to have snd_card_set_dev() in _probe(), but not a second one in snd_ml403_ac97cr_create(). Signed-off-by: Joachim Foerster Signed-off-by: Takashi Iwai --- sound/drivers/ml403-ac97cr.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 05a871aa7b81..b654007331de 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1191,8 +1191,6 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, return err; } - snd_card_set_dev(card, &pfdev->dev); - *rml403_ac97cr = ml403_ac97cr; return 0; } From 841b23d4d7b554c8d74fc9c34a701f85abc04875 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Mon, 17 Mar 2008 08:45:33 +0100 Subject: [PATCH 127/250] [ALSA] some fixes and cleanup for ICE1724 cards * removing the hack with NON_AKM ak4xxx type * support for card-specific flags in ak4114_stats * definition of the flags for corresponding cards Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- include/sound/ak4114.h | 1 + include/sound/ak4xxx-adda.h | 2 +- sound/i2c/other/ak4114.c | 2 +- sound/i2c/other/ak4xxx-adda.c | 16 ++++++++-------- sound/pci/ice1712/prodigy192.c | 33 ++++++++++----------------------- sound/pci/ice1712/revo.c | 4 ++++ 6 files changed, 25 insertions(+), 33 deletions(-) diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h index 4e80d3fe7381..d293d36a66b8 100644 --- a/include/sound/ak4114.h +++ b/include/sound/ak4114.h @@ -182,6 +182,7 @@ struct ak4114 { unsigned char rcs0; unsigned char rcs1; struct delayed_work work; + unsigned int check_flags; void *change_callback_private; void (*change_callback)(struct ak4114 *ak4114, unsigned char c0, unsigned char c1); }; diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h index 6153b91cdc3e..891cf1aea8b1 100644 --- a/include/sound/ak4xxx-adda.h +++ b/include/sound/ak4xxx-adda.h @@ -68,7 +68,7 @@ struct snd_akm4xxx { enum { SND_AK4524, SND_AK4528, SND_AK4529, SND_AK4355, SND_AK4358, SND_AK4381, - SND_AK5365, NON_AKM + SND_AK5365 } type; /* (array) information of combined codecs */ diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 9a90e830c423..d20d893b3b60 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -612,7 +612,7 @@ static void ak4114_stats(struct work_struct *work) struct ak4114 *chip = container_of(work, struct ak4114, work.work); if (!chip->init) - snd_ak4114_check_rate_and_errors(chip, 0); + snd_ak4114_check_rate_and_errors(chip, chip->check_flags); schedule_delayed_work(&chip->work, HZ / 10); } diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 35fbbf2cb9fa..288926d2e205 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -70,7 +70,8 @@ static void ak4524_reset(struct snd_akm4xxx *ak, int state) } /* reset procedure for AK4355 and AK4358 */ -static void ak4355_reset(struct snd_akm4xxx *ak, int state) +static void ak435X_reset(struct snd_akm4xxx *ak, int state, + unsigned char total_regs) { unsigned char reg; @@ -78,7 +79,7 @@ static void ak4355_reset(struct snd_akm4xxx *ak, int state) snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */ return; } - for (reg = 0x00; reg < 0x0b; reg++) + for (reg = 0x00; reg < total_regs; reg++) if (reg != 0x01) snd_akm4xxx_write(ak, 0, reg, snd_akm4xxx_get(ak, 0, reg)); @@ -118,8 +119,10 @@ void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state) /* FIXME: needed for ak4529? */ break; case SND_AK4355: + ak435X_reset(ak, state, 0x0b); + break; case SND_AK4358: - ak4355_reset(ak, state); + ak435X_reset(ak, state, 0x10); break; case SND_AK4381: ak4381_reset(ak, state); @@ -292,11 +295,6 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) case SND_AK5365: /* FIXME: any init sequence? */ return; - case NON_AKM: - /* fake value for non-akm codecs using akm infrastructure - * (e.g. of ice1724) - certainly FIXME - */ - return; default: snd_BUG(); return; @@ -374,6 +372,8 @@ static int put_ak_reg(struct snd_kcontrol *kcontrol, int addr, nval = mask - nval; if (AK_GET_NEEDSMSB(kcontrol->private_value)) nval |= 0x80; + /* printk(KERN_DEBUG "DEBUG - AK writing reg: chip %x addr %x, + nval %x\n", chip, addr, nval); */ snd_akm4xxx_write(ak, chip, addr, nval); return 1; } diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 48cf40a8f32a..25ceb67a9c16 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -319,12 +319,11 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, /* * Handler for setting correct codec rate - called when rate change is detected */ -static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) { unsigned char old, new; int idx; unsigned char changed[7]; - struct snd_ice1712 *ice = ak->private_data[0]; struct prodigy192_spec *spec = ice->spec; if (rate == 0) /* no hint - S/PDIF input is master, simply return */ @@ -357,16 +356,6 @@ static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) mutex_unlock(&spec->mute_mutex); } -/* using akm infrastructure for setting rate of the codec */ -static struct snd_akm4xxx akmlike_stac9460 __devinitdata = { - .type = NON_AKM, /* special value */ - .num_adcs = 6, /* not used in any way, just for completeness */ - .num_dacs = 2, - .ops = { - .set_rate_val = stac9460_set_rate_val - } -}; - static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -642,12 +631,19 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice) 0x41, 0x02, 0x2c, 0x00, 0x00 }; struct prodigy192_spec *spec = ice->spec; + int err; - return snd_ak4114_create(ice->card, + err = snd_ak4114_create(ice->card, prodigy192_ak4114_read, prodigy192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &spec->ak4114); + if (err < 0) + return err; + /* AK4114 in Prodigy192 cannot detect external rate correctly. + * No reason to stop capture stream due to incorrect checks */ + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; + return 0; } static void stac9460_proc_regs_read(struct snd_info_entry *entry, @@ -743,7 +739,6 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) }; const unsigned short *p; int err = 0; - struct snd_akm4xxx *ak; struct prodigy192_spec *spec; /* prodigy 192 */ @@ -761,15 +756,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) p = stac_inits_prodigy; for (; *p != (unsigned short)-1; p += 2) stac9460_put(ice, p[0], p[1]); - /* reusing the akm codecs infrastructure, - * for setting rate on stac9460 */ - ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (!ak) - return -ENOMEM; - ice->akm_codecs = 1; - err = snd_ice1712_akm4xxx_init(ak, &akmlike_stac9460, NULL, ice); - if (err < 0) - return err; + ice->gpio.set_pro_rate = stac9460_set_rate_val; /* MI/ODI/O add on card with AK4114 */ if (prodigy192_miodio_exists(ice)) { diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 7c930cc05f1d..dba93d8efbe0 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -488,6 +488,10 @@ static int __devinit ap192_ak4114_init(struct snd_ice1712 *ice) ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &ak); + /* AK4114 in Revo cannot detect external rate correctly. + * No reason to stop capture stream due to incorrect checks */ + ak->check_flags = AK4114_CHECK_NO_RATE; + return 0; /* error ignored; it's no fatal error */ } From 5a220c868e395bc3662d13ad4c2a18769075af54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Mar 2008 09:59:32 +0100 Subject: [PATCH 128/250] [ALSA] usb-audio - Add a proper error check The error in check_hw_params_convention() has to be checked and handled properly. Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f48838a078cb..5c40c031dd57 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1762,8 +1762,10 @@ static int check_hw_params_convention(struct snd_usb_substream *subs) channels = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); rates = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - if (!channels || !rates) + if (!channels || !rates) { + err = -ENOMEM; goto __out; + } list_for_each(p, &subs->fmt_list) { struct audioformat *f; @@ -1916,7 +1918,10 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre 1000 * MIN_PACKS_URB, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); - if (check_hw_params_convention(subs)) { + err = check_hw_params_convention(subs); + if (err < 0) + return err; + else if (err) { hwc_debug("setting extra hw constraints...\n"); if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, From 43337ac0dec5fe507b3028d2ad4404c3f1a28034 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Mar 2008 10:16:37 +0100 Subject: [PATCH 129/250] [ALSA] ice1724 - Fix return codes in some pointis callbacks Fixed the return codes (1 for changed values) in put callbacks of pontis. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/pontis.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 4945c81e8a96..203cdc1bf8da 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -246,7 +246,7 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val wm_put(ice, WM_ADC_MUX, nval); } mutex_unlock(&ice->gpio_mutex); - return 0; + return change; } /* @@ -450,7 +450,7 @@ static int cs_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu change = 1; } mutex_unlock(&ice->gpio_mutex); - return 0; + return change; } From b32300a4ce78b4df92f33c455ffe644b36f0ff1a Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 17 Mar 2008 10:23:35 +0100 Subject: [PATCH 130/250] [ALSA] sound/pci: remove unused variable The variable is_capture is initialized but never used otherwise. The semantic patch that makes this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // @@ type T; identifier i; constant C; @@ ( extern T i; | - T i; <+... when != i - i = C; ...+> ) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 9d5bb76229a8..beed5818338c 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -846,7 +846,6 @@ static int pcxhr_open(struct snd_pcm_substream *subs) struct pcxhr_mgr *mgr = chip->mgr; struct snd_pcm_runtime *runtime = subs->runtime; struct pcxhr_stream *stream; - int is_capture; mutex_lock(&mgr->setup_mutex); @@ -856,12 +855,10 @@ static int pcxhr_open(struct snd_pcm_substream *subs) if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) { snd_printdd("pcxhr_open playback chip%d subs%d\n", chip->chip_idx, subs->number); - is_capture = 0; stream = &chip->playback_stream[subs->number]; } else { snd_printdd("pcxhr_open capture chip%d subs%d\n", chip->chip_idx, subs->number); - is_capture = 1; if (mgr->mono_capture) runtime->hw.channels_max = 1; else From f5e09ef0985ff01af6b4a12954840467f153a41c Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Mon, 17 Mar 2008 14:36:24 +0100 Subject: [PATCH 131/250] [ALSA] at73c213: Add constraints for periods value The interrupt handler always provide runtime->period_size data, so it works correctly only if buffer_size was a multiple of period_size. This patch fixes periodic click noise. Signed-off-by: Atsushi Nemoto Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index aa571152f034..09802e8a6fb8 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -210,7 +210,13 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) { struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int err; + /* ensure buffer_size is a multiple of period_size */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + return err; snd_at73c213_playback_hw.rate_min = chip->bitrate; snd_at73c213_playback_hw.rate_max = chip->bitrate; runtime->hw = snd_at73c213_playback_hw; From 9ecaedae0de906f0b5f8dbc09cd610f7056ba975 Mon Sep 17 00:00:00 2001 From: Mariusz Kozlowski Date: Tue, 18 Mar 2008 09:03:03 +0100 Subject: [PATCH 132/250] [ALSA] sound/drivers/pcsp/pcsp.c build fix sound/drivers/pcsp/pcsp.c: In function 'snd_pcsp_create': sound/drivers/pcsp/pcsp.c:54: error: 'loops_per_jiffy' undeclared (first use in\ this function) sound/drivers/pcsp/pcsp.c:54: error: (Each undeclared identifier is reported on\ ly once sound/drivers/pcsp/pcsp.c:54: error: for each function it appears in.) Signed-off-by: Mariusz Kozlowski Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index ac57e87d01bc..d8f96219fd36 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include "pcsp_input.h" #include "pcsp.h" From f32a19e3e7e72cc896d02c3d104f58dc972d43ea Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Tue, 18 Mar 2008 09:27:08 +0100 Subject: [PATCH 133/250] [ALSA] hda-codec - Fix DAC assignment order in ALC883 Actually clfe and surround DACs are inverted in alc883_dac_nids array (see ALC883 datasheet). I discovered this while testing multi-channel setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped in 6 channel mode (also in 4 channel mode you didn't get rear left/right output). Other models also were affected by this bug, as can be seen by the mixer layouts that "workaround" this (the real bug was not noticed, and some other models simply played with mixer and initial verbs). Thus along with fixing the order of dac nids, also change the models that relied on previous dac ordering properly. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 56 ++++++++++++++--------------------- 1 file changed, 22 insertions(+), 34 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf729f518f87..d55987479dd3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6450,7 +6450,7 @@ static int patch_alc882(struct hda_codec *codec) static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03, 0x05 + 0x02, 0x03, 0x04, 0x05 }; static hda_nid_t alc883_adc_nids[2] = { @@ -6958,12 +6958,12 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -6998,12 +6998,12 @@ static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -7036,12 +7036,12 @@ static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -7332,26 +7332,14 @@ static struct hda_verb alc883_haier_w66_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_6st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ - { } -}; - static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; static struct hda_verb alc888_6st_dell_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -7985,7 +7973,7 @@ static struct alc_config_preset alc883_presets[] = { }, [ALC888_6ST_HP] = { .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, + .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, From 86d34b7ec878ea4b4c9f33ce92f1722c4326dbe7 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Tue, 18 Mar 2008 09:27:59 +0100 Subject: [PATCH 134/250] [ALSA] hda-codec - Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267) Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d55987479dd3..c21d7863e0dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7763,6 +7763,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), From 2add9b925394746eff692ff0875d21ea2d5289e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Mar 2008 09:47:06 +0100 Subject: [PATCH 135/250] [ALSA] hda-intel - Add barrier Add proper barriers in the RIRB communication code. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9a20fb1bef4f..557f269f83ab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -536,8 +536,9 @@ static void azx_update_rirb(struct azx *chip) if (res_ex & ICH6_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); else if (chip->rirb.cmds) { - chip->rirb.cmds--; chip->rirb.res = res; + smp_wmb(); + chip->rirb.cmds--; } } } @@ -556,8 +557,10 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) azx_update_rirb(chip); spin_unlock_irq(&chip->reg_lock); } - if (!chip->rirb.cmds) + if (!chip->rirb.cmds) { + smp_rmb(); return chip->rirb.res; /* the last value */ + } if (time_after(jiffies, timeout)) break; if (codec->bus->needs_damn_long_delay) From 117f257d7a9599ff9cb5ab7a6a10201c6294b5f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Mar 2008 09:53:23 +0100 Subject: [PATCH 136/250] [ALSA] hda-codec - Fix spekaer output of Panasonic CF-74 Add a new model "panasonic" for Panasonic CF-74 with STAC9200 codec to fix the speaker output. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_sigmatel.c | 10 +++++++++- 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 80a0629f4917..b02404395e52 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -992,6 +992,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dell-m26 Dell Inspiron 1501 dell-m27 Dell Inspiron E1705/9400 gateway Gateway laptops with EAPD control + panasonic Panasonic CF-74 STAC9205/9254 ref Reference board diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b8e69a1b93f0..a39fbd89a985 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -52,6 +52,7 @@ enum { STAC_9200_DELL_M26, STAC_9200_DELL_M27, STAC_9200_GATEWAY, + STAC_9200_PANASONIC, STAC_9200_MODELS }; @@ -1121,6 +1122,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_9200_DELL_M25] = dell9200_m25_pin_configs, [STAC_9200_DELL_M26] = dell9200_m26_pin_configs, [STAC_9200_DELL_M27] = dell9200_m27_pin_configs, + [STAC_9200_PANASONIC] = ref9200_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { @@ -1137,6 +1139,7 @@ static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_9200_DELL_M26] = "dell-m26", [STAC_9200_DELL_M27] = "dell-m27", [STAC_9200_GATEWAY] = "gateway", + [STAC_9200_PANASONIC] = "panasonic", }; static struct snd_pci_quirk stac9200_cfg_tbl[] = { @@ -1203,7 +1206,7 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6, "unknown Dell", STAC_9200_DELL_M26), /* Panasonic */ - SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF), + SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC), /* Gateway machines needs EAPD to be set on resume */ SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY), SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*", @@ -3302,6 +3305,11 @@ static int patch_stac9200(struct hda_codec *codec) spec->init = stac9200_core_init; spec->mixer = stac9200_mixer; + if (spec->board_config == STAC_9200_PANASONIC) { + spec->gpio_mask = spec->gpio_dir = 0x09; + spec->gpio_data = 0x00; + } + err = stac9200_parse_auto_config(codec); if (err < 0) { stac92xx_free(codec); From 888afa15418f001896bc11f498f9348e029611bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Mar 2008 09:57:50 +0100 Subject: [PATCH 137/250] [ALSA] hda-codec - keep the format verb at closing PCM streams Keep the format verb at closing PCM streams. Introduced snd_hda_codec_cleanup_stream() for the parcicular purpose. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 31 +++++++++++++++++++++---------- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/patch_analog.c | 6 ++---- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 7 +++---- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 3 +-- 9 files changed, 34 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e6bace83e7cf..689d177c17bd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -720,6 +720,19 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); } +void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) +{ + if (!nid) + return; + + snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); +#if 0 /* keep the format */ + msleep(1); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); +#endif +} + /* * amp access functions */ @@ -2204,7 +2217,7 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -2589,7 +2602,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, mutex_lock(&codec->spdif_mutex); if (mout->dig_out_used == HDA_DIG_ANALOG_DUP) /* already opened as analog dup; reset it once */ - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); mout->dig_out_used = HDA_DIG_EXCLUSIVE; mutex_unlock(&codec->spdif_mutex); return 0; @@ -2684,8 +2697,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, stream_tag, format); } else { mout->dig_out_used = 0; - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); } } mutex_unlock(&codec->spdif_mutex); @@ -2727,17 +2739,16 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, int i; for (i = 0; i < mout->num_dacs; i++) - snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, nids[i]); if (mout->hp_nid) - snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->hp_nid); for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) if (mout->extra_out_nid[i]) - snd_hda_codec_setup_stream(codec, - mout->extra_out_nid[i], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, + mout->extra_out_nid[i]); mutex_lock(&codec->spdif_mutex); if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) { - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); mout->dig_out_used = 0; } mutex_unlock(&codec->spdif_mutex); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 301b5227bfb1..dcd390b2bbaa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -722,6 +722,7 @@ int snd_hda_build_pcms(struct hda_bus *bus); void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format); +void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid); unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f9de7c467c25..59e4389c94a4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1007,8 +1007,8 @@ static int generic_pcm2_cleanup(struct hda_pcm_stream *hinfo, { struct hda_gspec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); - snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, spec->dac_node[1]->nid); return 0; } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 87db3c410a10..220784bb76a7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -296,8 +296,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -3325,8 +3324,7 @@ static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - snd_hda_codec_setup_stream(codec, 0x05 + substream->number, - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number); return 0; } diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 1892c81f1d11..8d142851aaca 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -513,7 +513,7 @@ static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct cmi_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c67613ff842d..36fd85260035 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -174,8 +174,7 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -243,7 +242,7 @@ static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; return 0; } @@ -1594,7 +1593,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) new_adc = spec->adc_nids[spec->cur_adc_idx]; if (spec->cur_adc && spec->cur_adc != new_adc) { /* stream is running, let's swap the current ADC */ - snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = new_adc; snd_hda_codec_setup_stream(codec, new_adc, spec->cur_adc_stream_tag, 0, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c21d7863e0dc..eb40f4820c8b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2409,8 +2409,8 @@ static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct alc_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, + spec->adc_nids[substream->number + 1]); return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a39fbd89a985..8eff8fe9dcfd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1932,7 +1932,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct sigmatel_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 09f1c25eb7e5..52b1d81a26f7 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -431,8 +431,7 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } From c0bbf48db35fec29c39c8a7826ca271069537e57 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Tue, 18 Mar 2008 12:08:35 +0100 Subject: [PATCH 138/250] [ALSA] soc - Add missing audio path between Mono Mixer and Mic PGAs Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9713.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index c3d0afdc0997..e3174c4d980d 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -499,6 +499,8 @@ static const char *audio_map[][3] = { {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"}, + {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"}, {"Mono Mixer", NULL, "Capture Mono Mux"}, /* DAC inv mux 1 */ From f081374b607f2656ca79a94d96d99cd5a2f60b68 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Mar 2008 12:13:03 +0100 Subject: [PATCH 139/250] [ALSA] hda-codec - Support of Lenovo Thinkpad X300 Added the model thinkpad for Lenovo Thinkpad X300 with AD1984A codec. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_analog.c | 94 +++++++++++++++++++ 2 files changed, 95 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index b02404395e52..42dd8f5855da 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -923,6 +923,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. desktop 3-stack desktop (default) laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing + thinkpad Lenovo Thinkpad X300 AD1884 N/A diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 220784bb76a7..2befeebd909e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3701,6 +3701,88 @@ static struct hda_verb ad1884a_laptop_verbs[] = { { } /* end */ }; +/* + * Thinkpad X300 + * 0x11 - HP + * 0x12 - speaker + * 0x14 - mic-in + * 0x17 - built-in mic + */ + +static struct hda_verb ad1984a_thinkpad_verbs[] = { + /* HP unmute */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* turn on EAPD */ + {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1984a_thinkpad_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x5 }, + { "Mix", 0x3 }, + }, +}; + +/* mute internal speaker if HP is plugged */ +static void ad1984a_thinkpad_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +/* unsolicited event for HP jack sensing */ +static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1984a_thinkpad_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1984a_thinkpad_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1984a_thinkpad_automute(codec); + return 0; +} + /* */ @@ -3708,6 +3790,7 @@ enum { AD1884A_DESKTOP, AD1884A_LAPTOP, AD1884A_MOBILE, + AD1884A_THINKPAD, AD1884A_MODELS }; @@ -3715,10 +3798,12 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_DESKTOP] = "desktop", [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", + [AD1884A_THINKPAD] = "thinkpad", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; @@ -3773,6 +3858,15 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; break; + case AD1884A_THINKPAD: + spec->mixers[0] = ad1984a_thinkpad_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1984a_thinkpad_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984a_thinkpad_capture_source; + codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; + codec->patch_ops.init = ad1984a_thinkpad_init; + break; } return 0; From 850f0e5212a73a548b9c29faf452b4a14d80f43b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Mar 2008 17:11:05 +0100 Subject: [PATCH 140/250] [ALSA] hda-intel - Add sync support Addded the support of sync streams to hda-intel driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 96 ++++++++++++++++++++++++++++++++------- 1 file changed, 79 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 557f269f83ab..bc3867e1945c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -221,6 +221,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* SD_CTL bits */ #define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ #define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ +#define SD_CTL_STRIPE (3 << 16) /* stripe control */ +#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ +#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ #define SD_CTL_STREAM_TAG_MASK (0xf << 20) #define SD_CTL_STREAM_TAG_SHIFT 20 @@ -1180,7 +1183,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { SNDRV_PCM_INFO_MMAP_VALID | /* No full-resume yet implemented */ /* SNDRV_PCM_INFO_RESUME |*/ - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -1242,6 +1246,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) spin_unlock_irqrestore(&chip->reg_lock, flags); runtime->private_data = azx_dev; + snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); return 0; } @@ -1326,37 +1331,94 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - int err = 0; + struct azx_dev *azx_dev; + struct snd_pcm_substream *s; + int start, nsync = 0, sbits = 0; + int nwait, timeout; - spin_lock(&chip->reg_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: - azx_stream_start(chip, azx_dev); - azx_dev->running = 1; + start = 1; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: - azx_stream_stop(chip, azx_dev); - azx_dev->running = 0; + start = 0; break; default: - err = -EINVAL; + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + sbits |= 1 << azx_dev->index; + nsync++; + snd_pcm_trigger_done(s, substream); + } + + spin_lock(&chip->reg_lock); + if (nsync > 1) { + /* first, set SYNC bits of corresponding streams */ + azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits); + } + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + if (start) + azx_stream_start(chip, azx_dev); + else + azx_stream_stop(chip, azx_dev); + azx_dev->running = start; } spin_unlock(&chip->reg_lock); - if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH || - cmd == SNDRV_PCM_TRIGGER_SUSPEND || - cmd == SNDRV_PCM_TRIGGER_STOP) { - int timeout = 5000; - while ((azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START) && - --timeout) - ; + if (start) { + if (nsync == 1) + return 0; + /* wait until all FIFOs get ready */ + for (timeout = 5000; timeout; timeout--) { + nwait = 0; + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + if (!(azx_sd_readb(azx_dev, SD_STS) & + SD_STS_FIFO_READY)) + nwait++; + } + if (!nwait) + break; + cpu_relax(); + } + } else { + /* wait until all RUN bits are cleared */ + for (timeout = 5000; timeout; timeout--) { + nwait = 0; + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + if (azx_sd_readb(azx_dev, SD_CTL) & + SD_CTL_DMA_START) + nwait++; + } + if (!nwait) + break; + cpu_relax(); + } } - return err; + if (nsync > 1) { + spin_lock(&chip->reg_lock); + /* reset SYNC bits */ + azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits); + spin_unlock(&chip->reg_lock); + } + return 0; } static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) From acf5850ea73bf82081fb65cf10dd36a9d7a890e9 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 18 Mar 2008 17:18:18 +0100 Subject: [PATCH 141/250] [ALSA] Removed deprecated sound/driver.h from Freescale MPC8610 drivers With commit 9004acc70e8c49c50c4c7b652f906f1e0ed5709d, include/sound/driver.h is deprecated. This patch removes the #include from fsl_ssi.c and fsl_dma.c. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai --- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 652514fc8142..78de7168d2ba 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -20,7 +20,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 145ad13d52d1..b2a11b0d2e4c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -15,7 +15,6 @@ #include #include -#include #include #include #include From f5b2368ba8c203eb5bb7e5bbd99f4d9064a6aac0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:14:01 +0100 Subject: [PATCH 142/250] [ALSA] oxygen: move WM8785 symbols to a header file Move the WM8786 register symbol definitions to their own header file. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 42 +----------------------------------- sound/pci/oxygen/wm8785.h | 45 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 41 deletions(-) create mode 100644 sound/pci/oxygen/wm8785.h diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 9a9941bb0460..542752442a95 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -40,6 +40,7 @@ #include "oxygen.h" #include "ak4396.h" #include "cm9780.h" +#include "wm8785.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 driver"); @@ -80,47 +81,6 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_LINE_MUTE CM9780_GPO0 -#define WM8785_R0 0 -#define WM8785_R1 1 -#define WM8785_R2 2 -#define WM8785_R7 7 - -/* R0 */ -#define WM8785_MCR_MASK 0x007 -#define WM8785_MCR_SLAVE 0x000 -#define WM8785_MCR_MASTER_128 0x001 -#define WM8785_MCR_MASTER_192 0x002 -#define WM8785_MCR_MASTER_256 0x003 -#define WM8785_MCR_MASTER_384 0x004 -#define WM8785_MCR_MASTER_512 0x005 -#define WM8785_MCR_MASTER_768 0x006 -#define WM8785_OSR_MASK 0x018 -#define WM8785_OSR_SINGLE 0x000 -#define WM8785_OSR_DOUBLE 0x008 -#define WM8785_OSR_QUAD 0x010 -#define WM8785_FORMAT_MASK 0x060 -#define WM8785_FORMAT_RJUST 0x000 -#define WM8785_FORMAT_LJUST 0x020 -#define WM8785_FORMAT_I2S 0x040 -#define WM8785_FORMAT_DSP 0x060 -/* R1 */ -#define WM8785_WL_MASK 0x003 -#define WM8785_WL_16 0x000 -#define WM8785_WL_20 0x001 -#define WM8785_WL_24 0x002 -#define WM8785_WL_32 0x003 -#define WM8785_LRP 0x004 -#define WM8785_BCLKINV 0x008 -#define WM8785_LRSWAP 0x010 -#define WM8785_DEVNO_MASK 0x0e0 -/* R2 */ -#define WM8785_HPFR 0x001 -#define WM8785_HPFL 0x002 -#define WM8785_SDODIS 0x004 -#define WM8785_PWRDNR 0x008 -#define WM8785_PWRDNL 0x010 -#define WM8785_TDM_MASK 0x1c0 - struct generic_data { u8 ak4396_ctl2; }; diff --git a/sound/pci/oxygen/wm8785.h b/sound/pci/oxygen/wm8785.h new file mode 100644 index 000000000000..8c23e315ae66 --- /dev/null +++ b/sound/pci/oxygen/wm8785.h @@ -0,0 +1,45 @@ +#ifndef WM8785_H_INCLUDED +#define WM8785_H_INCLUDED + +#define WM8785_R0 0 +#define WM8785_R1 1 +#define WM8785_R2 2 +#define WM8785_R7 7 + +/* R0 */ +#define WM8785_MCR_MASK 0x007 +#define WM8785_MCR_SLAVE 0x000 +#define WM8785_MCR_MASTER_128 0x001 +#define WM8785_MCR_MASTER_192 0x002 +#define WM8785_MCR_MASTER_256 0x003 +#define WM8785_MCR_MASTER_384 0x004 +#define WM8785_MCR_MASTER_512 0x005 +#define WM8785_MCR_MASTER_768 0x006 +#define WM8785_OSR_MASK 0x018 +#define WM8785_OSR_SINGLE 0x000 +#define WM8785_OSR_DOUBLE 0x008 +#define WM8785_OSR_QUAD 0x010 +#define WM8785_FORMAT_MASK 0x060 +#define WM8785_FORMAT_RJUST 0x000 +#define WM8785_FORMAT_LJUST 0x020 +#define WM8785_FORMAT_I2S 0x040 +#define WM8785_FORMAT_DSP 0x060 +/* R1 */ +#define WM8785_WL_MASK 0x003 +#define WM8785_WL_16 0x000 +#define WM8785_WL_20 0x001 +#define WM8785_WL_24 0x002 +#define WM8785_WL_32 0x003 +#define WM8785_LRP 0x004 +#define WM8785_BCLKINV 0x008 +#define WM8785_LRSWAP 0x010 +#define WM8785_DEVNO_MASK 0x0e0 +/* R2 */ +#define WM8785_HPFR 0x001 +#define WM8785_HPFL 0x002 +#define WM8785_SDODIS 0x004 +#define WM8785_PWRDNR 0x008 +#define WM8785_PWRDNL 0x010 +#define WM8785_TDM_MASK 0x1c0 + +#endif From 33fa724e291d3cc6c319f7db487e6e084ef5d4b5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:16:40 +0100 Subject: [PATCH 143/250] [ALSA] virtuoso: move PCM1796 symbols to a header file Move the PCM1796 register symbol definitions to their own header file. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/pcm1796.h | 58 +++++++++++++++++++++++++++++++++++++ sound/pci/oxygen/virtuoso.c | 55 +---------------------------------- 2 files changed, 59 insertions(+), 54 deletions(-) create mode 100644 sound/pci/oxygen/pcm1796.h diff --git a/sound/pci/oxygen/pcm1796.h b/sound/pci/oxygen/pcm1796.h new file mode 100644 index 000000000000..698bf46c710c --- /dev/null +++ b/sound/pci/oxygen/pcm1796.h @@ -0,0 +1,58 @@ +#ifndef PCM1796_H_INCLUDED +#define PCM1796_H_INCLUDED + +/* register 16 */ +#define PCM1796_ATL_MASK 0xff +/* register 17 */ +#define PCM1796_ATR_MASK 0xff +/* register 18 */ +#define PCM1796_MUTE 0x01 +#define PCM1796_DME 0x02 +#define PCM1796_DMF_MASK 0x0c +#define PCM1796_DMF_DISABLED 0x00 +#define PCM1796_DMF_48 0x04 +#define PCM1796_DMF_441 0x08 +#define PCM1796_DMF_32 0x0c +#define PCM1796_FMT_MASK 0x70 +#define PCM1796_FMT_16_RJUST 0x00 +#define PCM1796_FMT_20_RJUST 0x10 +#define PCM1796_FMT_24_RJUST 0x20 +#define PCM1796_FMT_24_LJUST 0x30 +#define PCM1796_FMT_16_I2S 0x40 +#define PCM1796_FMT_24_I2S 0x50 +#define PCM1796_ATLD 0x80 +/* register 19 */ +#define PCM1796_INZD 0x01 +#define PCM1796_FLT_MASK 0x02 +#define PCM1796_FLT_SHARP 0x00 +#define PCM1796_FLT_SLOW 0x02 +#define PCM1796_DFMS 0x04 +#define PCM1796_OPE 0x10 +#define PCM1796_ATS_MASK 0x60 +#define PCM1796_ATS_1 0x00 +#define PCM1796_ATS_2 0x20 +#define PCM1796_ATS_4 0x40 +#define PCM1796_ATS_8 0x60 +#define PCM1796_REV 0x80 +/* register 20 */ +#define PCM1796_OS_MASK 0x03 +#define PCM1796_OS_64 0x00 +#define PCM1796_OS_32 0x01 +#define PCM1796_OS_128 0x02 +#define PCM1796_CHSL_MASK 0x04 +#define PCM1796_CHSL_LEFT 0x00 +#define PCM1796_CHSL_RIGHT 0x04 +#define PCM1796_MONO 0x08 +#define PCM1796_DFTH 0x10 +#define PCM1796_DSD 0x20 +#define PCM1796_SRST 0x40 +/* register 21 */ +#define PCM1796_PCMZ 0x01 +#define PCM1796_DZ_MASK 0x06 +/* register 22 */ +#define PCM1796_ZFGL 0x01 +#define PCM1796_ZFGR 0x02 +/* register 23 */ +#define PCM1796_ID_MASK 0x1f + +#endif diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index e4e23789080a..127dd664fc16 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -47,6 +47,7 @@ #include #include "oxygen.h" #include "cm9780.h" +#include "pcm1796.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("Asus AV200 driver"); @@ -82,60 +83,6 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_LINE_MUTE CM9780_GPO0 -/* register 16 */ -#define PCM1796_ATL_MASK 0xff -/* register 17 */ -#define PCM1796_ATR_MASK 0xff -/* register 18 */ -#define PCM1796_MUTE 0x01 -#define PCM1796_DME 0x02 -#define PCM1796_DMF_MASK 0x0c -#define PCM1796_DMF_DISABLED 0x00 -#define PCM1796_DMF_48 0x04 -#define PCM1796_DMF_441 0x08 -#define PCM1796_DMF_32 0x0c -#define PCM1796_FMT_MASK 0x70 -#define PCM1796_FMT_16_RJUST 0x00 -#define PCM1796_FMT_20_RJUST 0x10 -#define PCM1796_FMT_24_RJUST 0x20 -#define PCM1796_FMT_24_LJUST 0x30 -#define PCM1796_FMT_16_I2S 0x40 -#define PCM1796_FMT_24_I2S 0x50 -#define PCM1796_ATLD 0x80 -/* register 19 */ -#define PCM1796_INZD 0x01 -#define PCM1796_FLT_MASK 0x02 -#define PCM1796_FLT_SHARP 0x00 -#define PCM1796_FLT_SLOW 0x02 -#define PCM1796_DFMS 0x04 -#define PCM1796_OPE 0x10 -#define PCM1796_ATS_MASK 0x60 -#define PCM1796_ATS_1 0x00 -#define PCM1796_ATS_2 0x20 -#define PCM1796_ATS_4 0x40 -#define PCM1796_ATS_8 0x60 -#define PCM1796_REV 0x80 -/* register 20 */ -#define PCM1796_OS_MASK 0x03 -#define PCM1796_OS_64 0x00 -#define PCM1796_OS_32 0x01 -#define PCM1796_OS_128 0x02 -#define PCM1796_CHSL_MASK 0x04 -#define PCM1796_CHSL_LEFT 0x00 -#define PCM1796_CHSL_RIGHT 0x04 -#define PCM1796_MONO 0x08 -#define PCM1796_DFTH 0x10 -#define PCM1796_DSD 0x20 -#define PCM1796_SRST 0x40 -/* register 21 */ -#define PCM1796_PCMZ 0x01 -#define PCM1796_DZ_MASK 0x06 -/* register 22 */ -#define PCM1796_ZFGL 0x01 -#define PCM1796_ZFGR 0x02 -/* register 23 */ -#define PCM1796_ID_MASK 0x1f - struct xonar_data { u8 is_d2x; u8 has_power; From fa5d8106cb52e5df28673f59cc25af520dc83382 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:17:33 +0100 Subject: [PATCH 144/250] [ALSA] oxygen: add monitor controls Add controls to enable monitoring of the analog and digital inputs. To allow monitoring after loading the driver when nothing has been played back or recorded yet, the I2S input and outputs are initialized to a valid configuration. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 16 ++-- sound/pci/oxygen/oxygen_mixer.c | 156 ++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+), 8 deletions(-) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 78c21155218e..a1abb50eedb5 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -267,20 +267,20 @@ static void oxygen_init(struct oxygen *chip) (OXYGEN_FORMAT_16 << OXYGEN_MULTICH_FORMAT_SHIFT)); oxygen_write8(chip, OXYGEN_REC_CHANNELS, OXYGEN_REC_CHANNELS_2_2_2); oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, - OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | - OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_RATE_48000 | chip->model->dac_i2s_format | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | - OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_RATE_48000 | chip->model->adc_i2s_format | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, - OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | - OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_RATE_48000 | chip->model->adc_i2s_format | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, - OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | - OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_RATE_48000 | chip->model->adc_i2s_format | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_SENSE_MASK | diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index a8e4623415d9..4e77b79b3b6e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -446,6 +446,50 @@ static int spdif_loopback_put(struct snd_kcontrol *ctl, return changed; } +static int monitor_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 1; + return 0; +} + +static int monitor_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u8 bit = ctl->private_value; + int invert = ctl->private_value & (1 << 8); + + value->value.integer.value[0] = + !!invert ^ !!(oxygen_read8(chip, OXYGEN_ADC_MONITOR) & bit); + return 0; +} + +static int monitor_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u8 bit = ctl->private_value; + int invert = ctl->private_value & (1 << 8); + u8 oldreg, newreg; + int changed; + + spin_lock_irq(&chip->reg_lock); + oldreg = oxygen_read8(chip, OXYGEN_ADC_MONITOR); + if ((!!value->value.integer.value[0] ^ !!invert) != 0) + newreg = oldreg | bit; + else + newreg = oldreg & ~bit; + changed = newreg != oldreg; + if (changed) + oxygen_write8(chip, OXYGEN_ADC_MONITOR, newreg); + spin_unlock_irq(&chip->reg_lock); + return changed; +} + static int ac97_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -608,6 +652,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, .private_value = ((codec) << 24) | (index), \ } +static DECLARE_TLV_DB_SCALE(monitor_db_scale, -1000, 1000, 0); static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0); static DECLARE_TLV_DB_SCALE(ac97_rec_db_scale, 0, 150, 0); @@ -692,6 +737,93 @@ static const struct snd_kcontrol_new controls[] = { }, }; +static const struct snd_kcontrol_new monitor_a_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Switch", + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_A, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_A_HALF_VOL | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, +}; +static const struct snd_kcontrol_new monitor_b_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Switch", + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, +}; +static const struct snd_kcontrol_new monitor_2nd_b_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Switch", + .index = 1, + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, +}; +static const struct snd_kcontrol_new monitor_c_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Input Monitor Switch", + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_C, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Input Monitor Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_C_HALF_VOL | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, +}; + static const struct snd_kcontrol_new ac97_controls[] = { AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC), AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1), @@ -778,6 +910,30 @@ int oxygen_mixer_init(struct oxygen *chip) err = add_controls(chip, controls, ARRAY_SIZE(controls)); if (err < 0) return err; + if (chip->model->used_channels & OXYGEN_CHANNEL_A) { + err = add_controls(chip, monitor_a_controls, + ARRAY_SIZE(monitor_a_controls)); + if (err < 0) + return err; + } else if (chip->model->used_channels & OXYGEN_CHANNEL_B) { + err = add_controls(chip, monitor_b_controls, + ARRAY_SIZE(monitor_b_controls)); + if (err < 0) + return err; + } + if ((chip->model->used_channels & (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B)) + == (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B)) { + err = add_controls(chip, monitor_2nd_b_controls, + ARRAY_SIZE(monitor_2nd_b_controls)); + if (err < 0) + return err; + } + if (chip->model->used_channels & OXYGEN_CHANNEL_C) { + err = add_controls(chip, monitor_c_controls, + ARRAY_SIZE(monitor_c_controls)); + if (err < 0) + return err; + } if (chip->has_ac97_0) { err = add_controls(chip, ac97_controls, ARRAY_SIZE(ac97_controls)); From f009ad9b39e6484d8e36e9e5029c07eab8c12e8f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:19:41 +0100 Subject: [PATCH 145/250] [ALSA] oxygen: change model-specific PCM device configuration When specifying which PCM devices to use, model drivers now use flags that also specify the routing between PCM devices and DMA channels instead of just DMA channel bits. This simplifies some code that checks for these flags. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 6 +- sound/pci/oxygen/oxygen.c | 22 ++-- sound/pci/oxygen/oxygen.h | 12 +- sound/pci/oxygen/oxygen_mixer.c | 202 ++++++++++++++++---------------- sound/pci/oxygen/oxygen_pcm.c | 76 ++++++------ sound/pci/oxygen/virtuoso.c | 8 +- 6 files changed, 175 insertions(+), 151 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 666f69a3312e..fa489ed3ed46 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -160,10 +160,10 @@ static const struct oxygen_model model_hifier = { .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, .model_data_size = sizeof(struct hifier_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, .dac_channels = 2, - .used_channels = OXYGEN_CHANNEL_A | - OXYGEN_CHANNEL_SPDIF | - OXYGEN_CHANNEL_MULTICH, .function_flags = 0, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 542752442a95..9faf43c949b2 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -303,12 +303,13 @@ static const struct oxygen_model model_generic = { .update_dac_mute = update_ak4396_mute, .ac97_switch_hook = cmi9780_switch_hook, .model_data_size = sizeof(struct generic_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + PLAYBACK_2_TO_AC97_1 | + CAPTURE_0_FROM_I2S_1 | + CAPTURE_1_FROM_SPDIF | + CAPTURE_2_FROM_AC97_1, .dac_channels = 8, - .used_channels = OXYGEN_CHANNEL_A | - OXYGEN_CHANNEL_C | - OXYGEN_CHANNEL_SPDIF | - OXYGEN_CHANNEL_MULTICH | - OXYGEN_CHANNEL_AC97, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -327,12 +328,13 @@ static const struct oxygen_model model_meridian = { .update_dac_mute = update_ak4396_mute, .ac97_switch_hook = cmi9780_switch_hook, .model_data_size = sizeof(struct generic_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + PLAYBACK_2_TO_AC97_1 | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF | + CAPTURE_2_FROM_AC97_1, .dac_channels = 8, - .used_channels = OXYGEN_CHANNEL_B | - OXYGEN_CHANNEL_C | - OXYGEN_CHANNEL_SPDIF | - OXYGEN_CHANNEL_MULTICH | - OXYGEN_CHANNEL_AC97, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index ad50fb8b206b..fde995cf2edf 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -16,6 +16,16 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +/* model-specific configuration of outputs/inputs */ +#define PLAYBACK_0_TO_I2S 0x001 +#define PLAYBACK_1_TO_SPDIF 0x004 +#define PLAYBACK_2_TO_AC97_1 0x008 +#define CAPTURE_0_FROM_I2S_1 0x010 +#define CAPTURE_0_FROM_I2S_2 0x020 +#define CAPTURE_1_FROM_SPDIF 0x080 +#define CAPTURE_2_FROM_I2S_2 0x100 +#define CAPTURE_2_FROM_AC97_1 0x200 + enum { CONTROL_SPDIF_PCM, CONTROL_SPDIF_INPUT_BITS, @@ -91,8 +101,8 @@ struct oxygen_model { unsigned int reg, int mute); void (*gpio_changed)(struct oxygen *chip); size_t model_data_size; + unsigned int pcm_dev_cfg; u8 dac_channels; - u8 used_channels; u8 function_flags; u16 dac_i2s_format; u16 adc_i2s_format; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4e77b79b3b6e..6b5ff6e0fadd 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -737,90 +737,111 @@ static const struct snd_kcontrol_new controls[] = { }, }; -static const struct snd_kcontrol_new monitor_a_controls[] = { +static const struct { + unsigned int pcm_dev; + struct snd_kcontrol_new controls[2]; +} monitor_controls[] = { { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", - .info = snd_ctl_boolean_mono_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_A, + .pcm_dev = CAPTURE_0_FROM_I2S_1, + .controls = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Switch", + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_A, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_A_HALF_VOL + | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, + }, }, { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = monitor_volume_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_A_HALF_VOL | (1 << 8), - .tlv = { .p = monitor_db_scale, }, - }, -}; -static const struct snd_kcontrol_new monitor_b_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", - .info = snd_ctl_boolean_mono_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_B, + .pcm_dev = CAPTURE_0_FROM_I2S_2, + .controls = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Switch", + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL + | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, + }, }, { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = monitor_volume_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL | (1 << 8), - .tlv = { .p = monitor_db_scale, }, - }, -}; -static const struct snd_kcontrol_new monitor_2nd_b_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", - .index = 1, - .info = snd_ctl_boolean_mono_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_B, + .pcm_dev = CAPTURE_2_FROM_I2S_2, + .controls = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Switch", + .index = 1, + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Input Monitor Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL + | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, + }, }, { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", - .index = 1, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = monitor_volume_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL | (1 << 8), - .tlv = { .p = monitor_db_scale, }, - }, -}; -static const struct snd_kcontrol_new monitor_c_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Monitor Switch", - .info = snd_ctl_boolean_mono_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_C, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Monitor Volume", - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = monitor_volume_info, - .get = monitor_get, - .put = monitor_put, - .private_value = OXYGEN_ADC_MONITOR_C_HALF_VOL | (1 << 8), - .tlv = { .p = monitor_db_scale, }, + .pcm_dev = CAPTURE_1_FROM_SPDIF, + .controls = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Input Monitor Switch", + .info = snd_ctl_boolean_mono_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_C, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Input Monitor Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = monitor_volume_info, + .get = monitor_get, + .put = monitor_put, + .private_value = OXYGEN_ADC_MONITOR_C_HALF_VOL + | (1 << 8), + .tlv = { .p = monitor_db_scale, }, + }, + }, }, }; @@ -905,32 +926,17 @@ static int add_controls(struct oxygen *chip, int oxygen_mixer_init(struct oxygen *chip) { + unsigned int i; int err; err = add_controls(chip, controls, ARRAY_SIZE(controls)); if (err < 0) return err; - if (chip->model->used_channels & OXYGEN_CHANNEL_A) { - err = add_controls(chip, monitor_a_controls, - ARRAY_SIZE(monitor_a_controls)); - if (err < 0) - return err; - } else if (chip->model->used_channels & OXYGEN_CHANNEL_B) { - err = add_controls(chip, monitor_b_controls, - ARRAY_SIZE(monitor_b_controls)); - if (err < 0) - return err; - } - if ((chip->model->used_channels & (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B)) - == (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B)) { - err = add_controls(chip, monitor_2nd_b_controls, - ARRAY_SIZE(monitor_2nd_b_controls)); - if (err < 0) - return err; - } - if (chip->model->used_channels & OXYGEN_CHANNEL_C) { - err = add_controls(chip, monitor_c_controls, - ARRAY_SIZE(monitor_c_controls)); + for (i = 0; i < ARRAY_SIZE(monitor_controls); ++i) { + if (!(chip->model->pcm_dev_cfg & monitor_controls[i].pcm_dev)) + continue; + err = add_controls(chip, monitor_controls[i].controls, + ARRAY_SIZE(monitor_controls[i].controls)); if (err < 0) return err; } diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index b70046aca657..b17c405e069d 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -119,7 +119,7 @@ static int oxygen_open(struct snd_pcm_substream *substream, runtime->private_data = (void *)(uintptr_t)channel; if (channel == PCM_B && chip->has_ac97_1 && - (chip->model->used_channels & OXYGEN_CHANNEL_AC97)) + (chip->model->pcm_dev_cfg & CAPTURE_2_FROM_AC97_1)) runtime->hw = oxygen_ac97_hardware; else runtime->hw = *oxygen_hardware[channel]; @@ -365,7 +365,7 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, return err; is_ac97 = chip->has_ac97_1 && - (chip->model->used_channels & OXYGEN_CHANNEL_AC97); + (chip->model->pcm_dev_cfg & CAPTURE_2_FROM_AC97_1); spin_lock_irq(&chip->reg_lock); oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, @@ -640,34 +640,39 @@ int oxygen_pcm_init(struct oxygen *chip) int outs, ins; int err; - outs = 1; /* OXYGEN_CHANNEL_MULTICH is always used */ - ins = !!(chip->model->used_channels & (OXYGEN_CHANNEL_A | - OXYGEN_CHANNEL_B)); - err = snd_pcm_new(chip->card, "Analog", 0, outs, ins, &pcm); - if (err < 0) - return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_multich_ops); - if (chip->model->used_channels & OXYGEN_CHANNEL_A) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &oxygen_rec_a_ops); - else if (chip->model->used_channels & OXYGEN_CHANNEL_B) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &oxygen_rec_b_ops); - pcm->private_data = chip; - pcm->private_free = oxygen_pcm_free; - strcpy(pcm->name, "Analog"); - snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 512 * 1024, 2048 * 1024); - if (ins) - snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_0_TO_I2S); + ins = !!(chip->model->pcm_dev_cfg & (CAPTURE_0_FROM_I2S_1 | + CAPTURE_0_FROM_I2S_2)); + if (outs | ins) { + err = snd_pcm_new(chip->card, "Analog", 0, outs, ins, &pcm); + if (err < 0) + return err; + if (outs) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &oxygen_multich_ops); + if (chip->model->pcm_dev_cfg & CAPTURE_0_FROM_I2S_1) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &oxygen_rec_a_ops); + else if (chip->model->pcm_dev_cfg & CAPTURE_0_FROM_I2S_2) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &oxygen_rec_b_ops); + pcm->private_data = chip; + pcm->private_free = oxygen_pcm_free; + strcpy(pcm->name, "Analog"); + if (outs) + snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 512 * 1024, 2048 * 1024); + if (ins) + snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 128 * 1024, 256 * 1024); + } - outs = !!(chip->model->used_channels & OXYGEN_CHANNEL_SPDIF); - ins = !!(chip->model->used_channels & OXYGEN_CHANNEL_C); + outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF); + ins = !!(chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF); if (outs | ins) { err = snd_pcm_new(chip->card, "Digital", 1, outs, ins, &pcm); if (err < 0) @@ -686,12 +691,13 @@ int oxygen_pcm_init(struct oxygen *chip) 128 * 1024, 256 * 1024); } - outs = chip->has_ac97_1 && - (chip->model->used_channels & OXYGEN_CHANNEL_AC97); - ins = outs || - (chip->model->used_channels & (OXYGEN_CHANNEL_A | - OXYGEN_CHANNEL_B)) - == (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B); + if (chip->has_ac97_1) { + outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_2_TO_AC97_1); + ins = !!(chip->model->pcm_dev_cfg & CAPTURE_2_FROM_AC97_1); + } else { + outs = 0; + ins = !!(chip->model->pcm_dev_cfg & CAPTURE_2_FROM_I2S_2); + } if (outs | ins) { err = snd_pcm_new(chip->card, outs ? "AC97" : "Analog2", 2, outs, ins, &pcm); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 127dd664fc16..5cd1fac14132 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -348,11 +348,11 @@ static const struct oxygen_model model_xonar = { .ac97_switch_hook = xonar_ac97_switch_hook, .gpio_changed = xonar_gpio_changed, .model_data_size = sizeof(struct xonar_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, .dac_channels = 8, - .used_channels = OXYGEN_CHANNEL_B | - OXYGEN_CHANNEL_C | - OXYGEN_CHANNEL_SPDIF | - OXYGEN_CHANNEL_MULTICH, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, From 87eedd2fd409d5cd515ccd6fc454cef15c5fa38b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:20:13 +0100 Subject: [PATCH 146/250] [ALSA] oxygen: make SPI/2-wire configuration model-specific Allow the model drivers to specify if the codec communication goes over SPI or a 2-wire bus. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 6 ++++-- sound/pci/oxygen/oxygen_lib.c | 10 +++++----- sound/pci/oxygen/virtuoso.c | 3 ++- 4 files changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index fa489ed3ed46..143d83d916dc 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -164,7 +164,7 @@ static const struct oxygen_model model_hifier = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_1, .dac_channels = 2, - .function_flags = 0, + .function_flags = OXYGEN_FUNCTION_SPI, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 9faf43c949b2..e9031ede9621 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -310,7 +310,8 @@ static const struct oxygen_model model_generic = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1, .dac_channels = 8, - .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -335,7 +336,8 @@ static const struct oxygen_model model_meridian = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1, .dac_channels = 8, - .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index a1abb50eedb5..b7c7eb33106c 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -240,12 +240,12 @@ static void oxygen_init(struct oxygen *chip) chip->has_ac97_0 = (i & OXYGEN_AC97_CODEC_0) != 0; chip->has_ac97_1 = (i & OXYGEN_AC97_CODEC_1) != 0; - oxygen_set_bits8(chip, OXYGEN_FUNCTION, - OXYGEN_FUNCTION_RESET_CODEC | - chip->model->function_flags); oxygen_write8_masked(chip, OXYGEN_FUNCTION, - OXYGEN_FUNCTION_SPI, - OXYGEN_FUNCTION_2WIRE_SPI_MASK); + OXYGEN_FUNCTION_RESET_CODEC | + chip->model->function_flags, + OXYGEN_FUNCTION_RESET_CODEC | + OXYGEN_FUNCTION_2WIRE_SPI_MASK | + OXYGEN_FUNCTION_ENABLE_SPI_4_5); oxygen_write8(chip, OXYGEN_DMA_STATUS, 0); oxygen_write8(chip, OXYGEN_DMA_PAUSE, 0); oxygen_write8(chip, OXYGEN_PLAY_CHANNELS, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 5cd1fac14132..5bf3661ab1fe 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -353,7 +353,8 @@ static const struct oxygen_model model_xonar = { CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF, .dac_channels = 8, - .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; From db12b8e301455cf18644aa3b765ae10869eb947c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:20:59 +0100 Subject: [PATCH 147/250] [ALSA] oxygen: move MIDI flag to model struct Put the flag that enables the MIDI port into the model structure instead of passing it as a separate parameter to oxygen_pci_probe(). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 3 ++- sound/pci/oxygen/oxygen.h | 3 ++- sound/pci/oxygen/oxygen_lib.c | 12 ++++++------ sound/pci/oxygen/virtuoso.c | 3 ++- 5 files changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 143d83d916dc..1e54a3cd721e 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -181,7 +181,7 @@ static int __devinit hifier_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], 0, &model_hifier); + err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index e9031ede9621..511ef34a43ca 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -336,6 +336,7 @@ static const struct oxygen_model model_meridian = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1, .dac_channels = 8, + .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -356,7 +357,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, return -ENOENT; } is_meridian = pci_id->driver_data; - err = oxygen_pci_probe(pci, index[dev], id[dev], is_meridian, + err = oxygen_pci_probe(pci, index[dev], id[dev], is_meridian ? &model_meridian : &model_generic); if (err >= 0) ++dev; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index fde995cf2edf..5103482f65e4 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -103,6 +103,7 @@ struct oxygen_model { size_t model_data_size; unsigned int pcm_dev_cfg; u8 dac_channels; + u8 misc_flags; u8 function_flags; u16 dac_i2s_format; u16 adc_i2s_format; @@ -110,7 +111,7 @@ struct oxygen_model { /* oxygen_lib.c */ -int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, int midi, +int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, const struct oxygen_model *model); void oxygen_pci_remove(struct pci_dev *pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b7c7eb33106c..87df2b81c573 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -253,11 +253,13 @@ static void oxygen_init(struct oxygen *chip) OXYGEN_DMA_A_BURST_8 | OXYGEN_DMA_MULTICH_BURST_8); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); - oxygen_write8_masked(chip, OXYGEN_MISC, 0, + oxygen_write8_masked(chip, OXYGEN_MISC, + chip->model->misc_flags, OXYGEN_MISC_WRITE_PCI_SUBID | OXYGEN_MISC_REC_C_FROM_SPDIF | OXYGEN_MISC_REC_B_FROM_AC97 | - OXYGEN_MISC_REC_A_FROM_MULTICH); + OXYGEN_MISC_REC_A_FROM_MULTICH | + OXYGEN_MISC_MIDI); oxygen_write8(chip, OXYGEN_REC_FORMAT, (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_A_SHIFT) | (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_B_SHIFT) | @@ -400,7 +402,7 @@ static void oxygen_card_free(struct snd_card *card) } int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, - int midi, const struct oxygen_model *model) + const struct oxygen_model *model) { struct snd_card *card; struct oxygen *chip; @@ -472,9 +474,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, if (err < 0) goto err_card; - oxygen_write8_masked(chip, OXYGEN_MISC, - midi ? OXYGEN_MISC_MIDI : 0, OXYGEN_MISC_MIDI); - if (midi) { + if (model->misc_flags & OXYGEN_MISC_MIDI) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, chip->addr + OXYGEN_MPU401, MPU401_INFO_INTEGRATED, 0, 0, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 5bf3661ab1fe..fa79db696e2d 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -353,6 +353,7 @@ static const struct oxygen_model model_xonar = { CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF, .dac_channels = 8, + .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -371,7 +372,7 @@ static int __devinit xonar_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], 1, &model_xonar); + err = oxygen_pci_probe(pci, index[dev], id[dev], &model_xonar); if (err >= 0) ++dev; return err; From 43dd89c7e7cde6b42edac88ca852ec61af610863 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Mar 2008 08:21:32 +0100 Subject: [PATCH 148/250] [ALSA] oxygen: disable clock of unused I2S inputs Disable the master clock outputs of any unused I2S inputs. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 29 ++++++++++++++++++----------- 1 file changed, 18 insertions(+), 11 deletions(-) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 87df2b81c573..858e6d4c9194 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -272,18 +272,25 @@ static void oxygen_init(struct oxygen *chip) OXYGEN_RATE_48000 | chip->model->dac_i2s_format | OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); - oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_RATE_48000 | chip->model->adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); - oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, - OXYGEN_RATE_48000 | chip->model->adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + if (chip->model->pcm_dev_cfg & CAPTURE_0_FROM_I2S_1) + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, + OXYGEN_RATE_48000 | chip->model->adc_i2s_format | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + else + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, + OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + if (chip->model->pcm_dev_cfg & (CAPTURE_0_FROM_I2S_2 | + CAPTURE_2_FROM_I2S_2)) + oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, + OXYGEN_RATE_48000 | chip->model->adc_i2s_format | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + else + oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, + OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, - OXYGEN_RATE_48000 | chip->model->adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_SENSE_MASK | OXYGEN_SPDIF_LOCK_MASK | From 91e24faa556548e0705e8940410b8dc3bd1d949d Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Thu, 20 Mar 2008 12:04:46 +0100 Subject: [PATCH 149/250] [ALSA] sound/pci/aw2/aw2-alsa.c needs dma-mapping.h sparc32: sound/pci/aw2/aw2-alsa.c: In function 'snd_aw2_create': sound/pci/aw2/aw2-alsa.c:282: error: 'DMA_32BIT_MASK' undeclared (first use in this function) sound/pci/aw2/aw2-alsa.c:282: error: (Each undeclared identifier is reported only once sound/pci/aw2/aw2-alsa.c:282: error: for each function it appears in.) Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-alsa.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 24e3e4961d9f..12d01c9c6637 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -22,6 +22,7 @@ *****************************************************************************/ #include #include +#include #include #include #include From ee0abefde5273c816bd3d4158e5cb9c591b82684 Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Thu, 20 Mar 2008 12:05:33 +0100 Subject: [PATCH 150/250] [ALSA] sound/pci/pcxhr/pcxhr_core.c: fix printk warning sound/pci/pcxhr/pcxhr_core.c: In function `pcxhr_set_pipe_state': sound/pci/pcxhr/pcxhr_core.c:899: warning: long int format, different type arg (arg 4) suseconds_t is int on sparc64. Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 846afbd30682..78aa81feaa4a 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -897,7 +897,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m #ifdef CONFIG_SND_DEBUG_DETECT do_gettimeofday(&my_tv2); snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n", - my_tv2.tv_usec - my_tv1.tv_usec, err); + (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); #endif return 0; } From ff73317ea7c648cf5f59b8bda4a810f7b5d0312c Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Thu, 20 Mar 2008 12:07:31 +0100 Subject: [PATCH 151/250] [ALSA] sound/pci/pcxhr/pcxhr.c: fix warnings sparc64: sound/pci/pcxhr/pcxhr.c: In function `pcxhr_update_r_buffer': sound/pci/pcxhr/pcxhr.c:459: warning: cast to pointer from integer of different size sound/pci/pcxhr/pcxhr.c: In function `pcxhr_trigger_tasklet': sound/pci/pcxhr/pcxhr.c:628: warning: long int format, different type arg (arg 4) Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index beed5818338c..7fdcdc8c6b64 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -458,7 +458,7 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", is_capture ? 'c' : 'p', - chip->chip_idx, (void*)subs->runtime->dma_addr, + chip->chip_idx, (void *)(long)subs->runtime->dma_addr, subs->runtime->dma_bytes, subs->number); pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS); @@ -626,7 +626,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) #ifdef CONFIG_SND_DEBUG_DETECT do_gettimeofday(&my_tv2); snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n", - my_tv2.tv_usec - my_tv1.tv_usec, err); + (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); #endif } From d16be8ed69f3e59d36be8c422508c3a10082fdaa Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Thu, 20 Mar 2008 12:10:27 +0100 Subject: [PATCH 152/250] [ALSA] ice1724 - Improved the Juli rate setting * moving most of clock-specific code to card-specific routines * support for ESI Juli * to-be-researched - monitoring of analog/digital inputs Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 + sound/pci/ice1712/ice1712.h | 9 + sound/pci/ice1712/ice1724.c | 313 ++++++++++++------------ sound/pci/ice1712/juli.c | 473 +++++++++++++++++++++++++++++++++--- 4 files changed, 600 insertions(+), 196 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 868183bef243..b5903eed6ef5 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -697,6 +697,7 @@ config SND_ICE1724 depends on SND select SND_MPU401_UART select SND_AC97_CODEC + select SND_VMASTER help Say Y here to include support for soundcards based on ICE/VT1724/1720 (Envy24HT/PT) chips. diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 303cffe08bd8..a3bea2247c7f 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -367,6 +367,15 @@ struct snd_ice1712 { /* other board-specific data */ void *spec; + + /* VT172x specific */ + int pro_rate_default; + int (*is_spdif_master)(struct snd_ice1712 *ice); + unsigned int (*get_rate)(struct snd_ice1712 *ice); + void (*set_rate)(struct snd_ice1712 *ice, unsigned int rate); + unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate); + void (*set_spdif_clock)(struct snd_ice1712 *ice); + }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3bfd70577d7a..ceac87056263 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -106,15 +106,19 @@ static unsigned int PRO_RATE_DEFAULT = 44100; * Basic I/O */ +/* + * default rates, default clock routines + */ + /* check whether the clock mode is spdif-in */ -static inline int is_spdif_master(struct snd_ice1712 *ice) +static inline int stdclock_is_spdif_master(struct snd_ice1712 *ice) { return (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER) ? 1 : 0; } static inline int is_pro_rate_locked(struct snd_ice1712 *ice) { - return is_spdif_master(ice) || PRO_RATE_LOCKED; + return ice->is_spdif_master(ice) || PRO_RATE_LOCKED; } /* @@ -391,51 +395,61 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd) #define DMA_PAUSES (VT1724_RDMA0_PAUSE|VT1724_PDMA0_PAUSE|VT1724_RDMA1_PAUSE|\ VT1724_PDMA1_PAUSE|VT1724_PDMA2_PAUSE|VT1724_PDMA3_PAUSE|VT1724_PDMA4_PAUSE) -static int get_max_rate(struct snd_ice1712 *ice) +static const unsigned int stdclock_rate_list[16] = { + 48000, 24000, 12000, 9600, 32000, 16000, 8000, 96000, 44100, + 22050, 11025, 88200, 176400, 0, 192000, 64000 +}; + +static unsigned int stdclock_get_rate(struct snd_ice1712 *ice) { + unsigned int rate; + rate = stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15]; + return rate; +} + +static void stdclock_set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(stdclock_rate_list); i++) { + if (stdclock_rate_list[i] == rate) { + outb(i, ICEMT1724(ice, RATE)); + return; + } + } +} + +static unsigned char stdclock_set_mclk(struct snd_ice1712 *ice, + unsigned int rate) +{ + unsigned char val, old; + /* check MT02 */ if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { - if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720) - return 192000; + val = old = inb(ICEMT1724(ice, I2S_FORMAT)); + if (rate > 96000) + val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */ else - return 96000; - } else - return 48000; + val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */ + if (val != old) { + outb(val, ICEMT1724(ice, I2S_FORMAT)); + /* master clock changed */ + return 1; + } + } + /* no change in master clock */ + return 0; } static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, int force) { unsigned long flags; - unsigned char val, old; - unsigned int i, mclk_change; + unsigned char mclk_change; + unsigned int i, old_rate; - if (rate > get_max_rate(ice)) + if (rate > ice->hw_rates->list[ice->hw_rates->count - 1]) return; - - switch (rate) { - case 8000: val = 6; break; - case 9600: val = 3; break; - case 11025: val = 10; break; - case 12000: val = 2; break; - case 16000: val = 5; break; - case 22050: val = 9; break; - case 24000: val = 1; break; - case 32000: val = 4; break; - case 44100: val = 8; break; - case 48000: val = 0; break; - case 64000: val = 15; break; - case 88200: val = 11; break; - case 96000: val = 7; break; - case 176400: val = 12; break; - case 192000: val = 14; break; - default: - snd_BUG(); - val = 0; - break; - } - spin_lock_irqsave(&ice->reg_lock, flags); - if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) || + if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) || (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); @@ -446,9 +460,9 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, return; } - old = inb(ICEMT1724(ice, RATE)); - if (force || old != val) - outb(val, ICEMT1724(ice, RATE)); + old_rate = ice->get_rate(ice); + if (force || (old_rate != rate)) + ice->set_rate(ice, rate); else if (rate == ice->cur_rate) { spin_unlock_irqrestore(&ice->reg_lock, flags); return; @@ -456,19 +470,9 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, ice->cur_rate = rate; - /* check MT02 */ - mclk_change = 0; - if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { - val = old = inb(ICEMT1724(ice, I2S_FORMAT)); - if (rate > 96000) - val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */ - else - val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */ - if (val != old) { - outb(val, ICEMT1724(ice, I2S_FORMAT)); - mclk_change = 1; - } - } + /* setting master clock */ + mclk_change = ice->set_mclk(ice, rate); + spin_unlock_irqrestore(&ice->reg_lock, flags); if (mclk_change && ice->gpio.i2s_mclk_changed) @@ -727,43 +731,32 @@ static const struct snd_pcm_hardware snd_vt1724_2ch_stereo = /* * set rate constraints */ -static int set_rate_constraints(struct snd_ice1712 *ice, - struct snd_pcm_substream *substream) +static void set_std_hw_rates(struct snd_ice1712 *ice) { - struct snd_pcm_runtime *runtime = substream->runtime; - if (ice->hw_rates) { - /* hardware specific */ - runtime->hw.rate_min = ice->hw_rates->list[0]; - runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1]; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - ice->hw_rates); - } if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { /* I2S */ /* VT1720 doesn't support more than 96kHz */ if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720) - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates_192); - else { - runtime->hw.rates = SNDRV_PCM_RATE_KNOT | - SNDRV_PCM_RATE_8000_96000; - runtime->hw.rate_max = 96000; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates_96); - } - } else if (ice->ac97) { + ice->hw_rates = &hw_constraints_rates_192; + else + ice->hw_rates = &hw_constraints_rates_96; + } else { /* ACLINK */ - runtime->hw.rate_max = 48000; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates_48); + ice->hw_rates = &hw_constraints_rates_48; } - return 0; +} + +static int set_rate_constraints(struct snd_ice1712 *ice, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = ice->hw_rates->list[0]; + runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ice->hw_rates); } /* multi-channel playback needs alignment 8x32bit regardless of the channels @@ -824,7 +817,7 @@ static int snd_vt1724_playback_pro_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->playback_pro_substream = NULL; return 0; @@ -835,7 +828,7 @@ static int snd_vt1724_capture_pro_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->capture_pro_substream = NULL; return 0; } @@ -980,7 +973,7 @@ static int snd_vt1724_playback_spdif_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->playback_con_substream = NULL; if (ice->spdif.ops.close) ice->spdif.ops.close(ice, substream); @@ -1016,7 +1009,7 @@ static int snd_vt1724_capture_spdif_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->capture_con_substream = NULL; if (ice->spdif.ops.close) ice->spdif.ops.close(ice, substream); @@ -1162,7 +1155,7 @@ static int snd_vt1724_playback_indep_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->playback_con_substream_ds[substream->number] = NULL; ice->pcm_reserved[substream->number] = NULL; @@ -1580,50 +1573,18 @@ int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol, static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char * const texts_1724[] = { - "8000", /* 0: 6 */ - "9600", /* 1: 3 */ - "11025", /* 2: 10 */ - "12000", /* 3: 2 */ - "16000", /* 4: 5 */ - "22050", /* 5: 9 */ - "24000", /* 6: 1 */ - "32000", /* 7: 4 */ - "44100", /* 8: 8 */ - "48000", /* 9: 0 */ - "64000", /* 10: 15 */ - "88200", /* 11: 11 */ - "96000", /* 12: 7 */ - "176400", /* 13: 12 */ - "192000", /* 14: 14 */ - "IEC958 Input", /* 15: -- */ - }; - static const char * const texts_1720[] = { - "8000", /* 0: 6 */ - "9600", /* 1: 3 */ - "11025", /* 2: 10 */ - "12000", /* 3: 2 */ - "16000", /* 4: 5 */ - "22050", /* 5: 9 */ - "24000", /* 6: 1 */ - "32000", /* 7: 4 */ - "44100", /* 8: 8 */ - "48000", /* 9: 0 */ - "64000", /* 10: 15 */ - "88200", /* 11: 11 */ - "96000", /* 12: 7 */ - "IEC958 Input", /* 13: -- */ - }; struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = ice->vt1720 ? 14 : 16; + uinfo->value.enumerated.items = ice->hw_rates->count + 1; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - ice->vt1720 ? texts_1720[uinfo->value.enumerated.item] : - texts_1724[uinfo->value.enumerated.item]); + if (uinfo->value.enumerated.item == uinfo->value.enumerated.items - 1) + strcpy(uinfo->value.enumerated.name, "IEC958 Input"); + else + sprintf(uinfo->value.enumerated.name, "%d", + ice->hw_rates->list[uinfo->value.enumerated.item]); return 0; } @@ -1631,68 +1592,79 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - static const unsigned char xlate[16] = { - 9, 6, 3, 1, 7, 4, 0, 12, 8, 5, 2, 11, 13, 255, 14, 10 - }; - unsigned char val; + unsigned int i, rate; spin_lock_irq(&ice->reg_lock); - if (is_spdif_master(ice)) { - ucontrol->value.enumerated.item[0] = ice->vt1720 ? 13 : 15; + if (ice->is_spdif_master(ice)) { + ucontrol->value.enumerated.item[0] = ice->hw_rates->count; } else { - val = xlate[inb(ICEMT1724(ice, RATE)) & 15]; - if (val == 255) { - snd_BUG(); - val = 0; + rate = ice->get_rate(ice); + ucontrol->value.enumerated.item[0] = 0; + for (i = 0; i < ice->hw_rates->count; i++) { + if (ice->hw_rates->list[i] == rate) { + ucontrol->value.enumerated.item[0] = i; + break; + } } - ucontrol->value.enumerated.item[0] = val; } spin_unlock_irq(&ice->reg_lock); return 0; } +/* setting clock to external - SPDIF */ +static void stdclock_set_spdif_clock(struct snd_ice1712 *ice) +{ + unsigned char oval; + unsigned char i2s_oval; + oval = inb(ICEMT1724(ice, RATE)); + outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); + /* setting 256fs */ + i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); + outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); +} + static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - unsigned char oval; - int rate; - int change = 0; - int spdif = ice->vt1720 ? 13 : 15; + unsigned int old_rate, new_rate; + unsigned int item = ucontrol->value.enumerated.item[0]; + unsigned int spdif = ice->hw_rates->count; + + if (item > spdif) + return -EINVAL; spin_lock_irq(&ice->reg_lock); - oval = inb(ICEMT1724(ice, RATE)); - if (ucontrol->value.enumerated.item[0] == spdif) { - unsigned char i2s_oval; - outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); - /* setting 256fs */ - i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); - outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, - ICEMT1724(ice, I2S_FORMAT)); + if (ice->is_spdif_master(ice)) + old_rate = 0; + else + old_rate = ice->get_rate(ice); + if (item == spdif) { + /* switching to external clock via SPDIF */ + ice->set_spdif_clock(ice); + new_rate = 0; } else { - rate = rates[ucontrol->value.integer.value[0] % 15]; - if (rate <= get_max_rate(ice)) { - PRO_RATE_DEFAULT = rate; - spin_unlock_irq(&ice->reg_lock); - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 1); - spin_lock_irq(&ice->reg_lock); - } + /* internal on-card clock */ + new_rate = ice->hw_rates->list[item]; + ice->pro_rate_default = new_rate; + spin_unlock_irq(&ice->reg_lock); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); + spin_lock_irq(&ice->reg_lock); } - change = inb(ICEMT1724(ice, RATE)) != oval; spin_unlock_irq(&ice->reg_lock); - if ((oval & VT1724_SPDIF_MASTER) != - (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER)) { + /* the first reset to the SPDIF master mode? */ + if (old_rate != new_rate && !new_rate) { /* notify akm chips as well */ - if (is_spdif_master(ice)) { - unsigned int i; - for (i = 0; i < ice->akm_codecs; i++) { - if (ice->akm[i].ops.set_rate_val) - ice->akm[i].ops.set_rate_val(&ice->akm[i], 0); - } + unsigned int i; + if (ice->gpio.set_pro_rate) + ice->gpio.set_pro_rate(ice, 0); + for (i = 0; i < ice->akm_codecs; i++) { + if (ice->akm[i].ops.set_rate_val) + ice->akm[i].ops.set_rate_val(&ice->akm[i], 0); } } - return change; + return old_rate != new_rate; } static struct snd_kcontrol_new snd_vt1724_pro_internal_clock __devinitdata = { @@ -2343,6 +2315,19 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, * was called so in ice1712 driver, and vt1724 driver is derived from * ice1712 driver. */ + ice->pro_rate_default = PRO_RATE_DEFAULT; + if (!ice->is_spdif_master) + ice->is_spdif_master = stdclock_is_spdif_master; + if (!ice->get_rate) + ice->get_rate = stdclock_get_rate; + if (!ice->set_rate) + ice->set_rate = stdclock_set_rate; + if (!ice->set_mclk) + ice->set_mclk = stdclock_set_mclk; + if (!ice->set_spdif_clock) + ice->set_spdif_clock = stdclock_set_spdif_clock; + if (!ice->hw_rates) + set_std_hw_rates(ice); if ((err = snd_vt1724_pcm_profi(ice, pcm_dev++)) < 0) { snd_card_free(card); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 4550609b4d47..b4e0c16852a6 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -4,6 +4,8 @@ * Lowlevel functions for ESI Juli@ cards * * Copyright (c) 2004 Jaroslav Kysela + * 2008 Pavel Hofman + * * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -27,11 +29,11 @@ #include #include #include +#include #include "ice1712.h" #include "envy24ht.h" #include "juli.h" - struct juli_spec { struct ak4114 *ak4114; unsigned int analog: 1; @@ -43,6 +45,32 @@ struct juli_spec { #define AK4114_ADDR 0x20 /* S/PDIF receiver */ #define AK4358_ADDR 0x22 /* DAC */ +/* + * Juli does not use the standard ICE1724 clock scheme. Juli's ice1724 chip is + * supplied by external clock provided by Xilinx array and MK73-1 PLL frequency + * multiplier. Actual frequency is set by ice1724 GPIOs hooked to the Xilinx. + * + * The clock circuitry is supplied by the two ice1724 crystals. This + * arrangement allows to generate independent clock signal for AK4114's input + * rate detection circuit. As a result, Juli, unlike most other + * ice1724+ak4114-based cards, detects spdif input rate correctly. + * This fact is applied in the driver, allowing to modify PCM stream rate + * parameter according to the actual input rate. + * + * Juli uses the remaining three stereo-channels of its DAC to optionally + * monitor analog input, digital input, and digital output. The corresponding + * I2S signals are routed by Xilinx, controlled by GPIOs. + * + * The master mute is implemented using output muting transistors (GPIO) in + * combination with smuting the DAC. + * + * The card itself has no HW master volume control, implemented using the + * vmaster control. + * + * TODO: + * researching and fixing the input monitors + */ + /* * GPIO pins */ @@ -55,17 +83,82 @@ struct juli_spec { #define GPIO_MULTI_2X (1<<2) #define GPIO_MULTI_1X (2<<2) /* also external */ #define GPIO_MULTI_HALF (3<<2) -#define GPIO_INTERNAL_CLOCK (1<<4) +#define GPIO_INTERNAL_CLOCK (1<<4) /* 0 = external, 1 = internal */ +#define GPIO_CLOCK_MASK (1<<4) #define GPIO_ANALOG_PRESENT (1<<5) /* RO only: 0 = present */ #define GPIO_RXMCLK_SEL (1<<7) /* must be 0 */ #define GPIO_AK5385A_CKS0 (1<<8) -#define GPIO_AK5385A_DFS0 (1<<9) /* swapped with DFS1 according doc? */ -#define GPIO_AK5385A_DFS1 (1<<10) +#define GPIO_AK5385A_DFS1 (1<<9) +#define GPIO_AK5385A_DFS0 (1<<10) #define GPIO_DIGOUT_MONITOR (1<<11) /* 1 = active */ #define GPIO_DIGIN_MONITOR (1<<12) /* 1 = active */ #define GPIO_ANAIN_MONITOR (1<<13) /* 1 = active */ -#define GPIO_AK5385A_MCLK (1<<14) /* must be 0 */ -#define GPIO_MUTE_CONTROL (1<<15) /* 0 = off, 1 = on */ +#define GPIO_AK5385A_CKS1 (1<<14) /* must be 0 */ +#define GPIO_MUTE_CONTROL (1<<15) /* output mute, 1 = muted */ + +#define GPIO_RATE_MASK (GPIO_FREQ_MASK | GPIO_MULTI_MASK | \ + GPIO_CLOCK_MASK) +#define GPIO_AK5385A_MASK (GPIO_AK5385A_CKS0 | GPIO_AK5385A_DFS0 | \ + GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS1) + +#define JULI_PCM_RATE (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +#define GPIO_RATE_16000 (GPIO_FREQ_32KHZ | GPIO_MULTI_HALF | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_22050 (GPIO_FREQ_44KHZ | GPIO_MULTI_HALF | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_24000 (GPIO_FREQ_48KHZ | GPIO_MULTI_HALF | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_32000 (GPIO_FREQ_32KHZ | GPIO_MULTI_1X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_44100 (GPIO_FREQ_44KHZ | GPIO_MULTI_1X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_48000 (GPIO_FREQ_48KHZ | GPIO_MULTI_1X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_64000 (GPIO_FREQ_32KHZ | GPIO_MULTI_2X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_88200 (GPIO_FREQ_44KHZ | GPIO_MULTI_2X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_96000 (GPIO_FREQ_48KHZ | GPIO_MULTI_2X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_176400 (GPIO_FREQ_44KHZ | GPIO_MULTI_4X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_192000 (GPIO_FREQ_48KHZ | GPIO_MULTI_4X | \ + GPIO_INTERNAL_CLOCK) + +/* + * Initial setup of the conversion array GPIO <-> rate + */ +static unsigned int juli_rates[] = { + 16000, 22050, 24000, 32000, + 44100, 48000, 64000, 88200, + 96000, 176400, 192000, +}; + +static unsigned int gpio_vals[] = { + GPIO_RATE_16000, GPIO_RATE_22050, GPIO_RATE_24000, GPIO_RATE_32000, + GPIO_RATE_44100, GPIO_RATE_48000, GPIO_RATE_64000, GPIO_RATE_88200, + GPIO_RATE_96000, GPIO_RATE_176400, GPIO_RATE_192000, +}; + +static struct snd_pcm_hw_constraint_list juli_rates_info = { + .count = ARRAY_SIZE(juli_rates), + .list = juli_rates, + .mask = 0, +}; + +static int get_gpio_val(int rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(juli_rates); i++) + if (juli_rates[i] == rate) + return gpio_vals[i]; + return 0; +} static void juli_ak4114_write(void *private_data, unsigned char reg, unsigned char val) { @@ -77,6 +170,10 @@ static unsigned char juli_ak4114_read(void *private_data, unsigned char reg) return snd_vt1724_read_i2c((struct snd_ice1712 *)private_data, AK4114_ADDR, reg); } +/* + * If SPDIF capture and slaved to SPDIF-IN, setting runtime rate + * to the external rate + */ static void juli_spdif_in_open(struct snd_ice1712 *ice, struct snd_pcm_substream *substream) { @@ -84,7 +181,8 @@ static void juli_spdif_in_open(struct snd_ice1712 *ice, struct snd_pcm_runtime *runtime = substream->runtime; int rate; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + !ice->is_spdif_master(ice)) return; rate = snd_ak4114_external_rate(spec->ak4114); if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) { @@ -115,57 +213,285 @@ static void juli_akm_write(struct snd_akm4xxx *ak, int chip, } /* - * change the rate of envy24HT, AK4358 + * change the rate of envy24HT, AK4358, AK5385 */ static void juli_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) { - unsigned char old, tmp, dfs; + unsigned char old, tmp, ak4358_dfs; + unsigned int ak5385_pins, old_gpio, new_gpio; + struct snd_ice1712 *ice = ak->private_data[0]; + struct juli_spec *spec = ice->spec; - if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + if (rate == 0) /* no hint - S/PDIF input is master or the new spdif + input rate undetected, simply return */ return; - + /* adjust DFS on codecs */ - if (rate > 96000) - dfs = 2; - else if (rate > 48000) - dfs = 1; - else - dfs = 0; - + if (rate > 96000) { + ak4358_dfs = 2; + ak5385_pins = GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS0; + } else if (rate > 48000) { + ak4358_dfs = 1; + ak5385_pins = GPIO_AK5385A_DFS0; + } else { + ak4358_dfs = 0; + ak5385_pins = 0; + } + /* AK5385 first, since it requires cold reset affecting both codecs */ + old_gpio = ice->gpio.get_data(ice); + new_gpio = (old_gpio & ~GPIO_AK5385A_MASK) | ak5385_pins; + /* printk(KERN_DEBUG "JULI - ak5385 set_rate_val: new gpio 0x%x\n", + new_gpio); */ + ice->gpio.set_data(ice, new_gpio); + + /* cold reset */ + old = inb(ICEMT1724(ice, AC97_CMD)); + outb(old | VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD)); + udelay(1); + outb(old & ~VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD)); + + /* AK4358 */ + /* set new value, reset DFS */ tmp = snd_akm4xxx_get(ak, 0, 2); - old = (tmp >> 4) & 0x03; - if (old == dfs) - return; - /* reset DFS */ snd_akm4xxx_reset(ak, 1); tmp = snd_akm4xxx_get(ak, 0, 2); tmp &= ~(0x03 << 4); - tmp |= dfs << 4; + tmp |= ak4358_dfs << 4; snd_akm4xxx_set(ak, 0, 2, tmp); snd_akm4xxx_reset(ak, 0); + + /* reinit ak4114 */ + snd_ak4114_reinit(spec->ak4114); } +#define AK_DAC(xname, xch) { .name = xname, .num_channels = xch } +#define PCM_VOLUME "PCM Playback Volume" +#define MONITOR_AN_IN_VOLUME "Monitor Analog In Volume" +#define MONITOR_DIG_IN_VOLUME "Monitor Digital In Volume" +#define MONITOR_DIG_OUT_VOLUME "Monitor Digital Out Volume" + +static const struct snd_akm4xxx_dac_channel juli_dac[] = { + AK_DAC(PCM_VOLUME, 2), + AK_DAC(MONITOR_AN_IN_VOLUME, 2), + AK_DAC(MONITOR_DIG_OUT_VOLUME, 2), + AK_DAC(MONITOR_DIG_IN_VOLUME, 2), +}; + + static struct snd_akm4xxx akm_juli_dac __devinitdata = { .type = SND_AK4358, - .num_dacs = 2, + .num_dacs = 8, /* DAC1 - analog out + DAC2 - analog in monitor + DAC3 - digital out monitor + DAC4 - digital in monitor + */ .ops = { .lock = juli_akm_lock, .unlock = juli_akm_unlock, .write = juli_akm_write, .set_rate_val = juli_akm_set_rate_val - } + }, + .dac_info = juli_dac, }; +#define juli_mute_info snd_ctl_boolean_mono_info + +static int juli_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val; + val = ice->gpio.get_data(ice) & (unsigned int) kcontrol->private_value; + if (kcontrol->private_value == GPIO_MUTE_CONTROL) + /* val 0 = signal on */ + ucontrol->value.integer.value[0] = (val) ? 0 : 1; + else + /* val 1 = signal on */ + ucontrol->value.integer.value[0] = (val) ? 1 : 0; + return 0; +} + +static int juli_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old_gpio, new_gpio; + old_gpio = ice->gpio.get_data(ice); + if (ucontrol->value.integer.value[0]) { + /* unmute */ + if (kcontrol->private_value == GPIO_MUTE_CONTROL) { + /* 0 = signal on */ + new_gpio = old_gpio & ~GPIO_MUTE_CONTROL; + /* un-smuting DAC */ + snd_akm4xxx_write(ice->akm, 0, 0x01, 0x01); + } else + /* 1 = signal on */ + new_gpio = old_gpio | + (unsigned int) kcontrol->private_value; + } else { + /* mute */ + if (kcontrol->private_value == GPIO_MUTE_CONTROL) { + /* 1 = signal off */ + new_gpio = old_gpio | GPIO_MUTE_CONTROL; + /* smuting DAC */ + snd_akm4xxx_write(ice->akm, 0, 0x01, 0x03); + } else + /* 0 = signal off */ + new_gpio = old_gpio & + ~((unsigned int) kcontrol->private_value); + } + /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \ + new_gpio 0x%x\n", + (unsigned int)ucontrol->value.integer.value[0], old_gpio, + new_gpio); */ + if (old_gpio != new_gpio) { + ice->gpio.set_data(ice, new_gpio); + return 1; + } + /* no change */ + return 0; +} + +static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_MUTE_CONTROL, + }, + /* Although the following functionality respects the succint NDA'd + * documentation from the card manufacturer, and the same way of + * operation is coded in OSS Juli driver, only Digital Out monitor + * seems to work. Surprisingly, Analog input monitor outputs Digital + * output data. The two are independent, as enabling both doubles + * volume of the monitor sound. + * + * Checking traces on the board suggests the functionality described + * by the manufacturer is correct - I2S from ADC and AK4114 + * go to ICE as well as to Xilinx, I2S inputs of DAC2,3,4 (the monitor + * inputs) are fed from Xilinx. + * + * I even checked traces on board and coded a support in driver for + * an alternative possiblity - the unused I2S ICE output channels + * switched to HW-IN/SPDIF-IN and providing the monitoring signal to + * the DAC - to no avail. The I2S outputs seem to be unconnected. + * + * The windows driver supports the monitoring correctly. + */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Analog In Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_ANAIN_MONITOR, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Digital Out Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_DIGOUT_MONITOR, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Digital In Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_DIGIN_MONITOR, + }, +}; + + +static void ak4358_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + int reg, val; + for (reg = 0; reg <= 0xf; reg++) { + val = snd_akm4xxx_get(ice->akm, 0, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void ak4358_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "ak4358_codec", &entry)) + snd_info_set_text_ops(entry, ice, ak4358_proc_regs_read); +} + +static char *slave_vols[] __devinitdata = { + PCM_VOLUME, + MONITOR_AN_IN_VOLUME, + MONITOR_DIG_IN_VOLUME, + MONITOR_DIG_OUT_VOLUME, + NULL +}; + +static __devinitdata +DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1); + +static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, + const char *name) +{ + struct snd_ctl_elem_id sid; + memset(&sid, 0, sizeof(sid)); + /* FIXME: strcpy is bad. */ + strcpy(sid.name, name); + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_find_id(card, &sid); +} + +static void __devinit add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + /* printk(KERN_DEBUG "add_slaves - %s\n", *list); */ + if (slave) { + /* printk(KERN_DEBUG "slave %s found\n", *list); */ + snd_ctl_add_slave(master, slave); + } + } +} + static int __devinit juli_add_controls(struct snd_ice1712 *ice) { struct juli_spec *spec = ice->spec; int err; + unsigned int i; + struct snd_kcontrol *vmaster; + err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; + + for (i = 0; i < ARRAY_SIZE(juli_mute_controls); i++) { + err = snd_ctl_add(ice->card, + snd_ctl_new1(&juli_mute_controls[i], ice)); + if (err < 0) + return err; + } + /* Create virtual master control */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + juli_master_db_scale); + if (!vmaster) + return -ENOMEM; + add_slaves(ice->card, vmaster, slave_vols); + err = snd_ctl_add(ice->card, vmaster); + if (err < 0) + return err; + /* only capture SPDIF over AK4114 */ err = snd_ak4114_build(spec->ak4114, NULL, - ice->pcm_pro->streams[SNDRV_PCM_STREAM_CAPTURE].substream); + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); + + ak4358_proc_init(ice); if (err < 0) return err; return 0; @@ -174,6 +500,74 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) /* * initialize the chip */ + +static inline int juli_is_spdif_master(struct snd_ice1712 *ice) +{ + return (ice->gpio.get_data(ice) & GPIO_INTERNAL_CLOCK) ? 0 : 1; +} + +static unsigned int juli_get_rate(struct snd_ice1712 *ice) +{ + int i; + unsigned char result; + + result = ice->gpio.get_data(ice) & GPIO_RATE_MASK; + for (i = 0; i < ARRAY_SIZE(gpio_vals); i++) + if (gpio_vals[i] == result) + return juli_rates[i]; + return 0; +} + +/* setting new rate */ +static void juli_set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned int old, new; + unsigned char val; + + old = ice->gpio.get_data(ice); + new = (old & ~GPIO_RATE_MASK) | get_gpio_val(rate); + /* printk(KERN_DEBUG "JULI - set_rate: old %x, new %x\n", + old & GPIO_RATE_MASK, + new & GPIO_RATE_MASK); */ + + ice->gpio.set_data(ice, new); + /* switching to external clock - supplied by external circuits */ + val = inb(ICEMT1724(ice, RATE)); + outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); +} + +static inline unsigned char juli_set_mclk(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* no change in master clock */ + return 0; +} + +/* setting clock to external - SPDIF */ +static void juli_set_spdif_clock(struct snd_ice1712 *ice) +{ + unsigned int old; + old = ice->gpio.get_data(ice); + /* external clock (= 0), multiply 1x, 48kHz */ + ice->gpio.set_data(ice, (old & ~GPIO_RATE_MASK) | GPIO_MULTI_1X | + GPIO_FREQ_48KHZ); +} + +/* Called when ak4114 detects change in the input SPDIF stream */ +static void juli_ak4114_change(struct ak4114 *ak4114, unsigned char c0, + unsigned char c1) +{ + struct snd_ice1712 *ice = ak4114->change_callback_private; + int rate; + if (ice->is_spdif_master(ice) && c1) { + /* only for SPDIF master mode, rate was changed */ + rate = snd_ak4114_external_rate(ak4114); + /* printk(KERN_DEBUG "ak4114 - input rate changed to %d\n", + rate); */ + juli_akm_set_rate_val(ice->akm, rate); + } +} + static int __devinit juli_init(struct snd_ice1712 *ice) { static const unsigned char ak4114_init_vals[] = { @@ -203,6 +597,11 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice, &spec->ak4114); if (err < 0) return err; + /* callback for codecs rate setting */ + spec->ak4114->change_callback = juli_ak4114_change; + spec->ak4114->change_callback_private = ice; + /* AK4114 in Juli can detect external rate correctly */ + spec->ak4114->check_flags = 0; #if 0 /* it seems that the analog doughter board detection does not work @@ -226,6 +625,14 @@ static int __devinit juli_init(struct snd_ice1712 *ice) return err; } + /* juli is clocked by Xilinx array */ + ice->hw_rates = &juli_rates_info; + ice->is_spdif_master = juli_is_spdif_master; + ice->get_rate = juli_get_rate; + ice->set_rate = juli_set_rate; + ice->set_mclk = juli_set_mclk; + ice->set_spdif_clock = juli_set_spdif_clock; + ice->spdif.ops.open = juli_spdif_in_open; return 0; } @@ -237,18 +644,20 @@ static int __devinit juli_init(struct snd_ice1712 *ice) */ static unsigned char juli_eeprom[] __devinitdata = { - [ICE_EEP2_SYSCONF] = 0x20, /* clock 512, mpu401, 1xADC, 1xDACs */ + [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, 1xADC, 1xDACs, + SPDIF in */ [ICE_EEP2_ACLINK] = 0x80, /* I2S */ [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ - [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR] = 0x9f, /* 5, 6:inputs; 7, 4-0 outputs*/ [ICE_EEP2_GPIO_DIR1] = 0xff, [ICE_EEP2_GPIO_DIR2] = 0x7f, - [ICE_EEP2_GPIO_MASK] = 0x9f, - [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK] = 0x60, /* 5, 6: locked; 7, 4-0 writable */ + [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0-7 writable */ [ICE_EEP2_GPIO_MASK2] = 0x7f, - [ICE_EEP2_GPIO_STATE] = 0x16, /* internal clock, multiple 1x, 48kHz */ - [ICE_EEP2_GPIO_STATE1] = 0x80, /* mute */ + [ICE_EEP2_GPIO_STATE] = GPIO_FREQ_48KHZ | GPIO_MULTI_1X | + GPIO_INTERNAL_CLOCK, /* internal clock, multiple 1x, 48kHz*/ + [ICE_EEP2_GPIO_STATE1] = 0x00, /* unmuted */ [ICE_EEP2_GPIO_STATE2] = 0x00, }; From 07bcb316cf3510d5048bc251bb23cd6452c16fc2 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 20 Mar 2008 12:10:57 +0100 Subject: [PATCH 153/250] [ALSA] hda: 92hd71bxxx DMIC nid Added missing DMIC verb to dell_4_1_pin_configs[]. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8eff8fe9dcfd..69cd3b23f5a7 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1317,7 +1317,7 @@ static unsigned int ref92hd71bxx_pin_configs[10] = { static unsigned int dell_m4_1_pin_configs[13] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, }; From 0e31daf7d6484c60e96f63a905eb9b959b975da5 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Thu, 20 Mar 2008 12:12:39 +0100 Subject: [PATCH 154/250] [ALSA] hda-codec - model for alc262 to support Lenovo 3000 This model is to support the Lenovo 3000 y410. ALSA bug#3856: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3856 Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 100 ++++++++++++++++++++++++++++++++++ 1 file changed, 100 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eb40f4820c8b..8b819072af3a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -98,6 +98,7 @@ enum { ALC262_SONY_ASSAMD, ALC262_BENQ_T31, ALC262_ULTRA, + ALC262_LENOVO_3000, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -8728,6 +8729,12 @@ static struct hda_verb alc262_fujitsu_unsol_verbs[] = { {} }; +static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + static struct hda_input_mux alc262_fujitsu_capture_source = { .num_items = 3, .items = { @@ -8808,6 +8815,46 @@ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { }, }; +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + + if (force || !spec->sense_updated) { + unsigned int present_int_hp; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present_int_hp = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present_int_hp & 0x80000000) != 0; + spec->sense_updated = 1; + } + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC_HP_EVENT) + return; + alc262_lenovo_3000_automute(codec, 1); +} + /* bind hp and internal speaker mute (with plug check) */ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -8849,6 +8896,44 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { } /* end */ }; +/* bind hp and internal speaker mute (with plug check) */ +static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp ? 0 : HDA_AMP_MUTE); + + if (change) + alc262_lenovo_3000_automute(codec, 0); + return change; +} + +static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_lenovo_3000_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + /* additional init verbs for Benq laptops */ static struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -9398,6 +9483,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", [ALC262_ULTRA] = "ultra", + [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_AUTO] = "auto", }; @@ -9434,6 +9520,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), @@ -9596,6 +9683,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_ultra_unsol_event, .init_hook = alc262_ultra_automute, }, + [ALC262_LENOVO_3000] = { + .mixers = { alc262_lenovo_3000_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, + alc262_lenovo_3000_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_capture_source, + .unsol_event = alc262_lenovo_3000_unsol_event, + }, }; static int patch_alc262(struct hda_codec *codec) From 5d85f8d02af56da5e3b76805da00a0f7f7427255 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Thu, 20 Mar 2008 12:13:46 +0100 Subject: [PATCH 155/250] [ALSA] hda-codec - Remove now uneeded 6stack-hp model from ALC883 After DAC assignment fix in ALC883, the 6stack-hp model is now the same as 6stack-dig. So just remove 6stack-hp model and replace its use with 6stack-dig. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 1 - sound/pci/hda/patch_realtek.c | 55 +------------------ 2 files changed, 1 insertion(+), 55 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 42dd8f5855da..0fb62f65938a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -879,7 +879,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 - 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b819072af3a..15b4704539fd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -198,7 +198,6 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC883_HAIER_W66, - ALC888_6ST_HP, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, @@ -6956,46 +6955,6 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7721,7 +7680,6 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC883_HAIER_W66] = "haier-w66", - [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", @@ -7740,7 +7698,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), @@ -7973,17 +7931,6 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, }, - [ALC888_6ST_HP] = { - .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, [ALC888_3ST_HP] = { .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, From f24dbdc61dd7ca6b97c525b40979ab7bd07c0934 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Thu, 20 Mar 2008 12:14:28 +0100 Subject: [PATCH 156/250] [ALSA] hda-codec - Use base ALC883 mixer for 6stack-dell model After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the same as alc883_base_mixer. Avoid duplicated code and use alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer definition. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 42 +---------------------------------- 1 file changed, 1 insertion(+), 41 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 15b4704539fd..e1e966982998 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6993,46 +6993,6 @@ static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7942,7 +7902,7 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, }, [ALC888_6ST_DELL] = { - .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer }, + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, From eea6419ea18ed9dfc16f9a262e96cdb832376e88 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Thu, 20 Mar 2008 12:14:59 +0100 Subject: [PATCH 157/250] [ALSA] hda-codec - Use common 3stack-6ch mixer for 3stack-hp model Forgot one more: 3stack-hp model also have now the same mixer as 3stack-6ch (after DAC assignment fix in ALC883), so use it avoiding duplicating the same mixer definition. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 42 ++--------------------------------- 1 file changed, 2 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1e966982998..198facf01780 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6955,44 +6955,6 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7890,9 +7852,9 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, - }, + }, [ALC888_3ST_HP] = { - .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, From cd97f47df377d2371940b69fa4c8b4d99a980580 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 20 Mar 2008 12:30:36 +0100 Subject: [PATCH 158/250] [ALSA] hda-codec - Add missing models in ALSA-Configuration.txt Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 0fb62f65938a..3eea0675ebb0 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -825,6 +825,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hippo_1 Hippo (Benq) with jack detection sony-assamd Sony ASSAMD ultra Samsung Q1 Ultra Vista model + lenovo-3000 Lenovo 3000 y410 basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) @@ -839,6 +840,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. $CONFIG_SND_DEBUG=y auto auto-config reading BIOS (default) + ALC269 + basic Basic preset + ALC662 3stack-dig 3-stack (2-channel) with SPDIF 3stack-6ch 3-stack (6-channel) @@ -882,6 +886,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D + clevo-m720r Clevo laptop M720R + fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) ALC861/660 From 95866d38028c98ea4d6df6947f6ea3fd77334382 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 22 Mar 2008 10:11:08 +0100 Subject: [PATCH 159/250] [ALSA] ymfpci - Fix race at removal free_irq() must be called first to avoid races at removal. Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci_main.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 42c1eb7d35f5..29b3056c5109 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2249,6 +2249,8 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) #ifdef CONFIG_PM vfree(chip->saved_regs); #endif + if (chip->irq >= 0) + free_irq(chip->irq, chip); release_and_free_resource(chip->mpu_res); release_and_free_resource(chip->fm_res); snd_ymfpci_free_gameport(chip); @@ -2257,8 +2259,6 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) if (chip->work_ptr.area) snd_dma_free_pages(&chip->work_ptr); - if (chip->irq >= 0) - free_irq(chip->irq, chip); release_and_free_resource(chip->res_reg_area); pci_write_config_word(chip->pci, 0x40, chip->old_legacy_ctrl); From c81d80cbf6dfe4c061719cb146659677c3c36c8e Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Sat, 22 Mar 2008 10:12:37 +0100 Subject: [PATCH 160/250] [ALSA] pcsp: remove downsampling pcsp: remove S16->U8 downsampling as dmix now supports U8 natively. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_lib.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 54253e9b4b02..ac6238e93513 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -18,8 +18,6 @@ module_param(nforce_wa, bool, 0444); MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " "(expect bad sound)"); -#define DMIX_WANTS_S16 1 - static void pcsp_start_timer(unsigned long dummy) { hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); @@ -49,7 +47,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) { unsigned long flags; unsigned char timer_cnt, val; - int fmt_size, periods_elapsed; + int periods_elapsed; u64 ns; size_t period_bytes, buffer_bytes; struct snd_pcm_substream *substream; @@ -94,11 +92,8 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) goto exit_nr_unlock2; runtime = substream->runtime; - fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3; - /* assume it is mono! */ - val = runtime->dma_area[chip->playback_ptr + fmt_size - 1]; - if (snd_pcm_format_signed(runtime->format)) - val ^= 0x80; + /* assume it is u8 mono */ + val = runtime->dma_area[chip->playback_ptr]; timer_cnt = val * CUR_DIV() / 256; if (timer_cnt && chip->enable) { @@ -116,7 +111,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - chip->playback_ptr += PCSP_INDEX_INC() * fmt_size; + chip->playback_ptr += PCSP_INDEX_INC(); periods_elapsed = chip->playback_ptr - chip->period_ptr; if (periods_elapsed < 0) { printk(KERN_WARNING "PCSP: playback_ptr inconsistent " @@ -275,11 +270,7 @@ static struct snd_pcm_hardware snd_pcsp_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_HALF_DUPLEX | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), - .formats = (SNDRV_PCM_FMTBIT_U8 -#if DMIX_WANTS_S16 - | SNDRV_PCM_FMTBIT_S16_LE -#endif - ), + .formats = SNDRV_PCM_FMTBIT_U8, .rates = SNDRV_PCM_RATE_KNOT, .rate_min = PCSP_DEFAULT_SRATE, .rate_max = PCSP_DEFAULT_SRATE, From 213f0bfe9061e077590f2775cb90c1e6c0c9faa6 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sat, 22 Mar 2008 10:25:30 +0100 Subject: [PATCH 161/250] [ALSA] hda-codec - Map clevo-m720r ALC883 model for Clevo M720SR Map clevo-m720r ALC883 model for Clevo M720SR. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 198facf01780..e137882d9459 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7650,6 +7650,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), + SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720R), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), From 0c4cc4430f40089bb85557e309038faa458247f1 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sat, 22 Mar 2008 10:26:05 +0100 Subject: [PATCH 162/250] [ALSA] hda-codec - Support mic automute for Clevo M720R/SR Add support for mic automute in clevo-m720r ALC883 model, and rename it to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt accordingly. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 2 +- sound/pci/hda/patch_realtek.c | 53 +++++++++++++------ 2 files changed, 39 insertions(+), 16 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 3eea0675ebb0..08256fcb2743 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -886,7 +886,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D - clevo-m720r Clevo laptop M720R + clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e137882d9459..e5d97c12381d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -201,7 +201,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, - ALC883_CLEVO_M720R, + ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, ALC883_AUTO, ALC883_MODEL_LAST, @@ -6661,7 +6661,7 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { +static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -7130,7 +7130,7 @@ static struct hda_verb alc883_mitac_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_clevo_m720r_verbs[] = { +static struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -7140,6 +7140,7 @@ static struct hda_verb alc883_clevo_m720r_verbs[] = { /* enable unsolicited event */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, { } /* end */ }; @@ -7330,7 +7331,7 @@ static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) } /* toggle speaker-output according to the hp-jack state */ -static void alc883_clevo_m720r_automute(struct hda_codec *codec) +static void alc883_clevo_m720_hp_automute(struct hda_codec *codec) { unsigned int present; unsigned char bits; @@ -7342,11 +7343,33 @@ static void alc883_clevo_m720r_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); } -static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, +static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +static void alc883_clevo_m720_automute(struct hda_codec *codec) +{ + alc883_clevo_m720_hp_automute(codec); + alc883_clevo_m720_mic_automute(codec); +} + +static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc883_clevo_m720r_automute(codec); + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_clevo_m720_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc883_clevo_m720_mic_automute(codec); + break; + } } /* toggle speaker-output according to the hp-jack state */ @@ -7605,7 +7628,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", - [ALC883_CLEVO_M720R] = "clevo-m720r", + [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC883_AUTO] = "auto", }; @@ -7649,8 +7672,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), - SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720R), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), + SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), @@ -7794,17 +7817,17 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, - [ALC883_CLEVO_M720R] = { - .mixers = { alc883_clevo_m720r_mixer }, - .init_verbs = { alc883_init_verbs, alc883_clevo_m720r_verbs }, + [ALC883_CLEVO_M720] = { + .mixers = { alc883_clevo_m720_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_clevo_m720r_unsol_event, - .init_hook = alc883_clevo_m720r_automute, + .unsol_event = alc883_clevo_m720_unsol_event, + .init_hook = alc883_clevo_m720_automute, }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, From e97f79994ac715e4c8724b201bd3328463ec9314 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 1 Apr 2008 10:02:18 +0200 Subject: [PATCH 163/250] [ALSA] oxygen: fix line-in recording selection (now for real) On C-Media cards, the GPIO pin 0 of the CM9780 must be handled exactly like on Xonar cards, so move the Xonar code to the common mixer code. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 33 -------------------- sound/pci/oxygen/oxygen.h | 2 -- sound/pci/oxygen/oxygen_lib.c | 2 ++ sound/pci/oxygen/oxygen_mixer.c | 33 +++++++++++++++++--- sound/pci/oxygen/virtuoso.c | 53 --------------------------------- 5 files changed, 31 insertions(+), 92 deletions(-) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 511ef34a43ca..b3b7771b54c0 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -39,7 +39,6 @@ #include #include "oxygen.h" #include "ak4396.h" -#include "cm9780.h" #include "wm8785.h" MODULE_AUTHOR("Clemens Ladisch "); @@ -79,8 +78,6 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_AK5385_DFS_DOUBLE 0x0001 #define GPIO_AK5385_DFS_QUAD 0x0002 -#define GPIO_LINE_MUTE CM9780_GPO0 - struct generic_data { u8 ak4396_ctl2; }; @@ -145,23 +142,16 @@ static void wm8785_init(struct oxygen *chip) snd_component_add(chip->card, "WM8785"); } -static void cmi9780_init(struct oxygen *chip) -{ - oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE); -} - static void generic_init(struct oxygen *chip) { ak4396_init(chip); wm8785_init(chip); - cmi9780_init(chip); } static void meridian_init(struct oxygen *chip) { ak4396_init(chip); ak5385_init(chip); - cmi9780_init(chip); } static void generic_cleanup(struct oxygen *chip) @@ -257,27 +247,6 @@ static void set_ak5385_params(struct oxygen *chip, value, GPIO_AK5385_DFS_MASK); } -static void cmi9780_switch_hook(struct oxygen *chip, unsigned int codec, - unsigned int reg, int mute) -{ - if (codec != 0) - return; - switch (reg) { - case AC97_LINE: - oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, - mute ? GPIO_LINE_MUTE : 0, - GPIO_LINE_MUTE); - break; - case AC97_MIC: - case AC97_CD: - case AC97_AUX: - if (!mute) - oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_STATUS, - GPIO_LINE_MUTE); - break; - } -} - static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static int ak4396_control_filter(struct snd_kcontrol_new *template) @@ -301,7 +270,6 @@ static const struct oxygen_model model_generic = { .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, - .ac97_switch_hook = cmi9780_switch_hook, .model_data_size = sizeof(struct generic_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | @@ -327,7 +295,6 @@ static const struct oxygen_model model_meridian = { .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, - .ac97_switch_hook = cmi9780_switch_hook, .model_data_size = sizeof(struct generic_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 5103482f65e4..2f25c8dbaf85 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -97,8 +97,6 @@ struct oxygen_model { struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); - void (*ac97_switch_hook)(struct oxygen *chip, unsigned int codec, - unsigned int reg, int mute); void (*gpio_changed)(struct oxygen *chip); size_t model_data_size; unsigned int pcm_dev_cfg; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 858e6d4c9194..4f3d9e5fc5db 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -360,6 +360,8 @@ static void oxygen_init(struct oxygen *chip) oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000); oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080); oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080); + oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, + CM9780_GPO0); /* power down unused ADCs and DACs */ oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN, AC97_PD_PR0 | AC97_PD_PR1); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 6b5ff6e0fadd..9a7c880eddbd 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -510,6 +510,19 @@ static int ac97_switch_get(struct snd_kcontrol *ctl, return 0; } +static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) +{ + unsigned int priv_idx = chip->controls[control]->private_value & 0xff; + u16 value; + + value = oxygen_read_ac97(chip, 0, priv_idx); + if (!(value & 0x8000)) { + oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000); + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->controls[control]->id); + } +} + static int ac97_switch_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -531,9 +544,22 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, change = newreg != oldreg; if (change) { oxygen_write_ac97(chip, codec, index, newreg); - if (bitnr == 15 && chip->model->ac97_switch_hook) - chip->model->ac97_switch_hook(chip, codec, index, - newreg & 0x8000); + if (index == AC97_LINE) { + oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, + newreg & 0x8000 ? + CM9780_GPO0 : 0, CM9780_GPO0); + if (!(newreg & 0x8000)) { + mute_ac97_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH); + mute_ac97_ctl(chip, CONTROL_CD_CAPTURE_SWITCH); + mute_ac97_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH); + } + } else if ((index == AC97_MIC || index == AC97_CD || + index == AC97_VIDEO || index == AC97_AUX) && + bitnr == 15 && !(newreg & 0x8000)) { + mute_ac97_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH); + oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, + CM9780_GPO0, CM9780_GPO0); + } } mutex_unlock(&chip->mutex); return change; @@ -849,7 +875,6 @@ static const struct snd_kcontrol_new ac97_controls[] = { AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC), AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1), AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0), - AC97_VOLUME("Line Capture Volume", 0, AC97_LINE), AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1), AC97_VOLUME("CD Capture Volume", 0, AC97_CD), AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1), diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index fa79db696e2d..2c3daf3ae4c3 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -30,10 +30,6 @@ * GPIO 5 <- external power present (D2X only) * GPIO 7 -> ALT * GPIO 8 -> enable output to speakers - * - * CM9780: - * - * GPIO 0 -> enable AC'97 bypass (line in -> ADC) */ #include @@ -81,8 +77,6 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_ALT 0x0080 #define GPIO_OUTPUT_ENABLE 0x0100 -#define GPIO_LINE_MUTE CM9780_GPO0 - struct xonar_data { u8 is_d2x; u8 has_power; @@ -134,7 +128,6 @@ static void xonar_init(struct oxygen *chip) & GPIO_EXT_POWER); } oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE); msleep(300); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE); oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); @@ -219,49 +212,6 @@ static void xonar_gpio_changed(struct oxygen *chip) } } -static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) -{ - unsigned int priv_idx = chip->controls[control]->private_value & 0xff; - u16 value; - - value = oxygen_read_ac97(chip, 0, priv_idx); - if (!(value & 0x8000)) { - oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->controls[control]->id); - } -} - -static void xonar_ac97_switch_hook(struct oxygen *chip, unsigned int codec, - unsigned int reg, int mute) -{ - if (codec != 0) - return; - /* line-in is exclusive */ - switch (reg) { - case AC97_LINE: - oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, - mute ? GPIO_LINE_MUTE : 0, - GPIO_LINE_MUTE); - if (!mute) { - mute_ac97_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH); - mute_ac97_ctl(chip, CONTROL_CD_CAPTURE_SWITCH); - mute_ac97_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH); - } - break; - case AC97_MIC: - case AC97_CD: - case AC97_VIDEO: - case AC97_AUX: - if (!mute) { - oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_STATUS, - GPIO_LINE_MUTE); - mute_ac97_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH); - } - break; - } -} - static int pcm1796_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -321,8 +271,6 @@ static int xonar_control_filter(struct snd_kcontrol_new *template) } else if (!strncmp(template->name, "CD Capture ", 11)) { /* CD in is actually connected to the video in pin */ template->private_value ^= AC97_CD ^ AC97_VIDEO; - } else if (!strcmp(template->name, "Line Capture Volume")) { - return 1; /* line-in bypasses the AC'97 mixer */ } return 0; } @@ -345,7 +293,6 @@ static const struct oxygen_model model_xonar = { .set_adc_params = set_cs5381_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, - .ac97_switch_hook = xonar_ac97_switch_hook, .gpio_changed = xonar_gpio_changed, .model_data_size = sizeof(struct xonar_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | From edab938e63e463da86e4aa7b94628ce8f2b8a137 Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Tue, 1 Apr 2008 15:33:22 +0200 Subject: [PATCH 164/250] [ALSA] fix comments in sound/core.h Two sentences seem to be spliced into one in comment, fix that and fix english. Also fix codingstyle. Signed-off-by: Pavel Machek Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/core.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 4fc0235ad784..356659de6b85 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -310,7 +310,7 @@ int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); #ifndef snd_card_set_dev -#define snd_card_set_dev(card,devptr) ((card)->dev = (devptr)) +#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) #endif /* device.c */ @@ -373,7 +373,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * snd_printd - debug printk * @fmt: format string * - * Compiled only when Works like snd_printk() for debugging purpose. + * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG is not set. */ #define snd_printd(fmt, args...) \ @@ -417,7 +417,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * snd_printdd - debug printk * @format: format string * - * Compiled only when Works like snd_printk() for debugging purpose. + * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG_DETECT is not set. */ #define snd_printdd(format, args...) snd_printk(format, ##args) From 7a4356747298d1c899a12a25260d5ff1b4feeb5e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 2 Apr 2008 10:56:30 +0200 Subject: [PATCH 165/250] [ALSA] aw2: remove duplicate MODULE_LICENSE "GPL 2" does not mean that there have to be two MODULE_LICENSE("GPL") entries. ;-) Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-alsa.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 12d01c9c6637..56f87cd33c19 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -36,7 +36,6 @@ #include "saa7146.h" #include "aw2-saa7146.h" -MODULE_LICENSE("GPL"); MODULE_AUTHOR("Cedric Bregardis , " "Jean-Christian Hassler "); MODULE_DESCRIPTION("Emagic Audiowerk 2 sound driver"); From 10e6d5f9b6edd4a12d678716d7fdb94278a83227 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:23:37 +0200 Subject: [PATCH 166/250] [ALSA] oxygen: add I2C support Add a function to write I2C registers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_io.c | 23 +++++++++++++++++++++-- 2 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 2f25c8dbaf85..d53c18c6fcd8 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -146,6 +146,7 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, unsigned int index, u16 data, u16 mask); void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data); +void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data); static inline void oxygen_set_bits8(struct oxygen *chip, unsigned int reg, u8 value) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 74e23ef9c946..5569606ee87f 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -190,12 +190,31 @@ void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) --count; } - spin_lock_irq(&chip->reg_lock); oxygen_write8(chip, OXYGEN_SPI_DATA1, data); oxygen_write8(chip, OXYGEN_SPI_DATA2, data >> 8); if (control & OXYGEN_SPI_DATA_LENGTH_3) oxygen_write8(chip, OXYGEN_SPI_DATA3, data >> 16); oxygen_write8(chip, OXYGEN_SPI_CONTROL, control); - spin_unlock_irq(&chip->reg_lock); } EXPORT_SYMBOL(oxygen_write_spi); + +void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data) +{ + unsigned long timeout; + + /* should not need more than about 300 us */ + timeout = jiffies + msecs_to_jiffies(1); + do { + if (!(oxygen_read16(chip, OXYGEN_2WIRE_BUS_STATUS) + & OXYGEN_2WIRE_BUSY)) + break; + udelay(1); + cond_resched(); + } while (time_after_eq(timeout, jiffies)); + + oxygen_write8(chip, OXYGEN_2WIRE_MAP, map); + oxygen_write8(chip, OXYGEN_2WIRE_DATA, data); + oxygen_write8(chip, OXYGEN_2WIRE_CONTROL, + device | OXYGEN_2WIRE_DIR_WRITE); +} +EXPORT_SYMBOL(oxygen_write_i2c); From 271ebfca5823875cc4f134515b6c3887d99b8dc2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:24:22 +0200 Subject: [PATCH 167/250] [ALSA] virtuoso: separate D2/D2X init functions Use separate model structures for the D2 and D2X so that the init function does not have to check for the model again. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 118 ++++++++++++++++++++++-------------- 1 file changed, 74 insertions(+), 44 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 2c3daf3ae4c3..b3259fa0a0b6 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -61,9 +61,14 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); +enum { + MODEL_D2, + MODEL_D2X, +}; + static struct pci_device_id xonar_ids[] __devinitdata = { - { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, /* Asus Xonar D2 */ - { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, /* Asus Xonar D2X */ + { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 }, + { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, { } }; MODULE_DEVICE_TABLE(pci, xonar_ids); @@ -78,7 +83,6 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_OUTPUT_ENABLE 0x0100 struct xonar_data { - u8 is_d2x; u8 has_power; }; @@ -97,13 +101,10 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, (reg << 8) | value); } -static void xonar_init(struct oxygen *chip) +static void xonar_d2_init(struct oxygen *chip) { - struct xonar_data *data = chip->model_data; unsigned int i; - data->is_d2x = chip->pci->subsystem_device == 0x82b7; - for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 18, PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); @@ -118,15 +119,6 @@ static void xonar_init(struct oxygen *chip) oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, GPIO_CS5381_M_SINGLE, GPIO_CS5381_M_MASK | GPIO_ALT); - if (data->is_d2x) { - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_EXT_POWER); - oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, - GPIO_EXT_POWER); - chip->interrupt_mask |= OXYGEN_INT_GPIO; - data->has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) - & GPIO_EXT_POWER); - } oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); msleep(300); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE); @@ -136,6 +128,18 @@ static void xonar_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void xonar_d2x_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + xonar_d2_init(chip); + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_EXT_POWER); + oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_EXT_POWER); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + data->has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) + & GPIO_EXT_POWER); +} + static void xonar_cleanup(struct oxygen *chip) { oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); @@ -196,8 +200,6 @@ static void xonar_gpio_changed(struct oxygen *chip) struct xonar_data *data = chip->model_data; u8 has_power; - if (!data->is_d2x) - return; has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_EXT_POWER); if (has_power != data->has_power) { @@ -280,31 +282,58 @@ static int xonar_mixer_init(struct oxygen *chip) return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); } -static const struct oxygen_model model_xonar = { - .shortname = "Asus AV200", - .longname = "Asus Virtuoso 200", - .chip = "AV200", - .owner = THIS_MODULE, - .init = xonar_init, - .control_filter = xonar_control_filter, - .mixer_init = xonar_mixer_init, - .cleanup = xonar_cleanup, - .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs5381_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .gpio_changed = xonar_gpio_changed, - .model_data_size = sizeof(struct xonar_data), - .pcm_dev_cfg = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF, - .dac_channels = 8, - .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_SPI | - OXYGEN_FUNCTION_ENABLE_SPI_4_5, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +static const struct oxygen_model xonar_models[] = { + [MODEL_D2] = { + .shortname = "Asus AV200", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .owner = THIS_MODULE, + .init = xonar_d2_init, + .control_filter = xonar_control_filter, + .mixer_init = xonar_mixer_init, + .cleanup = xonar_cleanup, + .set_dac_params = set_pcm1796_params, + .set_adc_params = set_cs5381_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .model_data_size = sizeof(struct xonar_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, + .dac_channels = 8, + .misc_flags = OXYGEN_MISC_MIDI, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + }, + [MODEL_D2X] = { + .shortname = "Asus AV200", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .owner = THIS_MODULE, + .init = xonar_d2x_init, + .control_filter = xonar_control_filter, + .mixer_init = xonar_mixer_init, + .cleanup = xonar_cleanup, + .set_dac_params = set_pcm1796_params, + .set_adc_params = set_cs5381_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .gpio_changed = xonar_gpio_changed, + .model_data_size = sizeof(struct xonar_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, + .dac_channels = 8, + .misc_flags = OXYGEN_MISC_MIDI, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + }, }; static int __devinit xonar_probe(struct pci_dev *pci, @@ -319,7 +348,8 @@ static int __devinit xonar_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], &model_xonar); + err = oxygen_pci_probe(pci, index[dev], id[dev], + &xonar_models[pci_id->driver_data]); if (err >= 0) ++dev; return err; From a694a6a0e4ab4752d1a145b9b32e231d7c9611b5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:25:30 +0200 Subject: [PATCH 168/250] [ALSA] virtuoso: allow both CS5381 and CS5361 Rename all CS5381 symbols to CS53x1 because they can also be used for Xonar models with a CS5361. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 29 +++++++++++++++-------------- 1 file changed, 15 insertions(+), 14 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index b3259fa0a0b6..1627197f1689 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -74,10 +74,11 @@ static struct pci_device_id xonar_ids[] __devinitdata = { MODULE_DEVICE_TABLE(pci, xonar_ids); -#define GPIO_CS5381_M_MASK 0x000c -#define GPIO_CS5381_M_SINGLE 0x0000 -#define GPIO_CS5381_M_DOUBLE 0x0004 -#define GPIO_CS5381_M_QUAD 0x0008 +#define GPIO_CS53x1_M_MASK 0x000c +#define GPIO_CS53x1_M_SINGLE 0x0000 +#define GPIO_CS53x1_M_DOUBLE 0x0004 +#define GPIO_CS53x1_M_QUAD 0x0008 + #define GPIO_EXT_POWER 0x0020 #define GPIO_ALT 0x0080 #define GPIO_OUTPUT_ENABLE 0x0100 @@ -115,10 +116,10 @@ static void xonar_d2_init(struct oxygen *chip) } oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_CS5381_M_MASK | GPIO_ALT); + GPIO_CS53x1_M_MASK | GPIO_ALT); oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - GPIO_CS5381_M_SINGLE, - GPIO_CS5381_M_MASK | GPIO_ALT); + GPIO_CS53x1_M_SINGLE, + GPIO_CS53x1_M_MASK | GPIO_ALT); oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); msleep(300); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE); @@ -180,19 +181,19 @@ static void update_pcm1796_mute(struct oxygen *chip) pcm1796_write(chip, i, 18, value); } -static void set_cs5381_params(struct oxygen *chip, +static void set_cs53x1_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { unsigned int value; if (params_rate(params) <= 54000) - value = GPIO_CS5381_M_SINGLE; + value = GPIO_CS53x1_M_SINGLE; else if (params_rate(params) <= 108000) - value = GPIO_CS5381_M_DOUBLE; + value = GPIO_CS53x1_M_DOUBLE; else - value = GPIO_CS5381_M_QUAD; + value = GPIO_CS53x1_M_QUAD; oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - value, GPIO_CS5381_M_MASK); + value, GPIO_CS53x1_M_MASK); } static void xonar_gpio_changed(struct oxygen *chip) @@ -293,7 +294,7 @@ static const struct oxygen_model xonar_models[] = { .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs5381_params, + .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .model_data_size = sizeof(struct xonar_data), @@ -318,7 +319,7 @@ static const struct oxygen_model xonar_models[] = { .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs5381_params, + .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .gpio_changed = xonar_gpio_changed, From af9af1741f5e7959d220fb0d83604ecb5ae26581 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:26:03 +0200 Subject: [PATCH 169/250] [ALSA] virtuoso: move some code to xonar_common_init() Move the code that is common to all Xonar models to a separate function, and make it more generic in preparation for another model. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 77 ++++++++++++++++++++++++------------- 1 file changed, 51 insertions(+), 26 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 1627197f1689..95c229acf855 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -79,11 +79,16 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_CS53x1_M_DOUBLE 0x0004 #define GPIO_CS53x1_M_QUAD 0x0008 -#define GPIO_EXT_POWER 0x0020 -#define GPIO_ALT 0x0080 -#define GPIO_OUTPUT_ENABLE 0x0100 +#define GPIO_D2X_EXT_POWER 0x0020 +#define GPIO_D2_ALT 0x0080 +#define GPIO_D2_OUTPUT_ENABLE 0x0100 struct xonar_data { + unsigned int anti_pop_delay; + u16 output_enable_bit; + u8 ext_power_reg; + u8 ext_power_int_reg; + u8 ext_power_bit; u8 has_power; }; @@ -102,10 +107,34 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, (reg << 8) | value); } +static void xonar_common_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + if (data->ext_power_reg) { + oxygen_set_bits8(chip, data->ext_power_int_reg, + data->ext_power_bit); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) + & data->ext_power_bit); + } + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); + oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); + msleep(data->anti_pop_delay); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + static void xonar_d2_init(struct oxygen *chip) { + struct xonar_data *data = chip->model_data; unsigned int i; + data->anti_pop_delay = 300; + data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 18, PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); @@ -115,15 +144,10 @@ static void xonar_d2_init(struct oxygen *chip) pcm1796_write(chip, i, 17, 0xff); } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_CS53x1_M_MASK | GPIO_ALT); - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - GPIO_CS53x1_M_SINGLE, - GPIO_CS53x1_M_MASK | GPIO_ALT); - oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - msleep(300); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); + + xonar_common_init(chip); snd_component_add(chip->card, "PCM1796"); snd_component_add(chip->card, "CS5381"); @@ -133,17 +157,18 @@ static void xonar_d2x_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; + data->ext_power_reg = OXYGEN_GPIO_DATA; + data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; + data->ext_power_bit = GPIO_D2X_EXT_POWER; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); xonar_d2_init(chip); - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_EXT_POWER); - oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_EXT_POWER); - chip->interrupt_mask |= OXYGEN_INT_GPIO; - data->has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) - & GPIO_EXT_POWER); } static void xonar_cleanup(struct oxygen *chip) { - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); + struct xonar_data *data = chip->model_data; + + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); } static void set_pcm1796_params(struct oxygen *chip, @@ -201,8 +226,8 @@ static void xonar_gpio_changed(struct oxygen *chip) struct xonar_data *data = chip->model_data; u8 has_power; - has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) - & GPIO_EXT_POWER); + has_power = !!(oxygen_read8(chip, data->ext_power_reg) + & data->ext_power_bit); if (has_power != data->has_power) { data->has_power = has_power; if (has_power) { @@ -231,7 +256,7 @@ static int alt_switch_get(struct snd_kcontrol *ctl, struct oxygen *chip = ctl->private_data; value->value.integer.value[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_ALT); + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_D2_ALT); return 0; } @@ -245,9 +270,9 @@ static int alt_switch_put(struct snd_kcontrol *ctl, spin_lock_irq(&chip->reg_lock); old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); if (value->value.integer.value[0]) - new_bits = old_bits | GPIO_ALT; + new_bits = old_bits | GPIO_D2_ALT; else - new_bits = old_bits & ~GPIO_ALT; + new_bits = old_bits & ~GPIO_D2_ALT; changed = new_bits != old_bits; if (changed) oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); @@ -265,7 +290,7 @@ static const struct snd_kcontrol_new alt_switch = { static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); -static int xonar_control_filter(struct snd_kcontrol_new *template) +static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { if (!strcmp(template->name, "Master Playback Volume")) { template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; @@ -290,7 +315,7 @@ static const struct oxygen_model xonar_models[] = { .chip = "AV200", .owner = THIS_MODULE, .init = xonar_d2_init, - .control_filter = xonar_control_filter, + .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, .set_dac_params = set_pcm1796_params, @@ -315,7 +340,7 @@ static const struct oxygen_model xonar_models[] = { .chip = "AV200", .owner = THIS_MODULE, .init = xonar_d2x_init, - .control_filter = xonar_control_filter, + .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, .set_dac_params = set_pcm1796_params, From d08267a9df99c3cf288ca05e75084d14479fe7cb Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:26:26 +0200 Subject: [PATCH 170/250] [ALSA] virtuoso: set PCM1796 oversampling rate When playing data at 96 kHz or higher, reduce the DAC oversampling rate to 32. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 95c229acf855..c6bd31b9f4b5 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -174,14 +174,12 @@ static void xonar_cleanup(struct oxygen *chip) static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { -#if 0 unsigned int i; u8 value; value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; for (i = 0; i < 4; ++i) pcm1796_write(chip, i, 20, value); -#endif } static void update_pcm1796_volume(struct oxygen *chip) From aef1a535c4dadff408412833b2b71bc7919e84a6 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:26:45 +0200 Subject: [PATCH 171/250] [ALSA] virtuoso: change card short name Change the card short name to show to show the card name instead of the chip name. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index c6bd31b9f4b5..2d7b96e3f49f 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -308,7 +308,7 @@ static int xonar_mixer_init(struct oxygen *chip) static const struct oxygen_model xonar_models[] = { [MODEL_D2] = { - .shortname = "Asus AV200", + .shortname = "Xonar D2", .longname = "Asus Virtuoso 200", .chip = "AV200", .owner = THIS_MODULE, @@ -333,7 +333,7 @@ static const struct oxygen_model xonar_models[] = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }, [MODEL_D2X] = { - .shortname = "Asus AV200", + .shortname = "Xonar D2X", .longname = "Asus Virtuoso 200", .chip = "AV200", .owner = THIS_MODULE, From 80647ee26e96d6394cab77332c69f60735396e67 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:27:01 +0200 Subject: [PATCH 172/250] [ALSA] virtuoso: fix typo Fix a (fortunately harmless) typo. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 2d7b96e3f49f..9459ca0a1202 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -292,7 +292,7 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { if (!strcmp(template->name, "Master Playback Volume")) { template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->info = pcm1796_volume_info, + template->info = pcm1796_volume_info; template->tlv.p = pcm1796_db_scale; } else if (!strncmp(template->name, "CD Capture ", 11)) { /* CD in is actually connected to the video in pin */ From a9d3cc485e65a56edc9ef78c034146cc8a5b3101 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 7 Apr 2008 10:29:44 +0200 Subject: [PATCH 173/250] [ALSA] virtuoso: add Xonar DX support Add support for the Asus Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 4 +- sound/pci/oxygen/cs4362a.h | 69 +++++++++++ sound/pci/oxygen/cs4398.h | 69 +++++++++++ sound/pci/oxygen/virtuoso.c | 227 +++++++++++++++++++++++++++++++++++- 4 files changed, 365 insertions(+), 4 deletions(-) create mode 100644 sound/pci/oxygen/cs4362a.h create mode 100644 sound/pci/oxygen/cs4398.h diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index b5903eed6ef5..581debf37dcb 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -914,12 +914,12 @@ config SND_VIA82XX_MODEM will be called snd-via82xx-modem. config SND_VIRTUOSO - tristate "Asus Virtuoso 200 (Xonar)" + tristate "Asus Virtuoso 100/200 (Xonar)" depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV200 chip, i.e., Xonar D2 and Xonar D2X. + Asus AV100/AV200 chips, i.e., Xonar D2, DX and D2X. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/oxygen/cs4362a.h b/sound/pci/oxygen/cs4362a.h new file mode 100644 index 000000000000..6a4fedf5e1ec --- /dev/null +++ b/sound/pci/oxygen/cs4362a.h @@ -0,0 +1,69 @@ +/* register 01h */ +#define CS4362A_PDN 0x01 +#define CS4362A_DAC1_DIS 0x02 +#define CS4362A_DAC2_DIS 0x04 +#define CS4362A_DAC3_DIS 0x08 +#define CS4362A_MCLKDIV 0x20 +#define CS4362A_FREEZE 0x40 +#define CS4362A_CPEN 0x80 +/* register 02h */ +#define CS4362A_DIF_MASK 0x70 +#define CS4362A_DIF_LJUST 0x00 +#define CS4362A_DIF_I2S 0x10 +#define CS4362A_DIF_RJUST_16 0x20 +#define CS4362A_DIF_RJUST_24 0x30 +#define CS4362A_DIF_RJUST_20 0x40 +#define CS4362A_DIF_RJUST_18 0x50 +/* register 03h */ +#define CS4362A_MUTEC_MASK 0x03 +#define CS4362A_MUTEC_6 0x00 +#define CS4362A_MUTEC_1 0x01 +#define CS4362A_MUTEC_3 0x03 +#define CS4362A_AMUTE 0x04 +#define CS4362A_MUTEC_POL 0x08 +#define CS4362A_RMP_UP 0x10 +#define CS4362A_SNGLVOL 0x20 +#define CS4362A_ZERO_CROSS 0x40 +#define CS4362A_SOFT_RAMP 0x80 +/* register 04h */ +#define CS4362A_RMP_DN 0x01 +#define CS4362A_DEM_MASK 0x06 +#define CS4362A_DEM_NONE 0x00 +#define CS4362A_DEM_44100 0x02 +#define CS4362A_DEM_48000 0x04 +#define CS4362A_DEM_32000 0x06 +#define CS4362A_FILT_SEL 0x10 +/* register 05h */ +#define CS4362A_INV_A1 0x01 +#define CS4362A_INV_B1 0x02 +#define CS4362A_INV_A2 0x04 +#define CS4362A_INV_B2 0x08 +#define CS4362A_INV_A3 0x10 +#define CS4362A_INV_B3 0x20 +/* register 06h */ +#define CS4362A_FM_MASK 0x03 +#define CS4362A_FM_SINGLE 0x00 +#define CS4362A_FM_DOUBLE 0x01 +#define CS4362A_FM_QUAD 0x02 +#define CS4362A_FM_DSD 0x03 +#define CS4362A_ATAPI_MASK 0x7c +#define CS4362A_ATAPI_B_MUTE 0x00 +#define CS4362A_ATAPI_B_R 0x04 +#define CS4362A_ATAPI_B_L 0x08 +#define CS4362A_ATAPI_B_LR 0x0c +#define CS4362A_ATAPI_A_MUTE 0x00 +#define CS4362A_ATAPI_A_R 0x10 +#define CS4362A_ATAPI_A_L 0x20 +#define CS4362A_ATAPI_A_LR 0x30 +#define CS4362A_ATAPI_MIX_LR_VOL 0x40 +#define CS4362A_A_EQ_B 0x80 +/* register 07h */ +#define CS4362A_VOL_MASK 0x7f +#define CS4362A_MUTE 0x80 +/* register 08h: like 07h */ +/* registers 09h..0Bh: like 06h..08h */ +/* registers 0Ch..0Eh: like 06h..08h */ +/* register 12h */ +#define CS4362A_REV_MASK 0x07 +#define CS4362A_PART_MASK 0xf8 +#define CS4362A_PART_CS4362A 0x50 diff --git a/sound/pci/oxygen/cs4398.h b/sound/pci/oxygen/cs4398.h new file mode 100644 index 000000000000..5faf5efc8826 --- /dev/null +++ b/sound/pci/oxygen/cs4398.h @@ -0,0 +1,69 @@ +/* register 1 */ +#define CS4398_REV_MASK 0x07 +#define CS4398_PART_MASK 0xf8 +#define CS4398_PART_CS4398 0x70 +/* register 2 */ +#define CS4398_FM_MASK 0x03 +#define CS4398_FM_SINGLE 0x00 +#define CS4398_FM_DOUBLE 0x01 +#define CS4398_FM_QUAD 0x02 +#define CS4398_FM_DSD 0x03 +#define CS4398_DEM_MASK 0x0c +#define CS4398_DEM_NONE 0x00 +#define CS4398_DEM_44100 0x04 +#define CS4398_DEM_48000 0x08 +#define CS4398_DEM_32000 0x0c +#define CS4398_DIF_MASK 0x70 +#define CS4398_DIF_LJUST 0x00 +#define CS4398_DIF_I2S 0x10 +#define CS4398_DIF_RJUST_16 0x20 +#define CS4398_DIF_RJUST_24 0x30 +#define CS4398_DIF_RJUST_20 0x40 +#define CS4398_DIF_RJUST_18 0x50 +#define CS4398_DSD_SRC 0x80 +/* register 3 */ +#define CS4398_ATAPI_MASK 0x1f +#define CS4398_ATAPI_B_MUTE 0x00 +#define CS4398_ATAPI_B_R 0x01 +#define CS4398_ATAPI_B_L 0x02 +#define CS4398_ATAPI_B_LR 0x03 +#define CS4398_ATAPI_A_MUTE 0x00 +#define CS4398_ATAPI_A_R 0x04 +#define CS4398_ATAPI_A_L 0x08 +#define CS4398_ATAPI_A_LR 0x0c +#define CS4398_ATAPI_MIX_LR_VOL 0x10 +#define CS4398_INVERT_B 0x20 +#define CS4398_INVERT_A 0x40 +#define CS4398_VOL_B_EQ_A 0x80 +/* register 4 */ +#define CS4398_MUTEP_MASK 0x03 +#define CS4398_MUTEP_AUTO 0x00 +#define CS4398_MUTEP_LOW 0x02 +#define CS4398_MUTEP_HIGH 0x03 +#define CS4398_MUTE_B 0x08 +#define CS4398_MUTE_A 0x10 +#define CS4398_MUTEC_A_EQ_B 0x20 +#define CS4398_DAMUTE 0x40 +#define CS4398_PAMUTE 0x80 +/* register 5 */ +#define CS4398_VOL_A_MASK 0xff +/* register 6 */ +#define CS4398_VOL_B_MASK 0xff +/* register 7 */ +#define CS4398_DIR_DSD 0x01 +#define CS4398_FILT_SEL 0x04 +#define CS4398_RMP_DN 0x10 +#define CS4398_RMP_UP 0x20 +#define CS4398_ZERO_CROSS 0x40 +#define CS4398_SOFT_RAMP 0x80 +/* register 8 */ +#define CS4398_MCLKDIV3 0x08 +#define CS4398_MCLKDIV2 0x10 +#define CS4398_FREEZE 0x20 +#define CS4398_CPEN 0x40 +#define CS4398_PDN 0x80 +/* register 9 */ +#define CS4398_DSD_PM_EN 0x01 +#define CS4398_DSD_PM_MODE 0x02 +#define CS4398_INVALID_DSD 0x04 +#define CS4398_STATIC_DSD 0x08 diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 9459ca0a1202..1db4aa5dfad4 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -18,6 +18,9 @@ */ /* + * Xonar D2/D2X + * ------------ + * * CMI8788: * * SPI 0 -> 1st PCM1796 (front) @@ -32,6 +35,33 @@ * GPIO 8 -> enable output to speakers */ +/* + * Xonar DX + * -------- + * + * CMI8788: + * + * I²C <-> CS4398 (front) + * <-> CS4362A (surround, center/LFE, back) + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> ALT? + * GPIO 2 -> M0 of CS5361 + * GPIO 3 -> M1 of CS5361 + * GPIO 8 -> line-in/mic-in/digital-out switch? + * + * CS4398: + * + * AD0 <- 1 + * AD1 <- 1 + * + * CS4362A: + * + * AD0 <- 0 + */ + #include #include #include @@ -44,11 +74,13 @@ #include "oxygen.h" #include "cm9780.h" #include "pcm1796.h" +#include "cs4398.h" +#include "cs4362a.h" MODULE_AUTHOR("Clemens Ladisch "); -MODULE_DESCRIPTION("Asus AV200 driver"); +MODULE_DESCRIPTION("Asus AVx00 driver"); MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Asus,AV200}}"); +MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -64,10 +96,12 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_D2, MODEL_D2X, + MODEL_DX, }; static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 }, + { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, { } }; @@ -83,6 +117,14 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_D2_ALT 0x0080 #define GPIO_D2_OUTPUT_ENABLE 0x0100 +#define GPI_DX_EXT_POWER 0x01 +#define GPIO_DX_OUTPUT_ENABLE 0x0001 +#define GPIO_DX_UNKNOWN1 0x0002 +#define GPIO_DX_UNKNOWN2 0x0100 + +#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ +#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ + struct xonar_data { unsigned int anti_pop_delay; u16 output_enable_bit; @@ -107,6 +149,16 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, (reg << 8) | value); } +static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); +} + +static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); +} + static void xonar_common_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -164,6 +216,66 @@ static void xonar_d2x_init(struct oxygen *chip) xonar_d2_init(chip); } +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + unsigned int i; + + for (i = 0; i < 8; ++i) + chip->dac_volume[i] = 127; + data->anti_pop_delay = 800; + data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + + /* XXX the DACs' datasheets say fast mode is not allowed */ + oxygen_set_bits16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_SPEED_FAST); + + /* set CPEN (control port mode) and power down */ + cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); + cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); + /* configure */ + cs4398_write(chip, 2, CS4398_FM_SINGLE | + CS4398_DEM_NONE | CS4398_DIF_LJUST); + cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); + cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); + cs4398_write(chip, 5, 0); + cs4398_write(chip, 6, 0); + cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | + CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); + cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); + cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE | + CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); + cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); + cs4362a_write(chip, 0x05, 0); + cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x07, 0); + cs4362a_write(chip, 0x08, 0); + cs4362a_write(chip, 0x0a, 0); + cs4362a_write(chip, 0x0b, 0); + cs4362a_write(chip, 0x0d, 0); + cs4362a_write(chip, 0x0e, 0); + /* clear power down */ + cs4398_write(chip, 8, CS4398_CPEN); + cs4362a_write(chip, 0x01, CS4362A_CPEN); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DX_UNKNOWN1 | GPIO_DX_UNKNOWN2); + + xonar_common_init(chip); + + snd_component_add(chip->card, "CS4398"); + snd_component_add(chip->card, "CS4362A"); + snd_component_add(chip->card, "CS5361"); +} + static void xonar_cleanup(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -171,6 +283,13 @@ static void xonar_cleanup(struct oxygen *chip) oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); } +static void xonar_dx_cleanup(struct oxygen *chip) +{ + xonar_cleanup(chip); + cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); + oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -219,6 +338,60 @@ static void set_cs53x1_params(struct oxygen *chip, value, GPIO_CS53x1_M_MASK); } +static void set_cs43xx_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + u8 fm_cs4398, fm_cs4362a; + + fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST; + fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + if (params_rate(params) <= 50000) { + fm_cs4398 |= CS4398_FM_SINGLE; + fm_cs4362a |= CS4362A_FM_SINGLE; + } else if (params_rate(params) <= 100000) { + fm_cs4398 |= CS4398_FM_DOUBLE; + fm_cs4362a |= CS4362A_FM_DOUBLE; + } else { + fm_cs4398 |= CS4398_FM_QUAD; + fm_cs4362a |= CS4362A_FM_QUAD; + } + cs4398_write(chip, 2, fm_cs4398); + cs4362a_write(chip, 0x06, fm_cs4362a); + cs4362a_write(chip, 0x09, fm_cs4362a); + cs4362a_write(chip, 0x0c, fm_cs4362a); +} + +static void update_cs4362a_volumes(struct oxygen *chip) +{ + u8 mute; + + mute = chip->dac_mute ? CS4362A_MUTE : 0; + cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); + cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); + cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); + cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); + cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); + cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); +} + +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write(chip, 4, reg); + update_cs4362a_volumes(chip); +} + static void xonar_gpio_changed(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -248,6 +421,16 @@ static int pcm1796_volume_info(struct snd_kcontrol *ctl, return 0; } +static int cs4362a_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 8; + info->value.integer.min = 0; + info->value.integer.max = 127; + return 0; +} + static int alt_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -287,6 +470,7 @@ static const struct snd_kcontrol_new alt_switch = { }; static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); +static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { @@ -301,6 +485,23 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) return 0; } +static int xonar_dx_control_filter(struct snd_kcontrol_new *template) +{ + if (!strcmp(template->name, "Master Playback Volume")) { + template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + template->info = cs4362a_volume_info; + template->tlv.p = cs4362a_db_scale; + } else if (!strncmp(template->name, "CD Capture ", 11)) { + return 1; /* no CD input */ + } else if (!strcmp(template->name, + SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK)) || + !strcmp(template->name, + SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) { + return 1; /* no digital input */ + } + return 0; +} + static int xonar_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); @@ -358,6 +559,28 @@ static const struct oxygen_model xonar_models[] = { .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }, + [MODEL_DX] = { + .shortname = "Xonar DX", + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .owner = THIS_MODULE, + .init = xonar_dx_init, + .control_filter = xonar_dx_control_filter, + .cleanup = xonar_dx_cleanup, + .set_dac_params = set_cs43xx_params, + .set_adc_params = set_cs53x1_params, + .update_dac_volume = update_cs43xx_volume, + .update_dac_mute = update_cs43xx_mute, + .gpio_changed = xonar_gpio_changed, + .model_data_size = sizeof(struct xonar_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 8, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + }, }; static int __devinit xonar_probe(struct pci_dev *pci, From 11864b4b84194b459fc20e0ec47906885bddb12e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 9 Apr 2008 09:16:14 +0200 Subject: [PATCH 174/250] [ALSA] virtuoso: correctly switch input jack on Xonar DX When selecting the capture source on the Xonar DX, the input jack must be routed to either the line input or the microphone input by setting a GPIO pin. This requires an additional callback so that the model driver can hook into the toggling of AC97 switches. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 2 ++ sound/pci/oxygen/oxygen_mixer.c | 4 ++++ sound/pci/oxygen/virtuoso.c | 22 ++++++++++++++++++---- 3 files changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index d53c18c6fcd8..7efbf54bc4ec 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -98,6 +98,8 @@ struct oxygen_model { void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); void (*gpio_changed)(struct oxygen *chip); + void (*ac97_switch)(struct oxygen *chip, + unsigned int reg, unsigned int mute); size_t model_data_size; unsigned int pcm_dev_cfg; u8 dac_channels; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 9a7c880eddbd..d0bef09a6999 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -518,6 +518,8 @@ static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) value = oxygen_read_ac97(chip, 0, priv_idx); if (!(value & 0x8000)) { oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000); + if (chip->model->ac97_switch) + chip->model->ac97_switch(chip, priv_idx, 0x8000); snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->controls[control]->id); } @@ -544,6 +546,8 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, change = newreg != oldreg; if (change) { oxygen_write_ac97(chip, codec, index, newreg); + if (codec == 0 && chip->model->ac97_switch) + chip->model->ac97_switch(chip, index, newreg & 0x8000); if (index == AC97_LINE) { oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, newreg & 0x8000 ? diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 1db4aa5dfad4..b678e2de4adf 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -47,10 +47,10 @@ * GPI 0 <- external power present * * GPIO 0 -> enable output to speakers - * GPIO 1 -> ALT? + * GPIO 1 -> ? * GPIO 2 -> M0 of CS5361 * GPIO 3 -> M1 of CS5361 - * GPIO 8 -> line-in/mic-in/digital-out switch? + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * * CS4398: * @@ -120,7 +120,7 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPI_DX_EXT_POWER 0x01 #define GPIO_DX_OUTPUT_ENABLE 0x0001 #define GPIO_DX_UNKNOWN1 0x0002 -#define GPIO_DX_UNKNOWN2 0x0100 +#define GPIO_DX_INPUT_ROUTE 0x0100 #define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ #define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ @@ -267,7 +267,8 @@ static void xonar_dx_init(struct oxygen *chip) cs4362a_write(chip, 0x01, CS4362A_CPEN); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_DX_UNKNOWN1 | GPIO_DX_UNKNOWN2); + GPIO_DX_UNKNOWN1 | GPIO_DX_INPUT_ROUTE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DX_INPUT_ROUTE); xonar_common_init(chip); @@ -469,6 +470,18 @@ static const struct snd_kcontrol_new alt_switch = { .put = alt_switch_put, }; +static void xonar_dx_ac97_switch(struct oxygen *chip, + unsigned int reg, unsigned int mute) +{ + if (reg == AC97_LINE) { + spin_lock_irq(&chip->reg_lock); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + mute ? GPIO_DX_INPUT_ROUTE : 0, + GPIO_DX_INPUT_ROUTE); + spin_unlock_irq(&chip->reg_lock); + } +} + static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); @@ -572,6 +585,7 @@ static const struct oxygen_model xonar_models[] = { .update_dac_volume = update_cs43xx_volume, .update_dac_mute = update_cs43xx_mute, .gpio_changed = xonar_gpio_changed, + .ac97_switch = xonar_dx_ac97_switch, .model_data_size = sizeof(struct xonar_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | From 1d98c7d4be6ac521e3391025ddffcfe0400c798c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 9 Apr 2008 09:16:33 +0200 Subject: [PATCH 175/250] [ALSA] oxygen: use SPDIF input only if present If the card model does not have a digital input or an AC97 codec, disable the respective interrupt and mixer controls. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 5 ---- sound/pci/oxygen/oxygen_lib.c | 49 +++++++++++++++++++++------------ sound/pci/oxygen/oxygen_mixer.c | 9 ++++++ sound/pci/oxygen/virtuoso.c | 5 ---- 4 files changed, 40 insertions(+), 28 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 1e54a3cd721e..50551ae0b3b4 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -132,11 +132,6 @@ static int hifier_control_filter(struct snd_kcontrol_new *template) template->tlv.p = ak4396_db_scale; } else if (!strcmp(template->name, "Stereo Upmixing")) { return 1; /* stereo only - we don't need upmixing */ - } else if (!strcmp(template->name, - SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK)) || - !strcmp(template->name, - SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) { - return 1; /* no digital input */ } return 0; } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 4f3d9e5fc5db..39e4b7a51740 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -291,20 +291,27 @@ static void oxygen_init(struct oxygen *chip) OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); - oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_SENSE_MASK | - OXYGEN_SPDIF_LOCK_MASK | - OXYGEN_SPDIF_RATE_MASK | - OXYGEN_SPDIF_LOCK_PAR | - OXYGEN_SPDIF_IN_CLOCK_96, - OXYGEN_SPDIF_OUT_ENABLE | - OXYGEN_SPDIF_LOOPBACK | - OXYGEN_SPDIF_SENSE_MASK | - OXYGEN_SPDIF_LOCK_MASK | - OXYGEN_SPDIF_RATE_MASK | - OXYGEN_SPDIF_SENSE_PAR | - OXYGEN_SPDIF_LOCK_PAR | - OXYGEN_SPDIF_IN_CLOCK_MASK); + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_OUT_ENABLE | + OXYGEN_SPDIF_LOOPBACK); + if (chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF) + oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_SENSE_MASK | + OXYGEN_SPDIF_LOCK_MASK | + OXYGEN_SPDIF_RATE_MASK | + OXYGEN_SPDIF_LOCK_PAR | + OXYGEN_SPDIF_IN_CLOCK_96, + OXYGEN_SPDIF_SENSE_MASK | + OXYGEN_SPDIF_LOCK_MASK | + OXYGEN_SPDIF_RATE_MASK | + OXYGEN_SPDIF_SENSE_PAR | + OXYGEN_SPDIF_LOCK_PAR | + OXYGEN_SPDIF_IN_CLOCK_MASK); + else + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_SENSE_MASK | + OXYGEN_SPDIF_LOCK_MASK | + OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK); oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0); @@ -327,9 +334,12 @@ static void oxygen_init(struct oxygen *chip) (2 << OXYGEN_A_MONITOR_ROUTE_2_SHIFT) | (3 << OXYGEN_A_MONITOR_ROUTE_3_SHIFT)); - oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, - OXYGEN_AC97_INT_READ_DONE | - OXYGEN_AC97_INT_WRITE_DONE); + if (chip->has_ac97_0 | chip->has_ac97_1) + oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, + OXYGEN_AC97_INT_READ_DONE | + OXYGEN_AC97_INT_WRITE_DONE); + else + oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0); oxygen_write32(chip, OXYGEN_AC97_OUT_CONFIG, 0); oxygen_write32(chip, OXYGEN_AC97_IN_CONFIG, 0); if (!(chip->has_ac97_0 | chip->has_ac97_1)) @@ -495,7 +505,10 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, oxygen_proc_init(chip); spin_lock_irq(&chip->reg_lock); - chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT | OXYGEN_INT_AC97; + if (chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF) + chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT; + if (chip->has_ac97_0 | chip->has_ac97_1) + chip->interrupt_mask |= OXYGEN_INT_AC97; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index d0bef09a6999..2cb914498a19 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -742,6 +742,9 @@ static const struct snd_kcontrol_new controls[] = { .get = spdif_pcm_get, .put = spdif_pcm_put, }, +}; + +static const struct snd_kcontrol_new spdif_input_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .device = 1, @@ -961,6 +964,12 @@ int oxygen_mixer_init(struct oxygen *chip) err = add_controls(chip, controls, ARRAY_SIZE(controls)); if (err < 0) return err; + if (chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF) { + err = add_controls(chip, spdif_input_controls, + ARRAY_SIZE(spdif_input_controls)); + if (err < 0) + return err; + } for (i = 0; i < ARRAY_SIZE(monitor_controls); ++i) { if (!(chip->model->pcm_dev_cfg & monitor_controls[i].pcm_dev)) continue; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index b678e2de4adf..07d7e9b6afbb 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -506,11 +506,6 @@ static int xonar_dx_control_filter(struct snd_kcontrol_new *template) template->tlv.p = cs4362a_db_scale; } else if (!strncmp(template->name, "CD Capture ", 11)) { return 1; /* no CD input */ - } else if (!strcmp(template->name, - SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK)) || - !strcmp(template->name, - SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) { - return 1; /* no digital input */ } return 0; } From 387fb6a206749e13377ef8847f77d5341c281e7b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 11 Apr 2008 10:24:48 +0200 Subject: [PATCH 176/250] [ALSA] virtuoso: add GPIO 1 mixer control Add a mixer control for switching whatever it is that is connected to GPIO pin 1 on the Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 53 +++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 07d7e9b6afbb..53d7289f21c0 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -470,6 +470,53 @@ static const struct snd_kcontrol_new alt_switch = { .put = alt_switch_put, }; +static int unknown_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item > 1) + info->value.enumerated.item = 1; + sprintf(info->value.enumerated.name, "%u", info->value.enumerated.item); + return 0; +} + +static int unknown_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.enumerated.item[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DX_UNKNOWN1); + return 0; +} + +static int unknown_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 old_reg, new_reg; + + spin_lock_irq(&chip->reg_lock); + old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (value->value.enumerated.item[0]) + new_reg = old_reg | GPIO_DX_UNKNOWN1; + else + new_reg = old_reg & ~GPIO_DX_UNKNOWN1; + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg); + spin_unlock_irq(&chip->reg_lock); + return old_reg != new_reg; +} + +static const struct snd_kcontrol_new unknown_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PanelConfig?", + .info = unknown_info, + .get = unknown_get, + .put = unknown_put, +}; + static void xonar_dx_ac97_switch(struct oxygen *chip, unsigned int reg, unsigned int mute) { @@ -515,6 +562,11 @@ static int xonar_mixer_init(struct oxygen *chip) return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); } +static int xonar_dx_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&unknown_switch, chip)); +} + static const struct oxygen_model xonar_models[] = { [MODEL_D2] = { .shortname = "Xonar D2", @@ -574,6 +626,7 @@ static const struct oxygen_model xonar_models[] = { .owner = THIS_MODULE, .init = xonar_dx_init, .control_filter = xonar_dx_control_filter, + .mixer_init = xonar_dx_mixer_init, .cleanup = xonar_dx_cleanup, .set_dac_params = set_cs43xx_params, .set_adc_params = set_cs53x1_params, From 80060ecc45bd101f3decafed5b7ff0879a188d28 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 11 Apr 2008 10:25:40 +0200 Subject: [PATCH 177/250] [ALSA] virtuoso: initialize two-wire control register On the Xonar DX, initialize all bits of the two-wire control register. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 53d7289f21c0..c7080d6b71c2 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -229,9 +229,10 @@ static void xonar_dx_init(struct oxygen *chip) data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->ext_power_bit = GPI_DX_EXT_POWER; - /* XXX the DACs' datasheets say fast mode is not allowed */ - oxygen_set_bits16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_SPEED_FAST); + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); /* set CPEN (control port mode) and power down */ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); From 4383fae0ec5bc269f9eb4383b223731e3ecd2fe3 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Mon, 14 Apr 2008 12:58:57 +0200 Subject: [PATCH 178/250] [ALSA] hda-codec - PCI quirk for MSI laptop Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5d97c12381d..0f6de1eb6423 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6864,6 +6864,9 @@ static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), @@ -7653,6 +7656,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), From 32f4876e62d5caba712ca76d96b0018dcc0f9601 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 14 Apr 2008 12:59:27 +0200 Subject: [PATCH 179/250] [ALSA] soc - Include register in DAPM debug output When logging register changes in DAPM debug output include the register number. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4c64560493f5..af3326c63504 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -226,7 +226,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(POP_TIME); } - dbg("reg old %x new %x change %d\n", old, new, change); + dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; } From 62cef8212ffa9df3e6c5b358ea2382d90489d590 Mon Sep 17 00:00:00 2001 From: Denys Vlasenko Date: Mon, 14 Apr 2008 13:04:18 +0200 Subject: [PATCH 180/250] [ALSA] sound/pci/rme9652/hdspm.c: stop inlining largish static functions sound/pci/rme9652/hdspm.c has unusually large number of static inline functions - 22. I looked through them and some of them seem to be too big to warrant inlining. This patch removes "inline" from these static functions (regardless of number of callsites - gcc nowadays auto-inlines statics with one callsite). Size difference on 32bit x86: text data bss dec hex filename 20437 2160 516 23113 5a49 linux-2.6-ALLYES/sound/pci/rme9652/hdspm.o 18036 2160 516 20712 50e8 linux-2.6.inline-ALLYES/sound/pci/rme9652/hdspm.o [coding fix by Takashi Iwai ] Signed-off-by: Denys Vlasenko Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 38c931c480d9..ab423bc82342 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -540,7 +540,8 @@ static void hdspm_set_sgbuf(struct hdspm * hdspm, struct snd_sg_buf *sgbuf, static inline int HDSPM_bit2freq(int n) { - static int bit2freq_tab[] = { 0, 32000, 44100, 48000, 64000, 88200, + static const int bit2freq_tab[] = { + 0, 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000 }; if (n < 1 || n > 9) return 0; @@ -582,7 +583,7 @@ static inline int hdspm_read_pb_gain(struct hdspm * hdspm, unsigned int chan, return hdspm->mixer->ch[chan].pb[pb]; } -static inline int hdspm_write_in_gain(struct hdspm * hdspm, unsigned int chan, +static int hdspm_write_in_gain(struct hdspm *hdspm, unsigned int chan, unsigned int in, unsigned short data) { if (chan >= HDSPM_MIXER_CHANNELS || in >= HDSPM_MIXER_CHANNELS) @@ -595,7 +596,7 @@ static inline int hdspm_write_in_gain(struct hdspm * hdspm, unsigned int chan, return 0; } -static inline int hdspm_write_pb_gain(struct hdspm * hdspm, unsigned int chan, +static int hdspm_write_pb_gain(struct hdspm *hdspm, unsigned int chan, unsigned int pb, unsigned short data) { if (chan >= HDSPM_MIXER_CHANNELS || pb >= HDSPM_MIXER_CHANNELS) @@ -621,7 +622,7 @@ static inline void snd_hdspm_enable_out(struct hdspm * hdspm, int i, int v) } /* check if same process is writing and reading */ -static inline int snd_hdspm_use_is_exclusive(struct hdspm * hdspm) +static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm) { unsigned long flags; int ret = 1; @@ -636,7 +637,7 @@ static inline int snd_hdspm_use_is_exclusive(struct hdspm * hdspm) } /* check for external sample rate */ -static inline int hdspm_external_sample_rate(struct hdspm * hdspm) +static int hdspm_external_sample_rate(struct hdspm *hdspm) { if (hdspm->is_aes32) { unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2); @@ -787,7 +788,7 @@ static inline void hdspm_stop_audio(struct hdspm * s) } /* should I silence all or only opened ones ? doit all for first even is 4MB*/ -static inline void hdspm_silence_playback(struct hdspm * hdspm) +static void hdspm_silence_playback(struct hdspm *hdspm) { int i; int n = hdspm->period_bytes; @@ -1057,7 +1058,7 @@ static inline int snd_hdspm_midi_output_possible (struct hdspm *hdspm, int id) return 0; } -static inline void snd_hdspm_flush_midi_input (struct hdspm *hdspm, int id) +static void snd_hdspm_flush_midi_input(struct hdspm *hdspm, int id) { while (snd_hdspm_midi_input_available (hdspm, id)) snd_hdspm_midi_read_byte (hdspm, id); From f24bfa53dab478e1bde2d7fd39d3c1a69dc518f1 Mon Sep 17 00:00:00 2001 From: Andreas Mueller Date: Mon, 14 Apr 2008 13:08:05 +0200 Subject: [PATCH 181/250] [ALSA] es1968: fix jitter on some maestro cards This patch suppresses jitter on several Maestro cards in stereo mode (ALSA of course). The patch is also incorporated in the *BSD drivers where I "ported" it from. Without this patch most of the stereo audio gets out of sync and really distorted (oss-emulation with mplayer at 48000khz worked somehow). Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 67f03264f871..f8f3bb662d17 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1827,6 +1827,23 @@ snd_es1968_pcm(struct es1968 *chip, int device) return 0; } +/* + * suppress jitter on some maestros when playing stereo + */ +static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es) +{ + unsigned int cp1; + unsigned int cp2; + unsigned int diff; + + cp1 = __apu_get_register(chip, 0, 5); + cp2 = __apu_get_register(chip, 1, 5); + diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1); + + if (diff > 1) { + __maestro_write(chip, IDR0_DATA_PORT, cp1); + } +} /* * update pointer @@ -1948,8 +1965,11 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id) struct esschan *es; spin_lock(&chip->substream_lock); list_for_each_entry(es, &chip->substream_list, list) { - if (es->running) + if (es->running) { snd_es1968_update_pcm(chip, es); + if (es->fmt & ESS_FMT_STEREO) + snd_es1968_suppress_jitter(chip, es); + } } spin_unlock(&chip->substream_lock); if (chip->in_measurement) { From 66c9aa6043798197e1760eaf4c5f510d6c69b95a Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Mon, 14 Apr 2008 13:09:33 +0200 Subject: [PATCH 182/250] [ALSA] es1968 - fix coding style in the last patch WARNING: braces {} are not necessary for single statement blocks #40: FILE: sound/pci/es1968.c:1831: + if (diff > 1) { + __maestro_write(chip, IDR0_DATA_PORT, cp1); + } total: 0 errors, 1 warnings, 35 lines checked ./patches/es1968-fix-jitter-on-some-maestro-cards.patch has style problems, please review. If any of these errors are false positives report them to the maintainer, see CHECKPATCH in MAINTAINERS. Please run checkpatch prior to sending patches Cc: Andreas Mueller Tested-by: Rene Herman Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index f8f3bb662d17..138379896061 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1840,9 +1840,8 @@ static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es) cp2 = __apu_get_register(chip, 1, 5); diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1); - if (diff > 1) { + if (diff > 1) __maestro_write(chip, IDR0_DATA_PORT, cp1); - } } /* From eaa9985b4edb5f8008998abdda8b85ddeba05f1f Mon Sep 17 00:00:00 2001 From: Frederik Deweerdt Date: Mon, 14 Apr 2008 13:11:44 +0200 Subject: [PATCH 183/250] [ALSA] hda-codec - Fix unbalanced mutex On Wed, Apr 02, 2008 at 08:19:29AM -0400, Miles Lane wrote: > [ 48.765906] [ BUG: bad unlock balance detected! ] > [ 48.765912] ------------------------------------- > [ 48.765918] pulseaudio/4277 is trying to release lock > (&codec->spdif_mutex) at: > [ 48.765930] [] mutex_unlock+0x8/0xa > [ 48.765945] but there are no more locks to release! > [ 48.765950] > [ 48.765952] other info that might help us debug this: > [ 48.765959] 2 locks held by pulseaudio/4277: > [ 48.765965] #0: (&pcm->open_mutex){--..}, at: [] > snd_pcm_open+0xc1/0x1ba [snd_pcm] > [ 48.766003] #1: (&chip->open_mutex){--..}, at: [] > azx_pcm_open+0x36/0x184 [snd_hda_intel] > [ 48.766057] > [ 48.766059] stack backtrace: > [ 48.766066] Pid: 4277, comm: pulseaudio Not tainted 2.6.25-rc8-mm1 #12 > [ 48.766086] [] print_unlock_inbalance_bug+0xce/0xd8 > [ 48.766107] [] ? save_stack_trace+0x1d/0x3b > [ 48.766130] [] ? __kernel_text_address+0x1b/0x27 > [ 48.766146] [] ? dump_trace+0xcd/0xd9 > [ 48.766160] [] ? save_stack_address+0x0/0x2c > [ 48.766176] [] ? find_usage_backwards+0xa4/0xc3 > [ 48.766193] [] lock_release_non_nested+0x84/0x120 > [ 48.766209] [] ? mutex_unlock+0x8/0xa > [ 48.766222] [] lock_release+0x16a/0x199 > [ 48.766238] [] __mutex_unlock_slowpath+0xa9/0x121 > [ 48.766252] [] mutex_unlock+0x8/0xa > [ 48.766263] [] snd_hda_multi_out_analog_open+0xd3/0xef > [snd_hda_intel] The following patch should fix it. Cc: "Miles Lane" Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 689d177c17bd..a6be6e3e8716 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2665,8 +2665,8 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, if (mout->spdif_maxbps < hinfo->maxbps) hinfo->maxbps = mout->spdif_maxbps; } + mutex_unlock(&codec->spdif_mutex); } - mutex_unlock(&codec->spdif_mutex); return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } From b9d43bcd061956c8144bcb453d07d13236b6ab28 Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Mon, 14 Apr 2008 13:12:47 +0200 Subject: [PATCH 184/250] [ALSA] usb audio: Fix another Dallas quirk Dallas USB speakers are buggy in more than one way. One of configs they offer does not work at all. Signed-off-by: Pavel Machek Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 5c40c031dd57..ff2e09a46133 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2676,12 +2676,23 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) int format; struct audioformat *fp; unsigned char *fmt, *csep; + int num; dev = chip->dev; /* parse the interface's altsettings */ iface = usb_ifnum_to_if(dev, iface_no); - for (i = 0; i < iface->num_altsetting; i++) { + + num = iface->num_altsetting; + + /* + * Dallas DS4201 workaround: It presents 5 altsettings, but the last + * one misses syncpipe, and does not produce any sound. + */ + if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + num = 4; + + for (i = 0; i < num; i++) { alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); /* skip invalid one */ From 2a56f51bcc3650ecff806450f7fdab5edf57618f Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Mon, 14 Apr 2008 13:14:22 +0200 Subject: [PATCH 185/250] [ALSA] usb audio: make quirk handling more readable, and fix commented-out code usb audio contains useful debugging code, protected by #if 0. Unfortunately, it will not compile because variable names changed; fix it. Dallas workaround is formatted in a way where it is not quite obvious what is normal code and what is quirk. Reformat it to make it obvious. Signed-off-by: Pavel Machek Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index ff2e09a46133..26fb5769d951 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1427,8 +1427,8 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; #if 0 - printk("setting done: format = %d, rate = %d, channels = %d\n", - fmt->format, fmt->rate, fmt->channels); + printk("setting done: format = %d, rate = %d..%d, channels = %d\n", + fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); printk(" datapipe = 0x%0x, syncpipe = 0x%0x\n", subs->datapipe, subs->syncpipe); #endif @@ -2468,11 +2468,12 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor } break; case USB_AUDIO_FORMAT_PCM8: - /* Dallas DS4201 workaround */ + pcm_format = SNDRV_PCM_FORMAT_U8; + + /* Dallas DS4201 workaround: it advertises U8 format, but really + supports S8. */ if (chip->usb_id == USB_ID(0x04fa, 0x4201)) pcm_format = SNDRV_PCM_FORMAT_S8; - else - pcm_format = SNDRV_PCM_FORMAT_U8; break; case USB_AUDIO_FORMAT_IEEE_FLOAT: pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; From 07f51a727462696ddea01c7a7750c27796a0e1f3 Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Mon, 14 Apr 2008 13:15:56 +0200 Subject: [PATCH 186/250] [ALSA] sound/usb/usbaudio.c: coding style Putting space between ! and variable is a strange coding style, fix that, also make it fit into 80 columns where that is easy. Signed-off-by: Pavel Machek Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 63 ++++++++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 31 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 26fb5769d951..7b3bcf1916d2 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -64,9 +64,10 @@ MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Vendor ID for this card */ -static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Product ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ +/* Vendor/product IDs for this card */ +static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; +static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int nrpacks = 8; /* max. number of packets per urb */ static int async_unlink = 1; static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/ @@ -687,7 +688,7 @@ static void snd_complete_urb(struct urb *urb) int err = 0; if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - ! subs->running || /* can be stopped during retire callback */ + !subs->running || /* can be stopped during retire callback */ (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { clear_bit(ctx->index, &subs->active_mask); @@ -710,7 +711,7 @@ static void snd_complete_sync_urb(struct urb *urb) int err = 0; if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - ! subs->running || /* can be stopped during retire callback */ + !subs->running || /* can be stopped during retire callback */ (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { clear_bit(ctx->index + 16, &subs->active_mask); @@ -740,7 +741,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s vfree(runtime->dma_area); } runtime->dma_area = vmalloc(size); - if (! runtime->dma_area) + if (!runtime->dma_area) return -ENOMEM; runtime->dma_bytes = size; return 0; @@ -772,12 +773,12 @@ static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sl async = !can_sleep && async_unlink; - if (! async && in_interrupt()) + if (!async && in_interrupt()) return 0; for (i = 0; i < subs->nurbs; i++) { if (test_bit(i, &subs->active_mask)) { - if (! test_and_set_bit(i, &subs->unlink_mask)) { + if (!test_and_set_bit(i, &subs->unlink_mask)) { struct urb *u = subs->dataurb[i].urb; if (async) usb_unlink_urb(u); @@ -789,7 +790,7 @@ static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sl if (subs->syncpipe) { for (i = 0; i < SYNC_URBS; i++) { if (test_bit(i+16, &subs->active_mask)) { - if (! test_and_set_bit(i+16, &subs->unlink_mask)) { + if (!test_and_set_bit(i+16, &subs->unlink_mask)) { struct urb *u = subs->syncurb[i].urb; if (async) usb_unlink_urb(u); @@ -1137,12 +1138,12 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri if (subs->fmt_type == USB_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); - if (! u->urb) + if (!u->urb) goto out_of_memory; u->urb->transfer_buffer = usb_buffer_alloc(subs->dev, u->buffer_size, GFP_KERNEL, &u->urb->transfer_dma); - if (! u->urb->transfer_buffer) + if (!u->urb->transfer_buffer) goto out_of_memory; u->urb->pipe = subs->datapipe; u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; @@ -1155,7 +1156,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri /* allocate and initialize sync urbs */ subs->syncbuf = usb_buffer_alloc(subs->dev, SYNC_URBS * 4, GFP_KERNEL, &subs->sync_dma); - if (! subs->syncbuf) + if (!subs->syncbuf) goto out_of_memory; for (i = 0; i < SYNC_URBS; i++) { struct snd_urb_ctx *u = &subs->syncurb[i]; @@ -1163,7 +1164,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri u->subs = subs; u->packets = 1; u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (! u->urb) + if (!u->urb) goto out_of_memory; u->urb->transfer_buffer = subs->syncbuf + i * 4; u->urb->transfer_dma = subs->sync_dma + i * 4; @@ -1463,7 +1464,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, rate = params_rate(hw_params); channels = params_channels(hw_params); fmt = find_format(subs, format, rate, channels); - if (! fmt) { + if (!fmt) { snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n", format, rate, channels); return -EINVAL; @@ -1584,7 +1585,7 @@ static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audiof struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* check the format */ - if (! snd_mask_test(fmts, fp->format)) { + if (!snd_mask_test(fmts, fp->format)) { hwc_debug(" > check: no supported format %d\n", fp->format); return 0; } @@ -1620,7 +1621,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (! hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(params, fp)) continue; if (changed++) { if (rmin > fp->rate_min) @@ -1633,7 +1634,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, } } - if (! changed) { + if (!changed) { hwc_debug(" --> get empty\n"); it->empty = 1; return -EINVAL; @@ -1674,7 +1675,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (! hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(params, fp)) continue; if (changed++) { if (rmin > fp->channels) @@ -1687,7 +1688,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, } } - if (! changed) { + if (!changed) { hwc_debug(" --> get empty\n"); it->empty = 1; return -EINVAL; @@ -1727,7 +1728,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (! hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(params, fp)) continue; fbits |= (1ULL << fp->format); } @@ -1736,7 +1737,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, oldbits[1] = fmt->bits[1]; fmt->bits[0] &= (u32)fbits; fmt->bits[1] &= (u32)(fbits >> 32); - if (! fmt->bits[0] && ! fmt->bits[1]) { + if (!fmt->bits[0] && !fmt->bits[1]) { hwc_debug(" --> get empty\n"); return -EINVAL; } @@ -2227,7 +2228,7 @@ static void proc_pcm_format_add(struct snd_usb_stream *stream) struct snd_card *card = stream->chip->card; sprintf(name, "stream%d", stream->pcm_index); - if (! snd_card_proc_new(card, name, &entry)) + if (!snd_card_proc_new(card, name, &entry)) snd_info_set_text_ops(entry, stream, proc_pcm_format_read); } @@ -2283,7 +2284,7 @@ static void free_substream(struct snd_usb_substream *subs) { struct list_head *p, *n; - if (! subs->num_formats) + if (!subs->num_formats) return; /* not initialized */ list_for_each_safe(p, n, &subs->fmt_list) { struct audioformat *fp = list_entry(p, struct audioformat, list); @@ -2333,7 +2334,7 @@ static int add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct aud if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (! subs->endpoint) + if (!subs->endpoint) continue; if (subs->endpoint == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); @@ -2359,7 +2360,7 @@ static int add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct aud /* create a new pcm */ as = kzalloc(sizeof(*as), GFP_KERNEL); - if (! as) + if (!as) return -ENOMEM; as->pcm_index = chip->pcm_devs; as->chip = chip; @@ -3392,14 +3393,14 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, static void proc_audio_usbbus_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_usb_audio *chip = entry->private_data; - if (! chip->shutdown) + if (!chip->shutdown) snd_iprintf(buffer, "%03d/%03d\n", chip->dev->bus->busnum, chip->dev->devnum); } static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_usb_audio *chip = entry->private_data; - if (! chip->shutdown) + if (!chip->shutdown) snd_iprintf(buffer, "%04x:%04x\n", USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); @@ -3408,9 +3409,9 @@ static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_ static void snd_usb_audio_create_proc(struct snd_usb_audio *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "usbbus", &entry)) + if (!snd_card_proc_new(chip->card, "usbbus", &entry)) snd_info_set_text_ops(entry, chip, proc_audio_usbbus_read); - if (! snd_card_proc_new(chip->card, "usbid", &entry)) + if (!snd_card_proc_new(chip->card, "usbid", &entry)) snd_info_set_text_ops(entry, chip, proc_audio_usbid_read); } @@ -3617,8 +3618,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev, snd_card_set_dev(chip->card, &intf->dev); break; } - if (! chip) { - snd_printk(KERN_ERR "no available usb audio device\n"); + if (!chip) { + printk(KERN_ERR "no available usb audio device\n"); goto __error; } } From 64654c2f9e7875a982a9c3b4456ed11ad811ec61 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Mon, 14 Apr 2008 13:26:21 +0200 Subject: [PATCH 187/250] [ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880 To mute the output of Pin widget 15 in ALC880, we should use the HDA_OUTPUT. However, current code looks like : snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); It may be a misspelling. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0f6de1eb6423..3aa182a16753 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1894,7 +1894,7 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) From 5b030389e4ba72cc4e9ad37a5770b32f0353564d Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Mon, 14 Apr 2008 13:26:53 +0200 Subject: [PATCH 188/250] [ALSA] hda - PCI quirk for laptop LG which use CMI9880 Please refer to [0003874] on the alsa mantis. This patch added the pci quirk. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cmedia.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 8d142851aaca..c73ce074a6ea 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = { static struct snd_pci_quirk cmi9880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG), + SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG), {} /* terminator */ }; From 06a9c30cdda43ca82d7f22c8ebeb93e691f85b5f Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 14 Apr 2008 13:31:45 +0200 Subject: [PATCH 189/250] [ALSA] hda - Fujitsu Lifebook PC speaker signal The legacy PC speaker signal was not routed to outputs. The codec is not prevented from powering down in this patch, although I suppose one could argue that perhaps it should be. Let me know if anyone feels strongly one way or the other. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3aa182a16753..830d6626e10e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8784,6 +8784,8 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), From 780c8be4ab6c7baf5cdfa1102f56480acb8a2479 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 14 Apr 2008 13:32:27 +0200 Subject: [PATCH 190/250] [ALSA] hda: Correct SPDIF out default config Several laptops have have the SPDIF out defined as 'Digital other out' when it should be 'SPDIF out' in the default config. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 69cd3b23f5a7..d79580c29862 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3778,6 +3778,13 @@ static int patch_stac927x(struct hda_codec *codec) spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + switch (codec->subsystem_id) { + case 0x10280209: + case 0x1028022e: + /* correct the device field to SPDIF out */ + stac92xx_set_config_reg(codec, 0x21, 0x01442070); + break; + }; /* configure the analog microphone on some laptops */ stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ From 0fc9dec46fae19da9899c580a6b870202103f8bb Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 14 Apr 2008 13:32:54 +0200 Subject: [PATCH 191/250] [ALSA] hda: EAPD power management Power management support for EAPD enabled laptops, when headphones are sensed it pulls the EAPD GPIO line low to power it down. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d79580c29862..e1d07ab5cd15 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -129,6 +129,7 @@ struct sigmatel_spec { unsigned int hp_detect: 1; /* gpio lines */ + unsigned int eapd_mask; unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; @@ -3183,6 +3184,10 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) for (i = 0; i < cfg->speaker_outs; i++) stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) @@ -3191,6 +3196,10 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data | + spec->eapd_mask); } } @@ -3478,7 +3487,7 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; switch (spec->board_config) { @@ -3584,7 +3593,10 @@ again: spec->aloopback_shift = 0; /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0x1; + spec->gpio_mask = 0x01; + spec->gpio_dir = 0x01; + spec->gpio_mask = 0x01; + spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3770,7 +3782,7 @@ static int patch_stac927x(struct hda_codec *codec) case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x01; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3794,7 +3806,7 @@ static int patch_stac927x(struct hda_codec *codec) /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x04; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; spec->gpio_data = 0x04; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; @@ -3806,7 +3818,7 @@ static int patch_stac927x(struct hda_codec *codec) break; default: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3910,6 +3922,7 @@ static int patch_stac9205(struct hda_codec *codec) (AC_USRSP_EN | STAC_HP_EVENT)); spec->gpio_dir = 0x0b; + spec->eapd_mask = 0x01; spec->gpio_mask = 0x1b; spec->gpio_mute = 0x10; /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, @@ -3919,7 +3932,7 @@ static int patch_stac9205(struct hda_codec *codec) break; default: /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; break; } From 8b45a209935c4b79905182608922736ba0e5579e Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Mon, 14 Apr 2008 13:33:36 +0200 Subject: [PATCH 192/250] [ALSA] sound: fix platform driver hotplug/coldplug Since 43cc71eed1250755986da4c0f9898f9a635cb3bf, the platform modalias is prefixed with "platform:". Add MODULE_ALIAS() to the hotpluggable sound platform drivers, to re-enable auto loading. [dbrownell@users.sourceforge.net: more drivers, registration fixes] Signed-off-by: Kay Sievers Signed-off-by: David Brownell Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 2 ++ sound/drivers/ml403-ac97cr.c | 4 ++++ sound/soc/soc-core.c | 2 ++ 3 files changed, 8 insertions(+) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 490729799e59..71fbf8d7ee82 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -424,6 +424,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .resume = pxa2xx_ac97_resume, .driver = { .name = "pxa2xx-ac97", + .owner = THIS_MODULE, }, }; @@ -443,3 +444,4 @@ module_exit(pxa2xx_ac97_exit); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index b654007331de..ecdbeb6d3603 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1328,11 +1328,15 @@ static int snd_ml403_ac97cr_remove(struct platform_device *pfdev) return 0; } +/* work with hotplug and coldplug */ +MODULE_ALIAS("platform:" SND_ML403_AC97CR_DRIVER); + static struct platform_driver snd_ml403_ac97cr_driver = { .probe = snd_ml403_ac97cr_probe, .remove = snd_ml403_ac97cr_remove, .driver = { .name = SND_ML403_AC97CR_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9eb5479787c1..e148db940cfc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -839,6 +839,7 @@ static int soc_remove(struct platform_device *pdev) static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", + .owner = THIS_MODULE, }, .probe = soc_probe, .remove = soc_remove, @@ -1601,3 +1602,4 @@ module_exit(snd_soc_exit); MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:soc-audio"); From eb5a66216658a1c879ab05dcdc65ce7005c4780b Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Mon, 14 Apr 2008 13:46:28 +0200 Subject: [PATCH 193/250] [ALSA] hda-intel: Add Quanta IL1 ALC267 model This adds support for Quanta IL1 mini-notebook to alsa, defining a new model for it. It comes with an ALC267 codec chip. Some notes about this model: * In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common amp mute, to avoid conflict with mixer switch (mixer and automute use the same nid). * The only connected capture sources in the hardware are the internal mic and external mic jack. So instead of using an input source selector like on other ALC268 models, the mic automute automatically switch between captures. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 3 +- sound/pci/hda/patch_realtek.c | 76 +++++++++++++++++++ 2 files changed, 78 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 08256fcb2743..3413644dff1e 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -829,7 +829,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) - ALC268 + ALC267/268 + quanta-il1 Quanta IL1 mini-notebook 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 830d6626e10e..181db2177a1d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -105,6 +105,7 @@ enum { /* ALC268 models */ enum { + ALC267_QUANTA_IL1, ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, @@ -9902,6 +9903,64 @@ static void alc268_dell_unsol_event(struct hda_codec *codec, #define alc268_dell_init_hook alc268_dell_automute +static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + { } +}; + +static struct hda_verb alc267_quanta_il1_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc267_quanta_il1_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); +} + +static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, + present ? 0x00 : 0x01); +} + +static void alc267_quanta_il1_automute(struct hda_codec *codec) +{ + alc267_quanta_il1_hp_automute(codec); + alc267_quanta_il1_mic_automute(codec); +} + +static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc267_quanta_il1_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc267_quanta_il1_mic_automute(codec); + break; + } +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -10324,6 +10383,7 @@ static void alc268_auto_init(struct hda_codec *codec) * configuration and preset */ static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC267_QUANTA_IL1] = "quanta-il1", [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", @@ -10346,11 +10406,27 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), + SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), {} }; static struct alc_config_preset alc268_presets[] = { + [ALC267_QUANTA_IL1] = { + .mixers = { alc267_quanta_il1_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc267_quanta_il1_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc267_quanta_il1_unsol_event, + .init_hook = alc267_quanta_il1_automute, + }, [ALC268_3ST] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, alc268_beep_mixer }, From a295e09e89d227506ae6c0a58e1cb6359c0cda1c Mon Sep 17 00:00:00 2001 From: Nick Andrew Date: Mon, 14 Apr 2008 15:22:11 +0200 Subject: [PATCH 194/250] [ALSA] sound: this amplifier only goes up to 7 sound: kernel log levels are 0-7 Kernel log levels are 0-7, not 0-9. Signed-off-by: Nick Andrew Signed-off-by: Takashi Iwai --- sound/core/misc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/misc.c b/sound/core/misc.c index 102d1c36cf26..38524f615d94 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -39,7 +39,7 @@ void snd_verbose_printk(const char *file, int line, const char *format, ...) { va_list args; - if (format[0] == '<' && format[1] >= '0' && format[1] <= '9' && format[2] == '>') { + if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') { char tmp[] = "<0>"; tmp[1] = format[1]; printk("%sALSA %s:%d: ", tmp, file, line); @@ -60,7 +60,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) { va_list args; - if (format[0] == '<' && format[1] >= '0' && format[1] <= '9' && format[2] == '>') { + if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') { char tmp[] = "<0>"; tmp[1] = format[1]; printk("%sALSA %s:%d: ", tmp, file, line); From 87b57fe2d3fb1ce33671b944db9a4cbe0cd065ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 14 Apr 2008 15:27:30 +0200 Subject: [PATCH 195/250] [ALSA] wm9713: Don't control touch screen power on suspend Leave the power bit for the touch screen alone when suspending the WM9713 so that the touch screen driver can handle it. This allows the touch screen to be used as a wakeup source when the system is suspended. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9713.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e3174c4d980d..1f241161445c 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1132,8 +1132,17 @@ static int wm9713_soc_suspend(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; + u16 reg; + + /* Disable everything except touchpanel - that will be handled + * by the touch driver and left disabled if touch is not in + * use. */ + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff); + ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + ac97_write(codec, AC97_POWERDOWN, 0x6f00); + ac97_write(codec, AC97_POWERDOWN, 0xffff); - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold); return 0; } From f57ab97e767d293132a29a43ca3ecb0f73f1d5bb Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 14 Apr 2008 15:28:19 +0200 Subject: [PATCH 196/250] [ALSA] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e6a730b0dd29..630684f4a0bc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -661,42 +661,52 @@ struct aic3x_rate_divs { static const struct aic3x_rate_divs aic3x_divs[] = { /* 8k */ {12000000, 8000, 48000, 0xa, 16, 3840}, + {19200000, 8000, 48000, 0xa, 10, 2400}, {22579200, 8000, 48000, 0xa, 8, 7075}, {33868800, 8000, 48000, 0xa, 5, 8049}, /* 11.025k */ {12000000, 11025, 44100, 0x6, 15, 528}, + {19200000, 11025, 44100, 0x6, 9, 4080}, {22579200, 11025, 44100, 0x6, 8, 0}, {33868800, 11025, 44100, 0x6, 5, 3333}, /* 16k */ {12000000, 16000, 48000, 0x4, 16, 3840}, + {19200000, 16000, 48000, 0x4, 10, 2400}, {22579200, 16000, 48000, 0x4, 8, 7075}, {33868800, 16000, 48000, 0x4, 5, 8049}, /* 22.05k */ {12000000, 22050, 44100, 0x2, 15, 528}, + {19200000, 22050, 44100, 0x2, 9, 4080}, {22579200, 22050, 44100, 0x2, 8, 0}, {33868800, 22050, 44100, 0x2, 5, 3333}, /* 32k */ {12000000, 32000, 48000, 0x1, 16, 3840}, + {19200000, 32000, 48000, 0x1, 10, 2400}, {22579200, 32000, 48000, 0x1, 8, 7075}, {33868800, 32000, 48000, 0x1, 5, 8049}, /* 44.1k */ {12000000, 44100, 44100, 0x0, 15, 528}, + {19200000, 44100, 44100, 0x0, 9, 4080}, {22579200, 44100, 44100, 0x0, 8, 0}, {33868800, 44100, 44100, 0x0, 5, 3333}, /* 48k */ {12000000, 48000, 48000, 0x0, 16, 3840}, + {19200000, 48000, 48000, 0x0, 10, 2400}, {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ {12000000, 64000, 96000, 0x1, 16, 3840}, + {19200000, 64000, 96000, 0x1, 10, 2400}, {22579200, 64000, 96000, 0x1, 8, 7075}, {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {12000000, 88200, 88200, 0x0, 15, 528}, + {19200000, 88200, 88200, 0x0, 9, 4080}, {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, /* 96k */ {12000000, 96000, 96000, 0x0, 16, 3840}, + {19200000, 96000, 96000, 0x0, 10, 2400}, {22579200, 96000, 96000, 0x0, 8, 7075}, {33868800, 96000, 96000, 0x0, 5, 8049}, }; @@ -818,6 +828,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, switch (freq) { case 12000000: + case 19200000: case 22579200: case 33868800: aic3x->sysclk = freq; From 8d048841e822f745187246a036d03f2793739b7f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 14 Apr 2008 15:39:14 +0200 Subject: [PATCH 197/250] [ALSA] snd_usb_caiaq: fix potential lockups locking This patch fixes potential lockups in snd_usb_caiaq by refining the locking mechanims and by using usb_kill_urb() in favor to usb_unlink_urb(). Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-audio.c | 73 ++++++++++++++-------------------- sound/usb/caiaq/caiaq-device.c | 4 +- 2 files changed, 32 insertions(+), 45 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 9cc4cd8283f9..1aa927942cc6 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -1,5 +1,5 @@ /* - * Copyright (c) 2006,2007 Daniel Mack, Karsten Wiese + * Copyright (c) 2006-2008 Daniel Mack, Karsten Wiese * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -77,10 +77,15 @@ static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { + unsigned long flags; + spin_lock_irqsave(&dev->spinlock, flags); + if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->sub_playback[sub->number] = NULL; else dev->sub_capture[sub->number] = NULL; + + spin_unlock_irqrestore(&dev->spinlock, flags); } static int @@ -97,13 +102,13 @@ static int stream_start(struct snd_usb_caiaqdev *dev) { int i, ret; - debug("stream_start(%p)\n", dev); - spin_lock_irq(&dev->spinlock); - if (dev->streaming) { - spin_unlock_irq(&dev->spinlock); - return -EINVAL; - } + debug("%s(%p)\n", __func__, dev); + if (dev->streaming) + return -EINVAL; + + memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); + memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); dev->input_panic = 0; dev->output_panic = 0; dev->first_packet = 1; @@ -112,37 +117,35 @@ static int stream_start(struct snd_usb_caiaqdev *dev) for (i = 0; i < N_URBS; i++) { ret = usb_submit_urb(dev->data_urbs_in[i], GFP_ATOMIC); if (ret) { - log("unable to trigger initial read #%d! (ret = %d)\n", - i, ret); + log("unable to trigger read #%d! (ret %d)\n", i, ret); dev->streaming = 0; - spin_unlock_irq(&dev->spinlock); return -EPIPE; } } - spin_unlock_irq(&dev->spinlock); return 0; } static void stream_stop(struct snd_usb_caiaqdev *dev) { int i; - - debug("stream_stop(%p)\n", dev); + + debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; dev->streaming = 0; + for (i = 0; i < N_URBS; i++) { - usb_unlink_urb(dev->data_urbs_in[i]); - usb_unlink_urb(dev->data_urbs_out[i]); + usb_kill_urb(dev->data_urbs_in[i]); + usb_kill_urb(dev->data_urbs_out[i]); } } static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream); - debug("snd_usb_caiaq_substream_open(%p)\n", substream); + debug("%s(%p)\n", __func__, substream); substream->runtime->hw = dev->pcm_info; snd_pcm_limit_hw_rates(substream->runtime); return 0; @@ -152,7 +155,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream); - debug("snd_usb_caiaq_substream_close(%p)\n", substream); + debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { /* when the last client has stopped streaming, @@ -160,24 +163,22 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) stream_stop(dev); dev->pcm_info.rates = dev->samplerates; } - + return 0; } static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - debug("snd_usb_caiaq_pcm_hw_params(%p)\n", sub); + debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); - debug("snd_usb_caiaq_pcm_hw_free(%p)\n", sub); - spin_lock_irq(&dev->spinlock); + debug("%s(%p)\n", __func__, sub); deactivate_substream(dev, sub); - spin_unlock_irq(&dev->spinlock); return snd_pcm_lib_free_pages(sub); } @@ -196,7 +197,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - debug("snd_usb_caiaq_pcm_prepare(%p)\n", substream); + debug("%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -247,15 +248,11 @@ static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - spin_lock(&dev->spinlock); activate_substream(dev, sub); - spin_unlock(&dev->spinlock); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - spin_lock(&dev->spinlock); deactivate_substream(dev, sub); - spin_unlock(&dev->spinlock); break; default: return -EINVAL; @@ -328,8 +325,6 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, if (all_substreams_zero(dev->sub_capture)) return; - spin_lock(&dev->spinlock); - for (i = 0; i < iso->actual_length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_capture[stream]; @@ -345,8 +340,6 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, } } } - - spin_unlock(&dev->spinlock); } static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, @@ -358,8 +351,6 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub; int stream, i; - spin_lock(&dev->spinlock); - for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { for (stream = 0; @@ -393,8 +384,6 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, } } } - - spin_unlock(&dev->spinlock); } static void read_in_urb(struct snd_usb_caiaqdev *dev, @@ -418,8 +407,6 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); } - - check_for_elapsed_periods(dev, dev->sub_capture); } static void fill_out_urb(struct snd_usb_caiaqdev *dev, @@ -429,8 +416,6 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - - spin_lock(&dev->spinlock); for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { @@ -456,9 +441,6 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); } - - spin_unlock(&dev->spinlock); - check_for_elapsed_periods(dev, dev->sub_playback); } static void read_completed(struct urb *urb) @@ -472,6 +454,7 @@ static void read_completed(struct urb *urb) return; dev = info->dev; + if (!dev->streaming) return; @@ -489,8 +472,12 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; if (len > 0) { + spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); read_in_urb(dev, urb, &urb->iso_frame_desc[frame]); + spin_unlock(&dev->spinlock); + check_for_elapsed_periods(dev, dev->sub_playback); + check_for_elapsed_periods(dev, dev->sub_capture); send_it = 1; } @@ -696,7 +683,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *dev) { - debug("snd_usb_caiaq_audio_free (%p)\n", dev); + debug("%s(%p)\n", __func__, dev); stream_stop(dev); free_urbs(dev->data_urbs_in); free_urbs(dev->data_urbs_out); diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 7c44a2c7f963..73c08b40cc5f 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.2"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.4"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -456,7 +456,7 @@ static void snd_disconnect(struct usb_interface *intf) struct snd_usb_caiaqdev *dev; struct snd_card *card = dev_get_drvdata(&intf->dev); - debug("snd_disconnect(%p)\n", intf); + debug("%s(%p)\n", __func__, intf); if (!card) return; From 6849d49c48718def95cf1b74154b9b0aee617c7e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 14 Apr 2008 15:39:47 +0200 Subject: [PATCH 198/250] [ALSA] snd_usb_caiaq: correct input channel order This patch corrects the input channel order of hardware supported by snd_usb_caiaq. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-audio.c | 2 +- sound/usb/caiaq/caiaq-device.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 1aa927942cc6..ebf32818a16d 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -202,7 +202,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; else - dev->audio_in_buf_pos[index] = 0; + dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; if (dev->streaming) return 0; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 73c08b40cc5f..379e6082756f 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.4"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.5"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," From 6e9fc6bd5db34a6580e1917bd0fea4b0754c7de8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 14 Apr 2008 15:40:31 +0200 Subject: [PATCH 199/250] [ALSA] snd_usb_caiaq: make high sample rates work with A8DJ This patch for snd_usb_caiaq makes sample rates higher dann 48KHz work with devices which have more than 2 stereo input/output pairs. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-audio.c | 8 ++++++-- sound/usb/caiaq/caiaq-device.c | 2 +- 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index ebf32818a16d..24970a5c888f 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -39,7 +39,8 @@ #define BYTES_PER_SAMPLE 3 #define BYTES_PER_SAMPLE_USB 4 #define MAX_BUFFER_SIZE (128*1024) - +#define MAX_ENDPOINT_SIZE 512 + #define ENDPOINT_CAPTURE 2 #define ENDPOINT_PLAYBACK 6 @@ -221,7 +222,10 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; - + + if (bpp > MAX_ENDPOINT_SIZE) + bpp = MAX_ENDPOINT_SIZE; + ret = snd_usb_caiaq_set_audio_params(dev, runtime->rate, runtime->sample_bits, bpp); if (ret) diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 379e6082756f..e97d8b2ac16a 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.5"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.6"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," From a8bb1bad9b16ab91de6568ac9356b8f705f7272b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 15 Apr 2008 08:57:31 +0200 Subject: [PATCH 200/250] [ALSA] virtuoso: fix DX front panel I/O Fix the GPIO 1 mixer control to enable I/O through the front panel connector of the Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 51 +++++++++++++++---------------------- 1 file changed, 20 insertions(+), 31 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index c7080d6b71c2..6f5c2531abd0 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -47,7 +47,7 @@ * GPI 0 <- external power present * * GPIO 0 -> enable output to speakers - * GPIO 1 -> ? + * GPIO 1 -> enable front panel I/O * GPIO 2 -> M0 of CS5361 * GPIO 3 -> M1 of CS5361 * GPIO 8 -> route input jack to line-in (0) or mic-in (1) @@ -119,7 +119,7 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPI_DX_EXT_POWER 0x01 #define GPIO_DX_OUTPUT_ENABLE 0x0001 -#define GPIO_DX_UNKNOWN1 0x0002 +#define GPIO_DX_FRONT_PANEL 0x0002 #define GPIO_DX_INPUT_ROUTE 0x0100 #define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ @@ -268,8 +268,9 @@ static void xonar_dx_init(struct oxygen *chip) cs4362a_write(chip, 0x01, CS4362A_CPEN); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_DX_UNKNOWN1 | GPIO_DX_INPUT_ROUTE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DX_INPUT_ROUTE); + GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); xonar_common_init(chip); @@ -471,51 +472,39 @@ static const struct snd_kcontrol_new alt_switch = { .put = alt_switch_put, }; -static int unknown_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item > 1) - info->value.enumerated.item = 1; - sprintf(info->value.enumerated.name, "%u", info->value.enumerated.item); - return 0; -} - -static int unknown_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static int front_panel_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; - value->value.enumerated.item[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DX_UNKNOWN1); + value->value.integer.value[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DX_FRONT_PANEL); return 0; } -static int unknown_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static int front_panel_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; u16 old_reg, new_reg; spin_lock_irq(&chip->reg_lock); old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA); - if (value->value.enumerated.item[0]) - new_reg = old_reg | GPIO_DX_UNKNOWN1; + if (value->value.integer.value[0]) + new_reg = old_reg | GPIO_DX_FRONT_PANEL; else - new_reg = old_reg & ~GPIO_DX_UNKNOWN1; + new_reg = old_reg & ~GPIO_DX_FRONT_PANEL; oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg); spin_unlock_irq(&chip->reg_lock); return old_reg != new_reg; } -static const struct snd_kcontrol_new unknown_switch = { +static const struct snd_kcontrol_new front_panel_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PanelConfig?", - .info = unknown_info, - .get = unknown_get, - .put = unknown_put, + .name = "Front Panel Switch", + .info = snd_ctl_boolean_mono_info, + .get = front_panel_get, + .put = front_panel_put, }; static void xonar_dx_ac97_switch(struct oxygen *chip, @@ -565,7 +554,7 @@ static int xonar_mixer_init(struct oxygen *chip) static int xonar_dx_mixer_init(struct oxygen *chip) { - return snd_ctl_add(chip->card, snd_ctl_new1(&unknown_switch, chip)); + return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); } static const struct oxygen_model xonar_models[] = { From ad5fada51d33b408ad3d2d0848ef6744b5daf06f Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Mon, 14 Apr 2008 18:31:35 +0200 Subject: [PATCH 201/250] [ALSA] sound/core.h: evil #ifdefs snd_minor_info_oss_* is an function returning int _or_ comment, depending on config parameters. That is truly evil, fix it. Signed-off-by: Pavel Machek Signed-off-by: Takashi Iwai --- include/sound/core.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 356659de6b85..695ee53488a3 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -277,8 +277,8 @@ int snd_minor_info_done(void); int snd_minor_info_oss_init(void); int snd_minor_info_oss_done(void); #else -#define snd_minor_info_oss_init() /*NOP*/ -#define snd_minor_info_oss_done() /*NOP*/ +static inline int snd_minor_info_oss_init(void) { return 0; } +static inline int snd_minor_info_oss_done(void) { return 0; } #endif /* memory.c */ From 3adb8abc70aaf5c071f27576069c8b01783cca83 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Apr 2008 18:46:42 +0200 Subject: [PATCH 202/250] [ALSA] hda - Add support of AD1989A/AD1989B Added the support of AD1989A and AD1989B codecs. These codecs can have multiple SPDIF devices, but currently we handle only one SPDIF. If any real devices with two SPDIF interfaces (likely one for SPDIF and one for HDMI), we'll fix this rightly. Otherwise, these codecs are pretty similar with AD1988. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 2 +- sound/pci/hda/patch_analog.c | 28 +++++++++++++++++-- 2 files changed, 26 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 3413644dff1e..fd4c32a031c9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -956,7 +956,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) ultra 2-channel with EAPD (Samsung Ultra tablet PC) - AD1988 + AD1988/AD1988B/AD1989A/AD1989B 6stack 6-jack 6stack-dig ditto with SPDIF 3stack 3-jack diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2befeebd909e..f486eb16a386 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2142,6 +2142,10 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { { } /* end */ }; +static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { } /* end */ +}; /* * initialization verbs @@ -2242,6 +2246,13 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { { } }; +/* AD1989 has no ADC -> SPDIF route */ +static struct hda_verb ad1989_spdif_init_verbs[] = { + /* SPDIF out pin */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } +}; + /* * verbs for 3stack (+dig) */ @@ -2949,10 +2960,19 @@ static int patch_ad1988(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = ad1988_capture_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs; if (spec->multiout.dig_out_nid) { - spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1988_spdif_init_verbs; + if (codec->vendor_id >= 0x11d4989a) { + spec->mixers[spec->num_mixers++] = + ad1989_spdif_out_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1989_spdif_init_verbs; + } else { + spec->mixers[spec->num_mixers++] = + ad1988_spdif_out_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1988_spdif_init_verbs; + } } - if (spec->dig_in_nid) + if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; codec->patch_ops = ad198x_patch_ops; @@ -4184,5 +4204,7 @@ struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, + { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 }, + { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 }, {} /* terminator */ }; From 0c0e6daf14183fb1cd0dea054ecf81165abbdc83 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Apr 2008 09:12:27 +0200 Subject: [PATCH 203/250] [ALSA] hifier: remove empty hifier_mixer_init() The empty hifier_mixer_init() function is useless; remove it. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 50551ae0b3b4..b67888f6745b 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -136,11 +136,6 @@ static int hifier_control_filter(struct snd_kcontrol_new *template) return 0; } -static int hifier_mixer_init(struct oxygen *chip) -{ - return 0; -} - static const struct oxygen_model model_hifier = { .shortname = "C-Media CMI8787", .longname = "C-Media Oxygen HD Audio", @@ -148,7 +143,6 @@ static const struct oxygen_model model_hifier = { .owner = THIS_MODULE, .init = hifier_init, .control_filter = hifier_control_filter, - .mixer_init = hifier_mixer_init, .cleanup = hifier_cleanup, .set_dac_params = set_ak4396_params, .set_adc_params = set_cs5340_params, From 193e813814775b1b1574515fc6f11e61b29a54f7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Apr 2008 09:13:36 +0200 Subject: [PATCH 204/250] [ALSA] oxygen: generalize handling of DAC volume limits Add fields for the DAC volume limits to the module structure so that model drivers do not need to install their own control info handlers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 2 ++ sound/pci/oxygen/oxygen.c | 4 ++++ sound/pci/oxygen/oxygen.h | 2 ++ sound/pci/oxygen/oxygen_lib.c | 2 +- sound/pci/oxygen/oxygen_mixer.c | 4 ++-- sound/pci/oxygen/virtuoso.c | 31 ++++++------------------------- 6 files changed, 17 insertions(+), 28 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index b67888f6745b..bf39c72a130c 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -153,6 +153,8 @@ static const struct oxygen_model model_hifier = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_1, .dac_channels = 2, + .dac_volume_min = 0, + .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index b3b7771b54c0..718169792c3b 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -278,6 +278,8 @@ static const struct oxygen_model model_generic = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1, .dac_channels = 8, + .dac_volume_min = 0, + .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -303,6 +305,8 @@ static const struct oxygen_model model_meridian = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1, .dac_channels = 8, + .dac_volume_min = 0, + .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 7efbf54bc4ec..36f2f81fefc5 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -103,6 +103,8 @@ struct oxygen_model { size_t model_data_size; unsigned int pcm_dev_cfg; u8 dac_channels; + u8 dac_volume_min; + u8 dac_volume_max; u8 misc_flags; u8 function_flags; u16 dac_i2s_format; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 39e4b7a51740..b7aa9fcb6b81 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -221,7 +221,7 @@ static void oxygen_init(struct oxygen *chip) chip->dac_routing = 1; for (i = 0; i < 8; ++i) - chip->dac_volume[i] = 0xff; + chip->dac_volume[i] = chip->model->dac_volume_max; chip->spdif_playback_enable = 1; chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL | (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 2cb914498a19..056581ecd915 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -32,8 +32,8 @@ static int dac_volume_info(struct snd_kcontrol *ctl, info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; info->count = chip->model->dac_channels; - info->value.integer.min = 0; - info->value.integer.max = 0xff; + info->value.integer.min = chip->model->dac_volume_min; + info->value.integer.max = chip->model->dac_volume_max; return 0; } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6f5c2531abd0..37f53a8c5888 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -219,10 +219,7 @@ static void xonar_d2x_init(struct oxygen *chip) static void xonar_dx_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; - unsigned int i; - for (i = 0; i < 8; ++i) - chip->dac_volume[i] = 127; data->anti_pop_delay = 800; data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; data->ext_power_reg = OXYGEN_GPI_DATA; @@ -414,26 +411,6 @@ static void xonar_gpio_changed(struct oxygen *chip) } } -static int pcm1796_volume_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = 8; - info->value.integer.min = 0x0f; - info->value.integer.max = 0xff; - return 0; -} - -static int cs4362a_volume_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = 8; - info->value.integer.min = 0; - info->value.integer.max = 127; - return 0; -} - static int alt_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -526,7 +503,6 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { if (!strcmp(template->name, "Master Playback Volume")) { template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->info = pcm1796_volume_info; template->tlv.p = pcm1796_db_scale; } else if (!strncmp(template->name, "CD Capture ", 11)) { /* CD in is actually connected to the video in pin */ @@ -539,7 +515,6 @@ static int xonar_dx_control_filter(struct snd_kcontrol_new *template) { if (!strcmp(template->name, "Master Playback Volume")) { template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->info = cs4362a_volume_info; template->tlv.p = cs4362a_db_scale; } else if (!strncmp(template->name, "CD Capture ", 11)) { return 1; /* no CD input */ @@ -577,6 +552,8 @@ static const struct oxygen_model xonar_models[] = { CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF, .dac_channels = 8, + .dac_volume_min = 0x0f, + .dac_volume_max = 0xff, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, @@ -603,6 +580,8 @@ static const struct oxygen_model xonar_models[] = { CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF, .dac_channels = 8, + .dac_volume_min = 0x0f, + .dac_volume_max = 0xff, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, @@ -629,6 +608,8 @@ static const struct oxygen_model xonar_models[] = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2, .dac_channels = 8, + .dac_volume_min = 0, + .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, From e983532e446ac7fabe829d9e3aeff8e26b0a277d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Apr 2008 09:14:30 +0200 Subject: [PATCH 205/250] [ALSA] oxygen: mute by default Initialize the playback volume controls as being muted and having minimal volume. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 6 +++--- sound/pci/oxygen/oxygen.c | 6 +++--- sound/pci/oxygen/oxygen_lib.c | 3 ++- sound/pci/oxygen/virtuoso.c | 23 ++++++++++++----------- 4 files changed, 20 insertions(+), 18 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index bf39c72a130c..dec9073d6ed1 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -66,12 +66,12 @@ static void hifier_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, AK4396_LCH_ATT, 0xff); - ak4396_write(chip, AK4396_RCH_ATT, 0xff); + ak4396_write(chip, AK4396_LCH_ATT, 0); + ak4396_write(chip, AK4396_RCH_ATT, 0); snd_component_add(chip->card, "AK4396"); snd_component_add(chip->card, "CS5340"); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 718169792c3b..636315612c32 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -112,7 +112,7 @@ static void ak4396_init(struct oxygen *chip) struct generic_data *data = chip->model_data; unsigned int i; - data->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); @@ -120,8 +120,8 @@ static void ak4396_init(struct oxygen *chip) AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, i, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, i, AK4396_LCH_ATT, 0xff); - ak4396_write(chip, i, AK4396_RCH_ATT, 0xff); + ak4396_write(chip, i, AK4396_LCH_ATT, 0); + ak4396_write(chip, i, AK4396_RCH_ATT, 0); } snd_component_add(chip->card, "AK4396"); } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b7aa9fcb6b81..f84f6a128ee9 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -221,7 +221,8 @@ static void oxygen_init(struct oxygen *chip) chip->dac_routing = 1; for (i = 0; i < 8; ++i) - chip->dac_volume[i] = chip->model->dac_volume_max; + chip->dac_volume[i] = chip->model->dac_volume_min; + chip->dac_mute = 1; chip->spdif_playback_enable = 1; chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL | (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 37f53a8c5888..fe4e289840b4 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -188,12 +188,13 @@ static void xonar_d2_init(struct oxygen *chip) data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 18, PCM1796_FMT_24_LJUST | PCM1796_ATLD); + pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED | + PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); pcm1796_write(chip, i, 20, PCM1796_OS_64); pcm1796_write(chip, i, 21, 0); - pcm1796_write(chip, i, 16, 0xff); /* set ATL/ATR after ATLD */ - pcm1796_write(chip, i, 17, 0xff); + pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */ + pcm1796_write(chip, i, 17, 0x0f); } oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); @@ -239,8 +240,8 @@ static void xonar_dx_init(struct oxygen *chip) CS4398_DEM_NONE | CS4398_DIF_LJUST); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); - cs4398_write(chip, 5, 0); - cs4398_write(chip, 6, 0); + cs4398_write(chip, 5, 0xfe); + cs4398_write(chip, 6, 0xfe); cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); @@ -250,16 +251,16 @@ static void xonar_dx_init(struct oxygen *chip) cs4362a_write(chip, 0x05, 0); cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE); + cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE); + cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); - cs4362a_write(chip, 0x07, 0); - cs4362a_write(chip, 0x08, 0); - cs4362a_write(chip, 0x0a, 0); - cs4362a_write(chip, 0x0b, 0); - cs4362a_write(chip, 0x0d, 0); - cs4362a_write(chip, 0x0e, 0); + cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE); + cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE); /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); From 4972a177fed34036498aee555335f84a70219bc1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Apr 2008 09:15:45 +0200 Subject: [PATCH 206/250] [ALSA] oxygen: generalize DAC volume TLV handling Add a pointer for DAC volume TLV data to the model structure so that the model driver do not need to manually assign it in their control filter. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 7 ++----- sound/pci/oxygen/oxygen.c | 13 ++----------- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_mixer.c | 5 +++++ sound/pci/oxygen/virtuoso.c | 15 +++++---------- 5 files changed, 15 insertions(+), 26 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index dec9073d6ed1..090dd4354a28 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -127,12 +127,8 @@ static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static int hifier_control_filter(struct snd_kcontrol_new *template) { - if (!strcmp(template->name, "Master Playback Volume")) { - template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->tlv.p = ak4396_db_scale; - } else if (!strcmp(template->name, "Stereo Upmixing")) { + if (!strcmp(template->name, "Stereo Upmixing")) return 1; /* stereo only - we don't need upmixing */ - } return 0; } @@ -148,6 +144,7 @@ static const struct oxygen_model model_hifier = { .set_adc_params = set_cs5340_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, + .dac_tlv = ak4396_db_scale, .model_data_size = sizeof(struct hifier_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 636315612c32..63f185c1ed1e 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -249,27 +249,18 @@ static void set_ak5385_params(struct oxygen *chip, static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); -static int ak4396_control_filter(struct snd_kcontrol_new *template) -{ - if (!strcmp(template->name, "Master Playback Volume")) { - template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->tlv.p = ak4396_db_scale; - } - return 0; -} - static const struct oxygen_model model_generic = { .shortname = "C-Media CMI8788", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .owner = THIS_MODULE, .init = generic_init, - .control_filter = ak4396_control_filter, .cleanup = generic_cleanup, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, + .dac_tlv = ak4396_db_scale, .model_data_size = sizeof(struct generic_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | @@ -291,12 +282,12 @@ static const struct oxygen_model model_meridian = { .chip = "CMI8788", .owner = THIS_MODULE, .init = meridian_init, - .control_filter = ak4396_control_filter, .cleanup = generic_cleanup, .set_dac_params = set_ak4396_params, .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, + .dac_tlv = ak4396_db_scale, .model_data_size = sizeof(struct generic_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 36f2f81fefc5..a71c6e059260 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -100,6 +100,7 @@ struct oxygen_model { void (*gpio_changed)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, unsigned int reg, unsigned int mute); + const unsigned int *dac_tlv; size_t model_data_size; unsigned int pcm_dev_cfg; u8 dac_channels; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 056581ecd915..cc0cddadd589 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -941,6 +941,11 @@ static int add_controls(struct oxygen *chip, return err; if (err == 1) continue; + if (!strcmp(template.name, "Master Playback Volume") && + chip->model->dac_tlv) { + template.tlv.p = chip->model->dac_tlv; + template.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + } ctl = snd_ctl_new1(&template, chip); if (!ctl) return -ENOMEM; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index fe4e289840b4..7f84fa5deca2 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -502,24 +502,16 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { - if (!strcmp(template->name, "Master Playback Volume")) { - template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->tlv.p = pcm1796_db_scale; - } else if (!strncmp(template->name, "CD Capture ", 11)) { + if (!strncmp(template->name, "CD Capture ", 11)) /* CD in is actually connected to the video in pin */ template->private_value ^= AC97_CD ^ AC97_VIDEO; - } return 0; } static int xonar_dx_control_filter(struct snd_kcontrol_new *template) { - if (!strcmp(template->name, "Master Playback Volume")) { - template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - template->tlv.p = cs4362a_db_scale; - } else if (!strncmp(template->name, "CD Capture ", 11)) { + if (!strncmp(template->name, "CD Capture ", 11)) return 1; /* no CD input */ - } return 0; } @@ -547,6 +539,7 @@ static const struct oxygen_model xonar_models[] = { .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, + .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | @@ -575,6 +568,7 @@ static const struct oxygen_model xonar_models[] = { .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .gpio_changed = xonar_gpio_changed, + .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | @@ -604,6 +598,7 @@ static const struct oxygen_model xonar_models[] = { .update_dac_mute = update_cs43xx_mute, .gpio_changed = xonar_gpio_changed, .ac97_switch = xonar_dx_ac97_switch, + .dac_tlv = cs4362a_db_scale, .model_data_size = sizeof(struct xonar_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | From 0a08478c0f7548211b492b578a67dacca5aea1a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Apr 2008 12:59:55 +0200 Subject: [PATCH 207/250] [ALSA] soc - wm9712: Remove unneeded AC97_EXTENDED_MID updates Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9712.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 524f7450804f..d2d79e182a45 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -581,22 +581,14 @@ static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) switch (event) { case SNDRV_CTL_POWER_D0: /* full On */ - /* liam - maybe enable thermal shutdown */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff; - ac97_write(codec, AC97_EXTENDED_MID, reg); - break; case SNDRV_CTL_POWER_D1: /* partial On */ case SNDRV_CTL_POWER_D2: /* partial On */ break; case SNDRV_CTL_POWER_D3hot: /* Off, with power */ - /* enable master bias and vmid */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff; - ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SNDRV_CTL_POWER_D3cold: /* Off, without power */ /* disable everything including AC link */ - ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; From 7c2ba97b8a99c857758fd03513350b39a8b242d6 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 16 Apr 2008 13:13:59 +0200 Subject: [PATCH 208/250] [ALSA] hda: Add 5.1 support for second headphone jack Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks, the second headphone jack should be used for the 5.1 surround sound. Add support for 'Headphone as Line Out' switch, which allows it be used in 5.1 surround sound. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 60 +++++++++++++++++++++++++++++++++- 1 file changed, 59 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e1d07ab5cd15..b3a15d616873 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -181,6 +181,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; unsigned int clfe_swap; + unsigned int hp_switch; unsigned int aloopback; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -1274,7 +1275,7 @@ static unsigned int ref92hd73xx_pin_configs[13] = { static unsigned int dell_m6_pin_configs[13] = { 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, - 0x03a11020, 0x03011050, 0x4f0000f0, 0x4f0000f0, + 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, }; @@ -2052,6 +2053,34 @@ static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info + +static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->hp_switch; + return 0; +} + +static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + spec->hp_switch = ucontrol->value.integer.value[0]; + + /* check to be sure that the ports are upto date with + * switch changes + */ + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + + return 1; +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2123,6 +2152,15 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, return 1; } +#define STAC_CODEC_HP_SWITCH(xname) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .info = stac92xx_hp_switch_info, \ + .get = stac92xx_hp_switch_get, \ + .put = stac92xx_hp_switch_put, \ + } + #define STAC_CODEC_IO_SWITCH(xname, xpval) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ @@ -2147,6 +2185,7 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2155,6 +2194,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2419,6 +2459,14 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } + if (cfg->hp_outs > 1) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_HP_SWITCH, + "Headphone as Line Out Switch", 0); + if (err < 0) + return err; + } + if (spec->line_switch) { nid = cfg->input_pins[AUTO_PIN_LINE]; pincap = snd_hda_param_read(codec, nid, @@ -3163,6 +3211,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; int i, presence; presence = 0; @@ -3173,11 +3222,15 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) for (i = 0; i < cfg->hp_outs; i++) { if (presence) break; + if (spec->hp_switch && cfg->hp_pins[i] == nid) + break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } if (presence) { /* disable lineouts, enable hp */ + if (spec->hp_switch) + stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); @@ -3190,6 +3243,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) ~spec->eapd_mask); } else { /* enable lineouts, disable hp */ + if (spec->hp_switch) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); @@ -3201,6 +3256,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) spec->gpio_dir, spec->gpio_data | spec->eapd_mask); } + if (!spec->hp_switch && cfg->hp_outs > 1 && presence) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); } static void stac92xx_pin_sense(struct hda_codec *codec, int idx) @@ -3459,6 +3516,7 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ + spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; From cb308f97aee2c816834240c8d5f7c98dd8aff157 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 16 Apr 2008 14:13:29 +0200 Subject: [PATCH 209/250] [ALSA] hda - Fix ALC889A codec support ALC889A is recognized ALC885/ALC882 but it's actually closer to ALC888/ALC883. Cc: Kasper Sandberg Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 181db2177a1d..fb09e4429b27 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14042,6 +14042,8 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, + { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", + .patch = patch_alc883 }, { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, From 7943a8aba93ab439bdfbd9b92221720a4a4d8153 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 16 Apr 2008 17:29:09 +0200 Subject: [PATCH 210/250] [ALSA] hda - Avoid unexpected breakage with ALC889A hack The last ALC889A hack may break on some devices with certain model presets since patch_alc*() have different model tables. So, now it's handled in the original patch_alc882() but fly to patch_alc883() in model=auto appropriately. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fb09e4429b27..732515dcc99d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2579,6 +2579,7 @@ static void alc_free(struct hda_codec *codec) kfree(spec->kctl_alloc); } kfree(spec); + codec->spec = NULL; /* to be sure */ } /* @@ -6336,6 +6337,8 @@ static void alc882_auto_init(struct hda_codec *codec) alc_sku_automute(codec); } +static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */ + static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; @@ -6365,6 +6368,11 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MBP3; break; default: + /* ALC889A is handled better as ALC888-compatible */ + if (codec->revision_id == 0x100103) { + alc_free(codec); + return patch_alc883(codec); + } printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); board_config = ALC882_AUTO; @@ -14043,7 +14051,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", - .patch = patch_alc883 }, + .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, From 369b240d6391aef41d376c9e8769fd939c7c6c4d Mon Sep 17 00:00:00 2001 From: Roel Kluin <12o3l@tiscali.nl> Date: Wed, 16 Apr 2008 19:30:30 +0200 Subject: [PATCH 211/250] [ALSA] sound/drivers/dummy.c: fix negative snd_pcm_format_width() check bps is unsigned, a negative snd_pcm_format_width() return value is not noticed Signed-off-by: Roel Kluin <12o3l@tiscali.nl> Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 83ac4980c5fb..4e4c69e6cb4c 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -230,13 +230,14 @@ static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dummy_pcm *dpcm = runtime->private_data; - unsigned int bps; + int bps; + + bps = snd_pcm_format_width(runtime->format) * runtime->rate * + runtime->channels / 8; - bps = runtime->rate * runtime->channels; - bps *= snd_pcm_format_width(runtime->format); - bps /= 8; if (bps <= 0) return -EINVAL; + dpcm->pcm_bps = bps; dpcm->pcm_hz = HZ; dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream); From 9a4f20fcbd9cd89d8e4cfcaece81802c19d467ae Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Wed, 16 Apr 2008 13:15:38 +0200 Subject: [PATCH 212/250] [ALSA] snd-powermac: enable headphone detection Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/pmac.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 613a565e04de..6f68ac9982e4 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -1300,9 +1300,9 @@ int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) snd_pmac_sound_feature(chip, 1); - /* reset */ - if (chip->model == PMAC_AWACS) - out_le32(&chip->awacs->control, 0x11); + /* reset & enable interrupts */ + if (chip->model <= PMAC_BURGUNDY) + out_le32(&chip->awacs->control, chip->control_mask); /* Powerbooks have odd ways of enabling inputs such as an expansion-bay CD or sound from an internal modem From 946cda7d64b81053beac5d31148996f3e379e89e Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Wed, 16 Apr 2008 13:16:05 +0200 Subject: [PATCH 213/250] [ALSA] snd-powermac: style pmac.c Coding style corrections for pmac.c. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/pmac.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 6f68ac9982e4..a38c0c790d2b 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -214,7 +214,7 @@ static int snd_pmac_pcm_prepare(struct snd_pmac *chip, struct pmac_stream *rec, int rate_index; long offset; struct pmac_stream *astr; - + rec->dma_size = snd_pcm_lib_buffer_bytes(subs); rec->period_size = snd_pcm_lib_period_bytes(subs); rec->nperiods = rec->dma_size / rec->period_size; @@ -643,7 +643,7 @@ static int snd_pmac_pcm_close(struct snd_pmac *chip, struct pmac_stream *rec, /* reset constraints */ astr->cur_freqs = chip->freqs_ok; astr->cur_formats = chip->formats_ok; - + return 0; } From a8c2a6bf464d983c642c8b8b001a57aabbf76673 Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Thu, 17 Apr 2008 17:55:30 +0200 Subject: [PATCH 214/250] [ALSA] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SL Add mixer controls and correct headphone detection bits for PowerMacs 7300/7500 (AWACS) and G3 Beige (Screamer), and iMac G3 Slot-loading (Screamer). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 171 +++++++++++++++++++++++++++++++++++++--------- sound/ppc/awacs.h | 16 ++++- 2 files changed, 151 insertions(+), 36 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 8441e780df00..db4e35d28244 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -544,7 +544,7 @@ static int snd_pmac_screamer_mic_boost_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; + uinfo->value.integer.max = 3; return 0; } @@ -552,16 +552,14 @@ static int snd_pmac_screamer_mic_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int val; + int val = 0; unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->awacs_reg[6] & MASK_MIC_BOOST) - val = 2; - else if (chip->awacs_reg[0] & MASK_GAINLINE) - val = 1; - else - val = 0; + val |= 2; + if (chip->awacs_reg[0] & MASK_GAINLINE) + val |= 1; spin_unlock_irqrestore(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = val; return 0; @@ -578,11 +576,10 @@ static int snd_pmac_screamer_mic_boost_put(struct snd_kcontrol *kcontrol, spin_lock_irqsave(&chip->reg_lock, flags); val0 = chip->awacs_reg[0] & ~MASK_GAINLINE; val6 = chip->awacs_reg[6] & ~MASK_MIC_BOOST; - if (ucontrol->value.integer.value[0] > 0) { + if (ucontrol->value.integer.value[0] & 1) val0 |= MASK_GAINLINE; - if (ucontrol->value.integer.value[0] > 1) - val6 |= MASK_MIC_BOOST; - } + if (ucontrol->value.integer.value[0] & 2) + val6 |= MASK_MIC_BOOST; if (val0 != chip->awacs_reg[0]) { snd_pmac_awacs_write_reg(chip, 0, val0); changed = 1; @@ -599,9 +596,32 @@ static int snd_pmac_screamer_mic_boost_put(struct snd_kcontrol *kcontrol, * lists of mixer elements */ static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __initdata = { - AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("Master Capture Switch", 1, SHIFT_LOOPTHRU, 0), - AWACS_VOLUME("Capture Volume", 0, 4, 0), + AWACS_VOLUME("Master Capture Volume", 0, 4, 0), +/* AWACS_SWITCH("Unknown Playback Switch", 6, SHIFT_PAROUT0, 0), */ +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = { + AWACS_VOLUME("Master Playback Volume", 2, 6, 1), + AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), + AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { + AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), + AWACS_VOLUME("Master Playback Volume", 5, 6, 1), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), +}; + +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = { + AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), + AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), +}; + +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { + AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; @@ -621,25 +641,49 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = { static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata = AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata = +AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1); + static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = { - AWACS_SWITCH("Mic Boost", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost", + .name = "Mic Boost Capture Volume", .info = snd_pmac_screamer_mic_boost_info, .get = snd_pmac_screamer_mic_boost_get, .put = snd_pmac_screamer_mic_boost_put, }, }; +static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("CD Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("Mic Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), +}; + static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = { AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), }; + static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata = AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac __initdata = +AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); + /* * add new mixer elements to the card @@ -708,6 +752,14 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) } #endif /* CONFIG_PM */ +#define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) +#define IS_IMAC (machine_is_compatible("PowerMac2,1") \ + || machine_is_compatible("PowerMac2,2") \ + || machine_is_compatible("PowerMac4,1")) + +static int imac; + #ifdef PMAC_SUPPORT_AUTOMUTE /* * auto-mute stuffs @@ -750,9 +802,16 @@ static void snd_pmac_awacs_update_automute(struct snd_pmac *chip, int do_notify) } else #endif { - int reg = chip->awacs_reg[1] | (MASK_HDMUTE|MASK_SPKMUTE); + int reg = chip->awacs_reg[1] + | (MASK_HDMUTE | MASK_SPKMUTE); + if (imac) { + reg &= ~MASK_SPKMUTE; + reg &= ~MASK_PAROUT1; + } if (snd_pmac_awacs_detect_headphone(chip)) reg &= ~MASK_HDMUTE; + else if (imac) + reg |= MASK_PAROUT1; else reg &= ~MASK_SPKMUTE; if (do_notify && reg == chip->awacs_reg[1]) @@ -778,8 +837,11 @@ static void snd_pmac_awacs_update_automute(struct snd_pmac *chip, int do_notify) int __init snd_pmac_awacs_init(struct snd_pmac *chip) { + int pm7500 = IS_PM7500; + int beige = IS_BEIGE; int err, vol; + imac = IS_IMAC; /* looks like MASK_GAINLINE triggers something, so we set here * as start-up */ @@ -826,20 +888,25 @@ snd_pmac_awacs_init(struct snd_pmac *chip) /* set headphone-jack detection bit */ switch (chip->model) { case PMAC_AWACS: - chip->hp_stat_mask = 0x04; + chip->hp_stat_mask = pm7500 ? MASK_HDPCONN + : MASK_LOCONN; break; case PMAC_SCREAMER: switch (chip->device_id) { case 0x08: - /* 1 = side jack, 2 = front jack */ - chip->hp_stat_mask = 0x03; + case 0x0B: + chip->hp_stat_mask = imac + ? MASK_LOCONN_IMAC | + MASK_HDPLCONN_IMAC | + MASK_HDPRCONN_IMAC + : MASK_HDPCONN; break; case 0x00: case 0x05: - chip->hp_stat_mask = 0x04; + chip->hp_stat_mask = MASK_LOCONN; break; default: - chip->hp_stat_mask = 0x08; + chip->hp_stat_mask = MASK_HDPCONN; break; } break; @@ -857,15 +924,37 @@ snd_pmac_awacs_init(struct snd_pmac *chip) if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers), snd_pmac_awacs_mixers)) < 0) return err; - if (chip->model == PMAC_SCREAMER) + if (beige) + ; + else if (chip->model == PMAC_SCREAMER) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2), snd_pmac_screamer_mixers2); - else + else if (!pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers2), snd_pmac_awacs_mixers2); if (err < 0) return err; - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_master_sw, chip); + if (pm7500) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), + snd_pmac_awacs_mixers_pmac7500); + else if (beige) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mixers_beige), + snd_pmac_screamer_mixers_beige); + else if (imac) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mixers_imac), + snd_pmac_screamer_mixers_imac); + else + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers_pmac), + snd_pmac_awacs_mixers_pmac); + if (err < 0) + return err; + chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac) + ? &snd_pmac_awacs_master_sw_imac + : &snd_pmac_awacs_master_sw, chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; #ifdef PMAC_AMP_AVAIL @@ -893,20 +982,34 @@ snd_pmac_awacs_init(struct snd_pmac *chip) if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), snd_pmac_awacs_speaker_vol)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_speaker_sw, chip); + chip->speaker_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_awacs_speaker_sw_imac + : &snd_pmac_awacs_speaker_sw, chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; } - if (chip->model == PMAC_SCREAMER) { - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mic_boost), - snd_pmac_screamer_mic_boost)) < 0) - return err; - } else { - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mic_boost), - snd_pmac_awacs_mic_boost)) < 0) - return err; - } + if (beige) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige), + snd_pmac_screamer_mic_boost_beige); + else if (imac) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost_imac), + snd_pmac_screamer_mic_boost_imac); + else if (chip->model == PMAC_SCREAMER) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost), + snd_pmac_screamer_mic_boost); + else if (pm7500) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mic_boost_pmac7500), + snd_pmac_awacs_mic_boost_pmac7500); + else + err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mic_boost), + snd_pmac_awacs_mic_boost); + if (err < 0) + return err; /* * set lowlevel callbacks diff --git a/sound/ppc/awacs.h b/sound/ppc/awacs.h index 1b2cc44eda57..684bfa7cfff3 100644 --- a/sound/ppc/awacs.h +++ b/sound/ppc/awacs.h @@ -116,6 +116,11 @@ struct awacs_regs { #define MASK_HDMUTE MASK_AMUTE #define SHIFT_HDMUTE 9 #define MASK_PAROUT (0x3 << 10) /* Parallel Out (???) */ +#define MASK_PAROUT0 (0x1 << 10) /* Parallel Out (???) */ +#define MASK_PAROUT1 (0x1 << 11) /* Parallel Out (enable speaker) */ +#define SHIFT_PAROUT 10 +#define SHIFT_PAROUT0 10 +#define SHIFT_PAROUT1 11 #define SAMPLERATE_48000 (0x0 << 3) /* 48 or 44.1 kHz */ #define SAMPLERATE_32000 (0x1 << 3) /* 32 or 29.4 kHz */ @@ -152,8 +157,15 @@ struct awacs_regs { #define MASK_REVISION (0xf << 12) /* Revision Number */ #define MASK_MFGID (0xf << 8) /* Mfg. ID */ #define MASK_CODSTATRES (0xf << 4) /* bits 4 - 7 reserved */ -#define MASK_INPPORT (0xf) /* Input Port */ -#define MASK_HDPCONN 8 /* headphone plugged in */ +#define MASK_INSENSE (0xf) /* port sense bits: */ +#define MASK_HDPCONN 8 /* headphone plugged in */ +#define MASK_LOCONN 4 /* line-out plugged in */ +#define MASK_LICONN 2 /* line-in plugged in */ +#define MASK_MICCONN 1 /* microphone plugged in */ +#define MASK_LICONN_IMAC 8 /* line-in plugged in */ +#define MASK_HDPRCONN_IMAC 4 /* headphone right plugged in */ +#define MASK_HDPLCONN_IMAC 2 /* headphone left plugged in */ +#define MASK_LOCONN_IMAC 1 /* line-out plugged in */ /* Clipping Count Reg Bit Masks */ /* -------- ----- --- --- ----- */ From 7ae44cfa7ab29b277691327e8de790d7b880722f Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Wed, 16 Apr 2008 19:39:27 +0200 Subject: [PATCH 215/250] [ALSA] snd-powermac: style awacs.s and awacs.h Coding style corrections for awacs.c and awacs.h. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 94 +++++++++++++++++++++++++++++------------------ sound/ppc/awacs.h | 5 ++- 2 files changed, 62 insertions(+), 37 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index db4e35d28244..566a6d0daf4a 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -141,7 +141,7 @@ static int snd_pmac_awacs_info_volume(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 15; return 0; } - + static int snd_pmac_awacs_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -267,7 +267,8 @@ static int snd_pmac_awacs_put_switch(struct snd_kcontrol *kcontrol, static void awacs_set_cuda(int reg, int val) { struct adb_request req; - cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a, reg, val); + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a, + reg, val); while (! req.complete) cuda_poll(); } @@ -289,11 +290,11 @@ static void awacs_amp_set_tone(struct awacs_amp *amp, int bass, int treble) /* * vol = 0 - 31 (attenuation), 32 = mute bit, stereo */ -static int awacs_amp_set_vol(struct awacs_amp *amp, int index, int lvol, int rvol, - int do_check) +static int awacs_amp_set_vol(struct awacs_amp *amp, int index, + int lvol, int rvol, int do_check) { if (do_check && amp->amp_vol[index][0] == lvol && - amp->amp_vol[index][1] == rvol) + amp->amp_vol[index][1] == rvol) return 0; awacs_set_cuda(3 + index, lvol); awacs_set_cuda(5 + index, rvol); @@ -337,7 +338,7 @@ static int snd_pmac_awacs_info_volume_amp(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 31; return 0; } - + static int snd_pmac_awacs_get_volume_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -361,8 +362,10 @@ static int snd_pmac_awacs_put_volume_amp(struct snd_kcontrol *kcontrol, snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - vol[0] = (31 - (ucontrol->value.integer.value[0] & 31)) | (amp->amp_vol[index][0] & 32); - vol[1] = (31 - (ucontrol->value.integer.value[1] & 31)) | (amp->amp_vol[index][1] & 32); + vol[0] = (31 - (ucontrol->value.integer.value[0] & 31)) + | (amp->amp_vol[index][0] & 32); + vol[1] = (31 - (ucontrol->value.integer.value[1] & 31)) + | (amp->amp_vol[index][1] & 32); return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1); } @@ -374,8 +377,10 @@ static int snd_pmac_awacs_get_switch_amp(struct snd_kcontrol *kcontrol, struct awacs_amp *amp = chip->mixer_data; snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32) ? 0 : 1; - ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32) ? 0 : 1; + ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32) + ? 0 : 1; + ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32) + ? 0 : 1; return 0; } @@ -389,8 +394,10 @@ static int snd_pmac_awacs_put_switch_amp(struct snd_kcontrol *kcontrol, snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32) | (amp->amp_vol[index][0] & 31); - vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32) | (amp->amp_vol[index][1] & 31); + vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32) + | (amp->amp_vol[index][0] & 31); + vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32) + | (amp->amp_vol[index][1] & 31); return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1); } @@ -403,7 +410,7 @@ static int snd_pmac_awacs_info_tone_amp(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 14; return 0; } - + static int snd_pmac_awacs_get_tone_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -445,7 +452,7 @@ static int snd_pmac_awacs_info_master_amp(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 99; return 0; } - + static int snd_pmac_awacs_get_master_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -688,12 +695,14 @@ AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); /* * add new mixer elements to the card */ -static int build_mixers(struct snd_pmac *chip, int nums, struct snd_kcontrol_new *mixers) +static int build_mixers(struct snd_pmac *chip, int nums, + struct snd_kcontrol_new *mixers) { int i, err; for (i = 0; i < nums; i++) { - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip))) < 0) + err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip)); + if (err < 0) return err; } return 0; @@ -743,8 +752,10 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) #ifdef PMAC_AMP_AVAIL if (chip->mixer_data) { struct awacs_amp *amp = chip->mixer_data; - awacs_amp_set_vol(amp, 0, amp->amp_vol[0][0], amp->amp_vol[0][1], 0); - awacs_amp_set_vol(amp, 1, amp->amp_vol[1][0], amp->amp_vol[1][1], 0); + awacs_amp_set_vol(amp, 0, + amp->amp_vol[0][0], amp->amp_vol[0][1], 0); + awacs_amp_set_vol(amp, 1, + amp->amp_vol[1][0], amp->amp_vol[1][1], 0); awacs_amp_set_tone(amp, amp->amp_tone[0], amp->amp_tone[1]); awacs_amp_set_master(amp, amp->amp_master); } @@ -849,7 +860,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) chip->awacs_reg[1] = MASK_CMUTE | MASK_AMUTE; /* FIXME: Only machines with external SRS module need MASK_PAROUT */ if (chip->has_iic || chip->device_id == 0x5 || - /*chip->_device_id == 0x8 || */ + /* chip->_device_id == 0x8 || */ chip->device_id == 0xb) chip->awacs_reg[1] |= MASK_PAROUT; /* get default volume from nvram */ @@ -860,8 +871,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip) chip->awacs_reg[2] = vol; chip->awacs_reg[4] = vol; if (chip->model == PMAC_SCREAMER) { - chip->awacs_reg[5] = vol; /* FIXME: screamer has loopthru vol control */ - chip->awacs_reg[6] = MASK_MIC_BOOST; /* FIXME: maybe should be vol << 3 for PCMCIA speaker */ + /* FIXME: screamer has loopthru vol control */ + chip->awacs_reg[5] = vol; + /* FIXME: maybe should be vol << 3 for PCMCIA speaker */ + chip->awacs_reg[6] = MASK_MIC_BOOST; chip->awacs_reg[7] = 0; } @@ -877,7 +890,8 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return -ENOMEM; chip->mixer_data = amp; chip->mixer_free = awacs_amp_free; - awacs_amp_set_vol(amp, 0, 63, 63, 0); /* mute and zero vol */ + /* mute and zero vol */ + awacs_amp_set_vol(amp, 0, 63, 63, 0); awacs_amp_set_vol(amp, 1, 63, 63, 0); awacs_amp_set_tone(amp, 7, 7); /* 0 dB */ awacs_amp_set_master(amp, 79); /* 0 dB */ @@ -921,8 +935,9 @@ snd_pmac_awacs_init(struct snd_pmac *chip) */ strcpy(chip->card->mixername, "PowerMac AWACS"); - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers), - snd_pmac_awacs_mixers)) < 0) + err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers), + snd_pmac_awacs_mixers); + if (err < 0) return err; if (beige) ; @@ -955,7 +970,8 @@ snd_pmac_awacs_init(struct snd_pmac *chip) chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac) ? &snd_pmac_awacs_master_sw_imac : &snd_pmac_awacs_master_sw, chip); - if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + err = snd_ctl_add(chip->card, chip->master_sw_ctl); + if (err < 0) return err; #ifdef PMAC_AMP_AVAIL if (chip->mixer_data) { @@ -965,27 +981,34 @@ snd_pmac_awacs_init(struct snd_pmac *chip) * screamer registers. * in this case, it seems the route C is not used. */ - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_amp_vol), - snd_pmac_awacs_amp_vol)) < 0) + err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_amp_vol), + snd_pmac_awacs_amp_vol); + if (err < 0) return err; /* overwrite */ - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw, chip); - if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw, + chip); + err = snd_ctl_add(chip->card, chip->master_sw_ctl); + if (err < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw, chip); - if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) + chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw, + chip); + err = snd_ctl_add(chip->card, chip->speaker_sw_ctl); + if (err < 0) return err; } else #endif /* PMAC_AMP_AVAIL */ { /* route A = headphone, route C = speaker */ - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), - snd_pmac_awacs_speaker_vol)) < 0) + err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), + snd_pmac_awacs_speaker_vol); + if (err < 0) return err; chip->speaker_sw_ctl = snd_ctl_new1(imac ? &snd_pmac_awacs_speaker_sw_imac : &snd_pmac_awacs_speaker_sw, chip); - if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) + err = snd_ctl_add(chip->card, chip->speaker_sw_ctl); + if (err < 0) return err; } @@ -1020,7 +1043,8 @@ snd_pmac_awacs_init(struct snd_pmac *chip) chip->resume = snd_pmac_awacs_resume; #endif #ifdef PMAC_SUPPORT_AUTOMUTE - if ((err = snd_pmac_add_automute(chip)) < 0) + err = snd_pmac_add_automute(chip); + if (err < 0) return err; chip->detect_headphone = snd_pmac_awacs_detect_headphone; chip->update_automute = snd_pmac_awacs_update_automute; diff --git a/sound/ppc/awacs.h b/sound/ppc/awacs.h index 684bfa7cfff3..c33e6a531cf7 100644 --- a/sound/ppc/awacs.h +++ b/sound/ppc/awacs.h @@ -144,7 +144,7 @@ struct awacs_regs { #define VOLLEFT(x) (((~(x)) << 6) & MASK_OUTVOLLEFT) /* address 6 */ -#define MASK_MIC_BOOST (0x4) /* screamer mic boost */ +#define MASK_MIC_BOOST (0x4) /* screamer mic boost */ #define SHIFT_MIC_BOOST 2 /* Audio Codec Status Reg Bit Masks */ @@ -175,7 +175,8 @@ struct awacs_regs { /* DBDMA ChannelStatus Bit Masks */ /* ----- ------------- --- ----- */ #define MASK_CSERR (0x1 << 7) /* Error */ -#define MASK_EOI (0x1 << 6) /* End of Input -- only for Input Channel */ +#define MASK_EOI (0x1 << 6) /* End of Input -- + only for Input Channel */ #define MASK_CSUNUSED (0x1f << 1) /* bits 1-5 not used */ #define MASK_WAIT (0x1) /* Wait */ From 44deee129c9af3759d3e5e772b82012742dc57a0 Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Wed, 16 Apr 2008 19:45:31 +0200 Subject: [PATCH 216/250] [ALSA] snd-powermac: Burgundy mixers for B&W and iMac Add mixer controls and correct headphone detection bits for PowerMac G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/burgundy.c | 441 +++++++++++++++++++++++++++++++++++-------- sound/ppc/burgundy.h | 31 ++- 2 files changed, 385 insertions(+), 87 deletions(-) diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 1a545ac0de04..f0c12a97fdbc 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -211,68 +211,94 @@ static int snd_pmac_burgundy_put_volume(struct snd_kcontrol *kcontrol, nvoices[1] != ucontrol->value.integer.value[1]); } -#define BURGUNDY_VOLUME(xname, xindex, addr, shift) \ +#define BURGUNDY_VOLUME_W(xname, xindex, addr, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ .info = snd_pmac_burgundy_info_volume,\ .get = snd_pmac_burgundy_get_volume,\ .put = snd_pmac_burgundy_put_volume,\ .private_value = ((ADDR2BASE(addr) & 0xff) | ((shift) << 8)) } -/* lineout/speaker */ - -static int snd_pmac_burgundy_info_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +/* + * Burgundy volume: 0 - 100, stereo, 2-byte reg + */ +static void +snd_pmac_burgundy_write_volume_2b(struct snd_pmac *chip, unsigned int address, + long *volume, int off) { - int stereo = (kcontrol->private_value >> 24) & 1; - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = stereo + 1; + int lvolume, rvolume; + + off |= off << 2; + lvolume = volume[0] ? volume[0] + BURGUNDY_VOLUME_OFFSET : 0; + rvolume = volume[1] ? volume[1] + BURGUNDY_VOLUME_OFFSET : 0; + + snd_pmac_burgundy_wcb(chip, address + off, lvolume); + snd_pmac_burgundy_wcb(chip, address + off + 0x500, rvolume); +} + +static void +snd_pmac_burgundy_read_volume_2b(struct snd_pmac *chip, unsigned int address, + long *volume, int off) +{ + volume[0] = snd_pmac_burgundy_rcb(chip, address + off); + if (volume[0] >= BURGUNDY_VOLUME_OFFSET) + volume[0] -= BURGUNDY_VOLUME_OFFSET; + else + volume[0] = 0; + volume[1] = snd_pmac_burgundy_rcb(chip, address + off + 0x100); + if (volume[1] >= BURGUNDY_VOLUME_OFFSET) + volume[1] -= BURGUNDY_VOLUME_OFFSET; + else + volume[1] = 0; +} + +static int snd_pmac_burgundy_info_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.integer.max = 100; return 0; } -static int snd_pmac_burgundy_get_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_get_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int lmask = kcontrol->private_value & 0xff; - int rmask = (kcontrol->private_value >> 8) & 0xff; - int stereo = (kcontrol->private_value >> 24) & 1; - int val = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; - if (stereo) - ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int off = kcontrol->private_value & 0x300; + snd_pmac_burgundy_read_volume_2b(chip, addr, + ucontrol->value.integer.value, off); return 0; } -static int snd_pmac_burgundy_put_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_put_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int lmask = kcontrol->private_value & 0xff; - int rmask = (kcontrol->private_value >> 8) & 0xff; - int stereo = (kcontrol->private_value >> 24) & 1; - int val, oval; - oval = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - val = oval & ~(lmask | rmask); - if (ucontrol->value.integer.value[0]) - val |= lmask; - if (stereo && ucontrol->value.integer.value[1]) - val |= rmask; - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, val); - return val != oval; + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int off = kcontrol->private_value & 0x300; + long nvoices[2]; + + snd_pmac_burgundy_write_volume_2b(chip, addr, + ucontrol->value.integer.value, off); + snd_pmac_burgundy_read_volume_2b(chip, addr, nvoices, off); + return (nvoices[0] != ucontrol->value.integer.value[0] || + nvoices[1] != ucontrol->value.integer.value[1]); } -#define BURGUNDY_OUTPUT_SWITCH(xname, xindex, lmask, rmask, stereo) \ +#define BURGUNDY_VOLUME_2B(xname, xindex, addr, off) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ - .info = snd_pmac_burgundy_info_switch_out,\ - .get = snd_pmac_burgundy_get_switch_out,\ - .put = snd_pmac_burgundy_put_switch_out,\ - .private_value = ((lmask) | ((rmask) << 8) | ((stereo) << 24)) } + .info = snd_pmac_burgundy_info_volume_2b,\ + .get = snd_pmac_burgundy_get_volume_2b,\ + .put = snd_pmac_burgundy_put_volume_2b,\ + .private_value = ((ADDR2BASE(addr) & 0xff) | ((off) << 8)) } -/* line/speaker output volume */ -static int snd_pmac_burgundy_info_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +/* + * Burgundy gain/attenuation: 0 - 15, mono/stereo, byte reg + */ +static int snd_pmac_burgundy_info_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int stereo = (kcontrol->private_value >> 24) & 1; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -282,60 +308,269 @@ static int snd_pmac_burgundy_info_volume_out(struct snd_kcontrol *kcontrol, return 0; } -static int snd_pmac_burgundy_get_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_get_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; + int atten = (kcontrol->private_value >> 25) & 1; int oval; - oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; + oval = snd_pmac_burgundy_rcb(chip, addr); + if (atten) + oval = ~oval & 0xff; ucontrol->value.integer.value[0] = oval & 0xf; if (stereo) ucontrol->value.integer.value[1] = (oval >> 4) & 0xf; return 0; } -static int snd_pmac_burgundy_put_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_put_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; - unsigned int oval, val; + int atten = (kcontrol->private_value >> 25) & 1; + int oval, val; - oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; - val = ucontrol->value.integer.value[0] & 15; + oval = snd_pmac_burgundy_rcb(chip, addr); + if (atten) + oval = ~oval & 0xff; + val = ucontrol->value.integer.value[0]; if (stereo) - val |= (ucontrol->value.integer.value[1] & 15) << 4; + val |= ucontrol->value.integer.value[1] << 4; else - val |= val << 4; - val = ~val & 0xff; + val |= ucontrol->value.integer.value[0] << 4; + if (atten) + val = ~val & 0xff; snd_pmac_burgundy_wcb(chip, addr, val); return val != oval; } -#define BURGUNDY_OUTPUT_VOLUME(xname, xindex, addr, stereo) \ +#define BURGUNDY_VOLUME_B(xname, xindex, addr, stereo, atten) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ - .info = snd_pmac_burgundy_info_volume_out,\ - .get = snd_pmac_burgundy_get_volume_out,\ - .put = snd_pmac_burgundy_put_volume_out,\ - .private_value = (ADDR2BASE(addr) | ((stereo) << 24)) } + .info = snd_pmac_burgundy_info_gain,\ + .get = snd_pmac_burgundy_get_gain,\ + .put = snd_pmac_burgundy_put_gain,\ + .private_value = (ADDR2BASE(addr) | ((stereo) << 24) | ((atten) << 25)) } +/* + * Burgundy switch: 0/1, mono/stereo, word reg + */ +static int snd_pmac_burgundy_info_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int stereo = (kcontrol->private_value >> 24) & 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = stereo + 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_pmac_burgundy_get_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = 1 << (kcontrol->private_value & 0xff); + int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff); + int stereo = (kcontrol->private_value >> 24) & 1; + int val = snd_pmac_burgundy_rcw(chip, addr); + ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; + if (stereo) + ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; + return 0; +} + +static int snd_pmac_burgundy_put_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = 1 << (kcontrol->private_value & 0xff); + int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff); + int stereo = (kcontrol->private_value >> 24) & 1; + int val, oval; + oval = snd_pmac_burgundy_rcw(chip, addr); + val = oval & ~(lmask | (stereo ? rmask : 0)); + if (ucontrol->value.integer.value[0]) + val |= lmask; + if (stereo && ucontrol->value.integer.value[1]) + val |= rmask; + snd_pmac_burgundy_wcw(chip, addr, val); + return val != oval; +} + +#define BURGUNDY_SWITCH_W(xname, xindex, addr, lbit, rbit, stereo) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ + .info = snd_pmac_burgundy_info_switch_w,\ + .get = snd_pmac_burgundy_get_switch_w,\ + .put = snd_pmac_burgundy_put_switch_w,\ + .private_value = ((lbit) | ((rbit) << 8)\ + | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) } + +/* + * Burgundy switch: 0/1, mono/stereo, byte reg, bit mask + */ +static int snd_pmac_burgundy_info_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int stereo = (kcontrol->private_value >> 24) & 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = stereo + 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_pmac_burgundy_get_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = kcontrol->private_value & 0xff; + int rmask = (kcontrol->private_value >> 8) & 0xff; + int stereo = (kcontrol->private_value >> 24) & 1; + int val = snd_pmac_burgundy_rcb(chip, addr); + ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; + if (stereo) + ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; + return 0; +} + +static int snd_pmac_burgundy_put_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = kcontrol->private_value & 0xff; + int rmask = (kcontrol->private_value >> 8) & 0xff; + int stereo = (kcontrol->private_value >> 24) & 1; + int val, oval; + oval = snd_pmac_burgundy_rcb(chip, addr); + val = oval & ~(lmask | rmask); + if (ucontrol->value.integer.value[0]) + val |= lmask; + if (stereo && ucontrol->value.integer.value[1]) + val |= rmask; + snd_pmac_burgundy_wcb(chip, addr, val); + return val != oval; +} + +#define BURGUNDY_SWITCH_B(xname, xindex, addr, lmask, rmask, stereo) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ + .info = snd_pmac_burgundy_info_switch_b,\ + .get = snd_pmac_burgundy_get_switch_b,\ + .put = snd_pmac_burgundy_put_switch_b,\ + .private_value = ((lmask) | ((rmask) << 8)\ + | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) } + +/* + * Burgundy mixers + */ static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = { - BURGUNDY_VOLUME("Master Playback Volume", 0, MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), - BURGUNDY_VOLUME("Line Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLLINE, 16), - BURGUNDY_VOLUME("CD Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLCD, 16), - BURGUNDY_VOLUME("Mic Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLMIC, 16), - BURGUNDY_OUTPUT_VOLUME("PC Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENHP, 0), - /*BURGUNDY_OUTPUT_VOLUME("PCM Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1),*/ - BURGUNDY_OUTPUT_VOLUME("Headphone Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1), -}; -static struct snd_kcontrol_new snd_pmac_burgundy_master_sw __initdata = -BURGUNDY_OUTPUT_SWITCH("Headphone Playback Switch", 0, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw __initdata = -BURGUNDY_OUTPUT_SWITCH("PC Speaker Playback Switch", 0, BURGUNDY_OUTPUT_INTERN, 0, 0); + BURGUNDY_VOLUME_W("Master Playback Volume", 0, + MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), + BURGUNDY_VOLUME_W("CD Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLCD, 16), + BURGUNDY_VOLUME_2B("Input Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIX01, 2), + BURGUNDY_VOLUME_2B("Mixer Playback Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIX23, 0), + BURGUNDY_VOLUME_B("CD Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINCD, 1, 0), + BURGUNDY_SWITCH_W("Master Capture Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTENABLES, 24, 0, 0), + BURGUNDY_SWITCH_W("CD Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 0, 16, 1), + BURGUNDY_SWITCH_W("CD Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 0, 16, 1), +/* BURGUNDY_SWITCH_W("Loop Capture Switch", 0, + * MASK_ADDR_BURGUNDY_CAPTURESELECTS, 8, 24, 1), + * BURGUNDY_SWITCH_B("Mixer out Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x02, 0, 0), + * BURGUNDY_SWITCH_B("Mixer Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x01, 0, 0), + * BURGUNDY_SWITCH_B("PCM out Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFEH, 0x02, 0, 0), + */ BURGUNDY_SWITCH_B("PCM Capture Switch", 0, + MASK_ADDR_BURGUNDY_HOSTIFEH, 0x01, 0, 0) +}; +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = { + BURGUNDY_VOLUME_W("Line in Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLLINE, 16), + BURGUNDY_VOLUME_W("Mic Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIC, 16), + BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINLINE, 1, 0), + BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), + BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), + BURGUNDY_VOLUME_B("Line out Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1), + BURGUNDY_VOLUME_B("Headphone Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENHP, 1, 1), + BURGUNDY_SWITCH_W("Line in Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 1, 17, 1), + BURGUNDY_SWITCH_W("Mic Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1), + BURGUNDY_SWITCH_W("Line in Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 1, 17, 1), + BURGUNDY_SWITCH_W("Mic Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1), + BURGUNDY_SWITCH_B("Mic Boost Capture Switch", 0, + MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) +}; +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = { + BURGUNDY_VOLUME_W("Line in Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIC, 16), + BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), + BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1), + BURGUNDY_VOLUME_B("Line out Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), + BURGUNDY_SWITCH_W("Line in Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1), + BURGUNDY_SWITCH_W("Line in Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1), +/* BURGUNDY_SWITCH_B("Line in Boost Capture Switch", 0, + * MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) */ +}; +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __initdata = +BURGUNDY_SWITCH_B("Master Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT | BURGUNDY_LINEOUT_LEFT | BURGUNDY_HP_LEFT, + BURGUNDY_OUTPUT_RIGHT | BURGUNDY_LINEOUT_RIGHT | BURGUNDY_HP_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __initdata = +BURGUNDY_SWITCH_B("Master Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_INTERN + | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __initdata = +BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __initdata = +BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_INTERN, 0, 0); +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __initdata = +BURGUNDY_SWITCH_B("Line out Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_LINEOUT_LEFT, BURGUNDY_LINEOUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __initdata = +BURGUNDY_SWITCH_B("Line out Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __initdata = +BURGUNDY_SWITCH_B("Headphone Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_HP_LEFT, BURGUNDY_HP_RIGHT, 1); #ifdef PMAC_SUPPORT_AUTOMUTE @@ -350,16 +585,26 @@ static int snd_pmac_burgundy_detect_headphone(struct snd_pmac *chip) static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify) { if (chip->auto_mute) { + int imac = machine_is_compatible("iMac"); int reg, oreg; - reg = oreg = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - reg &= ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT | BURGUNDY_OUTPUT_INTERN); + reg = oreg = snd_pmac_burgundy_rcb(chip, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); + reg &= imac ? ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT + | BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT) + : ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT + | BURGUNDY_OUTPUT_INTERN); if (snd_pmac_burgundy_detect_headphone(chip)) - reg |= BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT; + reg |= imac ? (BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT) + : (BURGUNDY_OUTPUT_LEFT + | BURGUNDY_OUTPUT_RIGHT); else - reg |= BURGUNDY_OUTPUT_INTERN; + reg |= imac ? (BURGUNDY_OUTPUT_LEFT + | BURGUNDY_OUTPUT_RIGHT) + : (BURGUNDY_OUTPUT_INTERN); if (do_notify && reg == oreg) return; - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg); + snd_pmac_burgundy_wcb(chip, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg); if (do_notify) { snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_sw_ctl->id); @@ -378,6 +623,7 @@ static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_noti */ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) { + int imac = machine_is_compatible("iMac"); int i, err; /* Checks to see the chip is alive and kicking */ @@ -386,7 +632,7 @@ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) return 1; } - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES, + snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES, DEF_BURGUNDY_OUTPUTENABLES); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, DEF_BURGUNDY_MORE_OUTPUTENABLES); @@ -396,7 +642,8 @@ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL21, DEF_BURGUNDY_INPSEL21); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL3, - DEF_BURGUNDY_INPSEL3); + imac ? DEF_BURGUNDY_INPSEL3_IMAC + : DEF_BURGUNDY_INPSEL3_PMAC); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINCD, DEF_BURGUNDY_GAINCD); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINLINE, @@ -422,23 +669,55 @@ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_VOLMIC, DEF_BURGUNDY_VOLMIC); - if (chip->hp_stat_mask == 0) + if (chip->hp_stat_mask == 0) { /* set headphone-jack detection bit */ - chip->hp_stat_mask = 0x04; - + if (imac) + chip->hp_stat_mask = BURGUNDY_HPDETECT_IMAC_UPPER + | BURGUNDY_HPDETECT_IMAC_LOWER + | BURGUNDY_HPDETECT_IMAC_SIDE; + else + chip->hp_stat_mask = BURGUNDY_HPDETECT_PMAC_BACK; + } /* * build burgundy mixers */ strcpy(chip->card->mixername, "PowerMac Burgundy"); for (i = 0; i < ARRAY_SIZE(snd_pmac_burgundy_mixers); i++) { - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip)); + if (err < 0) return err; } - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_burgundy_master_sw, chip); + for (i = 0; i < (imac ? ARRAY_SIZE(snd_pmac_burgundy_mixers_imac) + : ARRAY_SIZE(snd_pmac_burgundy_mixers_pmac)); i++) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(imac ? &snd_pmac_burgundy_mixers_imac[i] + : &snd_pmac_burgundy_mixers_pmac[i], chip)); + if (err < 0) + return err; + } + chip->master_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_master_sw_imac + : &snd_pmac_burgundy_master_sw_pmac, chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_burgundy_speaker_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_line_sw_imac + : &snd_pmac_burgundy_line_sw_pmac, chip); + err = snd_ctl_add(chip->card, chip->master_sw_ctl); + if (err < 0) + return err; + if (imac) { + chip->master_sw_ctl = snd_ctl_new1( + &snd_pmac_burgundy_hp_sw_imac, chip); + err = snd_ctl_add(chip->card, chip->master_sw_ctl); + if (err < 0) + return err; + } + chip->speaker_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_speaker_sw_imac + : &snd_pmac_burgundy_speaker_sw_pmac, chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; #ifdef PMAC_SUPPORT_AUTOMUTE diff --git a/sound/ppc/burgundy.h b/sound/ppc/burgundy.h index ebb457a8342c..7a7f9cf3d299 100644 --- a/sound/ppc/burgundy.h +++ b/sound/ppc/burgundy.h @@ -22,6 +22,7 @@ #ifndef __BURGUNDY_H #define __BURGUNDY_H +#define MASK_ADDR_BURGUNDY_INPBOOST (0x10 << 12) #define MASK_ADDR_BURGUNDY_INPSEL21 (0x11 << 12) #define MASK_ADDR_BURGUNDY_INPSEL3 (0x12 << 12) @@ -35,7 +36,10 @@ #define MASK_ADDR_BURGUNDY_VOLCH3 (0x22 << 12) #define MASK_ADDR_BURGUNDY_VOLCH4 (0x23 << 12) +#define MASK_ADDR_BURGUNDY_CAPTURESELECTS (0x2A << 12) #define MASK_ADDR_BURGUNDY_OUTPUTSELECTS (0x2B << 12) +#define MASK_ADDR_BURGUNDY_VOLMIX01 (0x2D << 12) +#define MASK_ADDR_BURGUNDY_VOLMIX23 (0x2E << 12) #define MASK_ADDR_BURGUNDY_OUTPUTENABLES (0x2F << 12) #define MASK_ADDR_BURGUNDY_MASTER_VOLUME (0x30 << 12) @@ -45,6 +49,10 @@ #define MASK_ADDR_BURGUNDY_ATTENSPEAKER (0x62 << 12) #define MASK_ADDR_BURGUNDY_ATTENLINEOUT (0x63 << 12) #define MASK_ADDR_BURGUNDY_ATTENHP (0x64 << 12) +#define MASK_ADDR_BURGUNDY_ATTENMONO (0x65 << 12) + +#define MASK_ADDR_BURGUNDY_HOSTIFAD (0x78 << 12) +#define MASK_ADDR_BURGUNDY_HOSTIFEH (0x79 << 12) #define MASK_ADDR_BURGUNDY_VOLCD (MASK_ADDR_BURGUNDY_VOLCH1) #define MASK_ADDR_BURGUNDY_VOLLINE (MASK_ADDR_BURGUNDY_VOLCH2) @@ -59,21 +67,22 @@ /* These are all default values for the burgundy */ #define DEF_BURGUNDY_INPSEL21 (0xAA) -#define DEF_BURGUNDY_INPSEL3 (0x0A) +#define DEF_BURGUNDY_INPSEL3_IMAC (0x0A) +#define DEF_BURGUNDY_INPSEL3_PMAC (0x05) #define DEF_BURGUNDY_GAINCD (0x33) #define DEF_BURGUNDY_GAINLINE (0x44) #define DEF_BURGUNDY_GAINMIC (0x44) #define DEF_BURGUNDY_GAINMODEM (0x06) -/* Remember: lowest volume here is 0x9b */ +/* Remember: lowest volume here is 0x9B (155) */ #define DEF_BURGUNDY_VOLCD (0xCCCCCCCC) #define DEF_BURGUNDY_VOLLINE (0x00000000) #define DEF_BURGUNDY_VOLMIC (0x00000000) #define DEF_BURGUNDY_VOLMODEM (0xCCCCCCCC) -#define DEF_BURGUNDY_OUTPUTSELECTS (0x010f010f) -#define DEF_BURGUNDY_OUTPUTENABLES (0x0A) +#define DEF_BURGUNDY_OUTPUTSELECTS (0x010F010F) +#define DEF_BURGUNDY_OUTPUTENABLES (0x0100000A) /* #define DEF_BURGUNDY_MASTER_VOLUME (0xFFFFFFFF) */ /* too loud */ #define DEF_BURGUNDY_MASTER_VOLUME (0xDDDDDDDD) @@ -84,12 +93,22 @@ #define DEF_BURGUNDY_ATTENLINEOUT (0xCC) #define DEF_BURGUNDY_ATTENHP (0xCC) -/* OUTPUTENABLES bits */ +/* MORE_OUTPUTENABLES bits */ #define BURGUNDY_OUTPUT_LEFT 0x02 #define BURGUNDY_OUTPUT_RIGHT 0x04 +#define BURGUNDY_LINEOUT_LEFT 0x08 +#define BURGUNDY_LINEOUT_RIGHT 0x10 +#define BURGUNDY_HP_LEFT 0x20 +#define BURGUNDY_HP_RIGHT 0x40 #define BURGUNDY_OUTPUT_INTERN 0x80 -/* volume offset */ +/* Headphone detection bits */ +#define BURGUNDY_HPDETECT_PMAC_BACK 0x04 +#define BURGUNDY_HPDETECT_IMAC_SIDE 0x04 +#define BURGUNDY_HPDETECT_IMAC_UPPER 0x08 +#define BURGUNDY_HPDETECT_IMAC_LOWER 0x01 + +/* Volume offset */ #define BURGUNDY_VOLUME_OFFSET 155 #endif /* __BURGUNDY_H */ From 20861fa7b20a40ca045393df634d4d51e61efa58 Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Wed, 16 Apr 2008 19:45:51 +0200 Subject: [PATCH 217/250] [ALSA] snd-powermac: style burgundy.c Coding style corrections for burgundy.c. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/burgundy.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index f0c12a97fdbc..f860d39af36b 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -102,7 +102,8 @@ snd_pmac_burgundy_rcw(struct snd_pmac *chip, unsigned addr) } static void -snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr, unsigned int val) +snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr, + unsigned int val) { out_le32(&chip->awacs->codec_ctrl, addr + 0x300000 + (val & 0xff)); snd_pmac_burgundy_busy_wait(chip); @@ -126,8 +127,11 @@ snd_pmac_burgundy_rcb(struct snd_pmac *chip, unsigned int addr) return val; } +#define BASE2ADDR(base) ((base) << 12) +#define ADDR2BASE(addr) ((addr) >> 12) + /* - * Burgundy volume: 0 - 100, stereo + * Burgundy volume: 0 - 100, stereo, word reg */ static void snd_pmac_burgundy_write_volume(struct snd_pmac *chip, unsigned int address, @@ -168,13 +172,6 @@ snd_pmac_burgundy_read_volume(struct snd_pmac *chip, unsigned int address, volume[1] = 0; } - -/* - */ - -#define BASE2ADDR(base) ((base) << 12) -#define ADDR2BASE(addr) ((addr) >> 12) - static int snd_pmac_burgundy_info_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -191,8 +188,8 @@ static int snd_pmac_burgundy_get_volume(struct snd_kcontrol *kcontrol, struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); int shift = (kcontrol->private_value >> 8) & 0xff; - snd_pmac_burgundy_read_volume(chip, addr, ucontrol->value.integer.value, - shift); + snd_pmac_burgundy_read_volume(chip, addr, + ucontrol->value.integer.value, shift); return 0; } @@ -204,8 +201,8 @@ static int snd_pmac_burgundy_put_volume(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0xff; long nvoices[2]; - snd_pmac_burgundy_write_volume(chip, addr, ucontrol->value.integer.value, - shift); + snd_pmac_burgundy_write_volume(chip, addr, + ucontrol->value.integer.value, shift); snd_pmac_burgundy_read_volume(chip, addr, nvoices, shift); return (nvoices[0] != ucontrol->value.integer.value[0] || nvoices[1] != ucontrol->value.integer.value[1]); @@ -700,7 +697,8 @@ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) chip->master_sw_ctl = snd_ctl_new1(imac ? &snd_pmac_burgundy_master_sw_imac : &snd_pmac_burgundy_master_sw_pmac, chip); - if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + err = snd_ctl_add(chip->card, chip->master_sw_ctl); + if (err < 0) return err; chip->master_sw_ctl = snd_ctl_new1(imac ? &snd_pmac_burgundy_line_sw_imac @@ -718,10 +716,12 @@ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) chip->speaker_sw_ctl = snd_ctl_new1(imac ? &snd_pmac_burgundy_speaker_sw_imac : &snd_pmac_burgundy_speaker_sw_pmac, chip); - if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) + err = snd_ctl_add(chip->card, chip->speaker_sw_ctl); + if (err < 0) return err; #ifdef PMAC_SUPPORT_AUTOMUTE - if ((err = snd_pmac_add_automute(chip)) < 0) + err = snd_pmac_add_automute(chip); + if (err < 0) return err; chip->detect_headphone = snd_pmac_burgundy_detect_headphone; From 73d38b13ffb105ab633bd91969c8d218b2de38d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Apr 2008 12:50:47 +0200 Subject: [PATCH 218/250] [ALSA] Fix the race of card instance unregistration Move the call of device_unregister() for the card instance in snd_card_disconnect() to avoid the race of sysfs card entry, which can be typically found on usb-audio reconnection. Signed-off-by: Takashi Iwai --- sound/core/init.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index f045f7db3ab4..07acac77ea45 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -360,6 +360,12 @@ int snd_card_disconnect(struct snd_card *card) snd_printk(KERN_ERR "not all devices for card %i can be disconnected\n", card->number); snd_info_card_disconnect(card); +#ifndef CONFIG_SYSFS_DEPRECATED + if (card->card_dev) { + device_unregister(card->card_dev); + card->card_dev = NULL; + } +#endif return 0; } @@ -401,10 +407,6 @@ static int snd_card_do_free(struct snd_card *card) snd_printk(KERN_WARNING "unable to free card info\n"); /* Not fatal error */ } -#ifndef CONFIG_SYSFS_DEPRECATED - if (card->card_dev) - device_unregister(card->card_dev); -#endif kfree(card); return 0; } From f18638dcf0c481eca2430206ebcdc7295aec8623 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Apr 2008 12:52:02 +0200 Subject: [PATCH 219/250] [ALSA] Clean up snd_card_free*() A little clean up of snd_card_free*(). Removed snd_card_free_prepare() since it's actually almost identical with snd_card_disconnect(). Signed-off-by: Takashi Iwai --- sound/core/init.c | 26 +++++++++----------------- 1 file changed, 9 insertions(+), 17 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 07acac77ea45..ac0573416130 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -311,6 +311,9 @@ int snd_card_disconnect(struct snd_card *card) struct file *file; int err; + if (!card) + return -EINVAL; + spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); @@ -322,6 +325,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 1: disable fops (user space) operations for ALSA API */ mutex_lock(&snd_card_mutex); snd_cards[card->number] = NULL; + snd_cards_lock &= ~(1 << card->number); mutex_unlock(&snd_card_mutex); /* phase 2: replace file->f_op with special dummy operations */ @@ -365,6 +369,9 @@ int snd_card_disconnect(struct snd_card *card) device_unregister(card->card_dev); card->card_dev = NULL; } +#endif +#ifdef CONFIG_PM + wake_up(&card->power_sleep); #endif return 0; } @@ -411,25 +418,10 @@ static int snd_card_do_free(struct snd_card *card) return 0; } -static int snd_card_free_prepare(struct snd_card *card) -{ - if (card == NULL) - return -EINVAL; - (void) snd_card_disconnect(card); - mutex_lock(&snd_card_mutex); - snd_cards[card->number] = NULL; - snd_cards_lock &= ~(1 << card->number); - mutex_unlock(&snd_card_mutex); -#ifdef CONFIG_PM - wake_up(&card->power_sleep); -#endif - return 0; -} - int snd_card_free_when_closed(struct snd_card *card) { int free_now = 0; - int ret = snd_card_free_prepare(card); + int ret = snd_card_disconnect(card); if (ret) return ret; @@ -449,7 +441,7 @@ EXPORT_SYMBOL(snd_card_free_when_closed); int snd_card_free(struct snd_card *card) { - int ret = snd_card_free_prepare(card); + int ret = snd_card_disconnect(card); if (ret) return ret; From 9eb70e68f38bbc5996a2193e7b7dc0b5487a08cb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Apr 2008 12:53:26 +0200 Subject: [PATCH 220/250] [ALSA] usb-audio - Fix race in reconnection Fix the race at reconnection of the device. The disconnected usb_chip[] must be cleared before the next probe call properly. Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 7b3bcf1916d2..410be4aff1ba 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3424,7 +3424,6 @@ static void snd_usb_audio_create_proc(struct snd_usb_audio *chip) static int snd_usb_audio_free(struct snd_usb_audio *chip) { - usb_chip[chip->index] = NULL; kfree(chip); return 0; } @@ -3689,6 +3688,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) list_for_each(p, &chip->mixer_list) { snd_usb_mixer_disconnect(p); } + usb_chip[chip->index] = NULL; mutex_unlock(®ister_mutex); snd_card_free_when_closed(card); } else { From e34ba212225a27cdf5f974be22cc539ae7ee7ca5 Mon Sep 17 00:00:00 2001 From: Roel Kluin <12o3l@tiscali.nl> Date: Thu, 17 Apr 2008 18:58:34 +0200 Subject: [PATCH 221/250] [ALSA] SOC: fix tests in cs4270_hw_params() cs4270_hw_params does several times: ret = snd_soc_write() if (ret < 0) ... This only works when ret is signed. Signed-off-by: Roel Kluin <12o3l@tiscali.nl> Signed-off-by: Takashi Iwai --- sound/soc/codecs/cs4270.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index bf2ab72d49bf..e73fcfd9f5cd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -372,7 +372,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct cs4270_private *cs4270 = codec->private_data; - unsigned int ret = 0; + int ret; unsigned int i; unsigned int rate; unsigned int ratio; From 0d52cea487c0213d6d7bca9c951210081e7b653b Mon Sep 17 00:00:00 2001 From: Roel Kluin <12o3l@tiscali.nl> Date: Fri, 18 Apr 2008 12:25:41 +0200 Subject: [PATCH 222/250] OSS: dmabuf: fix negative DMAbuf_get_buffer_pointer() check Since unsigned active_offs < 0 is even true when DMAbuf_get_buffer_pointer() returns negative Signed-off-by: Roel Kluin <12o3l@tiscali.nl> Signed-off-by: Takashi Iwai --- sound/oss/dmabuf.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index eaf69971bf92..1e90d769b62e 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -795,9 +795,9 @@ static int find_output_space(int dev, char **buf, int *size) #ifdef BE_CONSERVATIVE active_offs = dmap->byte_counter + dmap->qhead * dmap->fragment_size; #else - active_offs = DMAbuf_get_buffer_pointer(dev, dmap, DMODE_OUTPUT); + active_offs = max(DMAbuf_get_buffer_pointer(dev, dmap, DMODE_OUTPUT), 0); /* Check for pointer wrapping situation */ - if (active_offs < 0 || active_offs >= dmap->bytes_in_use) + if (active_offs >= dmap->bytes_in_use) active_offs = 0; active_offs += dmap->byte_counter; #endif From 988f0664779674c7c06252a6d549eee8debd5d76 Mon Sep 17 00:00:00 2001 From: Karsten Wiese Date: Tue, 22 Apr 2008 12:52:15 +0200 Subject: [PATCH 223/250] [ALSA] ice1724.c: toggle "chip reset" and "eeprom based setup" sequence Let "chip reset" become first. Increasement of the "chip reset" related timeout leads to correctly read eeprom's contents here. Signed-off-by: Karsten Wiese Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ceac87056263..6b15e1cbfe4f 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2045,12 +2045,16 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, -static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) +static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) { outb(VT1724_RESET , ICEREG1724(ice, CONTROL)); - udelay(200); + msleep(10); outb(0, ICEREG1724(ice, CONTROL)); - udelay(200); + msleep(10); +} + +static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) +{ outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG)); outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); outb(ice->eeprom.data[ICE_EEP2_I2S], ICEREG1724(ice, I2S_FEATURES)); @@ -2223,6 +2227,7 @@ static int __devinit snd_vt1724_create(struct snd_card *card, ice->irq = pci->irq; + snd_vt1724_chip_reset(ice); if (snd_vt1724_read_eeprom(ice, modelname) < 0) { snd_vt1724_free(ice); return -EIO; From 775c199e6af5e4212bfa7ebeadee09563c14694b Mon Sep 17 00:00:00 2001 From: Karsten Wiese Date: Tue, 22 Apr 2008 12:52:45 +0200 Subject: [PATCH 224/250] [ALSA] Don't set gpio mask register in snd_ice1712_gpio_write_bits() Some calls to snd_ice1712_gpio_write() go wrong, if snd_ice1712_gpio_write_bits() ran before and changed the gpio mask register. Read the actual gpio value and combine it with the to be set bits in the cpu instead. Signed-off-by: Karsten Wiese Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index a3bea2247c7f..3208901c740e 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -438,10 +438,14 @@ int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu static inline void snd_ice1712_gpio_write_bits(struct snd_ice1712 *ice, unsigned int mask, unsigned int bits) { + unsigned val; + ice->gpio.direction |= mask; snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); - snd_ice1712_gpio_set_mask(ice, ~mask); - snd_ice1712_gpio_write(ice, mask & bits); + val = snd_ice1712_gpio_read(ice); + val &= ~mask; + val |= mask & bits; + snd_ice1712_gpio_write(ice, val); } static inline int snd_ice1712_gpio_read_bits(struct snd_ice1712 *ice, From 8a87c9cf999542db846c3ab93c1065d446427f37 Mon Sep 17 00:00:00 2001 From: Karsten Wiese Date: Tue, 22 Apr 2008 12:53:12 +0200 Subject: [PATCH 225/250] [ALSA] Audiophile 192: Fix ad converter initialization Correct some arguments in calls to snd_ice1712_gpio_write_bits() from ap192_set_rate_val(). Signed-off-by: Karsten Wiese Signed-off-by: Takashi Iwai --- sound/pci/ice1712/revo.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index dba93d8efbe0..4d2631434dc8 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -327,7 +327,7 @@ static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) revo_set_rate_val(ak, rate); /* reset CKS */ - snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 : 0); + snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 << 8 : 0); /* reset DFS pins of AK5385A for ADC, too */ if (rate > 96000) dfs = 2; @@ -338,7 +338,7 @@ static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) snd_ice1712_gpio_write_bits(ice, 3 << 9, dfs << 9); /* reset ADC */ snd_ice1712_gpio_write_bits(ice, 1 << 11, 0); - snd_ice1712_gpio_write_bits(ice, 1 << 11, 1); + snd_ice1712_gpio_write_bits(ice, 1 << 11, 1 << 11); } static const struct snd_akm4xxx_dac_channel ap192_dac[] = { From f000fd80937c0d94c67f9f3e7026f1fbc8ef8873 Mon Sep 17 00:00:00 2001 From: Jeff Garzik Date: Tue, 22 Apr 2008 13:50:34 +0200 Subject: [PATCH 226/250] [ALSA] Fix synchronize_irq() bugs, redundancies free_irq() calls synchronize_irq() for you, so there is no need for drivers to manually do the same thing (again). Thus, calls where sync-irq immediately precedes free-irq can be simplified. However, during this audit several bugs were noticed, where free-irq is preceded by a "irq >= 0" check... but the sync-irq call is not covered by the same check. So, where sync-irq could not be eliminated completely, the missing check was added. Signed-off-by: Jeff Garzik Signed-off-by: Takashi Iwai --- drivers/media/video/cx88/cx88-alsa.c | 4 +--- drivers/media/video/saa7134/saa7134-alsa.c | 4 +--- sound/pci/ad1889.c | 2 -- sound/pci/ali5451/ali5451.c | 4 +--- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.c | 2 -- sound/pci/azt3328.c | 3 ++- sound/pci/cmipci.c | 2 -- sound/pci/ens1370.c | 3 ++- sound/pci/es1938.c | 5 +---- sound/pci/es1968.c | 3 ++- sound/pci/hda/hda_intel.c | 5 +---- sound/pci/ice1712/ice1712.c | 5 ++--- sound/pci/ice1712/ice1724.c | 4 +--- sound/pci/intel8x0.c | 3 +-- sound/pci/intel8x0m.c | 3 ++- sound/pci/maestro3.c | 4 +--- sound/pci/oxygen/oxygen_lib.c | 4 +--- sound/pci/sis7019.c | 1 - sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- 22 files changed, 23 insertions(+), 46 deletions(-) diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c index 316b106c3511..03feb5b49e1b 100644 --- a/drivers/media/video/cx88/cx88-alsa.c +++ b/drivers/media/video/cx88/cx88-alsa.c @@ -690,10 +690,8 @@ MODULE_DEVICE_TABLE(pci, cx88_audio_pci_tbl); static int snd_cx88_free(snd_cx88_card_t *chip) { - if (chip->irq >= 0){ - synchronize_irq(chip->irq); + if (chip->irq >= 0) free_irq(chip->irq, chip); - } cx88_core_put(chip->core,chip->pci); diff --git a/drivers/media/video/saa7134/saa7134-alsa.c b/drivers/media/video/saa7134/saa7134-alsa.c index 047add8f3010..ec6bdb9680a3 100644 --- a/drivers/media/video/saa7134/saa7134-alsa.c +++ b/drivers/media/video/saa7134/saa7134-alsa.c @@ -954,10 +954,8 @@ static void snd_saa7134_free(struct snd_card * card) if (chip->dev->dmasound.priv_data == NULL) return; - if (chip->irq >= 0) { - synchronize_irq(chip->irq); + if (chip->irq >= 0) free_irq(chip->irq, &chip->dev->dmasound); - } chip->dev->dmasound.priv_data = NULL; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 1edb6448946d..39ec55b57b1e 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -854,8 +854,6 @@ snd_ad1889_free(struct snd_ad1889 *chip) spin_unlock_irq(&chip->lock); - synchronize_irq(chip->irq); - if (chip->irq >= 0) free_irq(chip->irq, chip); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index fc04d3da1af5..1a0fd65ec280 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2047,10 +2047,8 @@ static int snd_ali_free(struct snd_ali * codec) { if (codec->hw_initialized) snd_ali_disable_address_interrupt(codec); - if (codec->irq >= 0) { - synchronize_irq(codec->irq); + if (codec->irq >= 0) free_irq(codec->irq, codec); - } if (codec->port) pci_release_regions(codec->pci); pci_disable_device(codec->pci); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 4594186b83ee..457228fb22aa 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1553,7 +1553,7 @@ static int snd_atiixp_free(struct atiixp *chip) if (chip->irq < 0) goto __hw_end; snd_atiixp_chip_stop(chip); - synchronize_irq(chip->irq); + __hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index a67a869180d4..d457a32a7939 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1197,7 +1197,7 @@ static int snd_atiixp_free(struct atiixp_modem *chip) if (chip->irq < 0) goto __hw_end; snd_atiixp_chip_stop(chip); - synchronize_irq(chip->irq); + __hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 26819e2f5761..68368e490074 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -126,7 +126,6 @@ static int snd_vortex_dev_free(struct snd_device *device) vortex_gameport_unregister(vortex); vortex_core_shutdown(vortex); // Take down PCI interface. - synchronize_irq(vortex->irq); free_irq(vortex->irq, vortex); iounmap(vortex->mmio); pci_release_regions(vortex->pci_dev); @@ -220,7 +219,6 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) return 0; alloc_out: - synchronize_irq(chip->irq); free_irq(chip->irq, chip); irq_out: vortex_core_shutdown(chip); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index be87d3113ee4..5f63af6b88a2 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1514,7 +1514,8 @@ snd_azf3328_free(struct snd_azf3328 *chip) /* well, at least we know how to disable the timer IRQ */ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); - synchronize_irq(chip->irq); + if (chip->irq >= 0) + synchronize_irq(chip->irq); __end_hw: snd_azf3328_free_joystick(chip); if (chip->irq >= 0) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 4074584a7d90..9971b5b7735b 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2933,8 +2933,6 @@ static int snd_cmipci_free(struct cmipci *cm) /* reset mixer */ snd_cmipci_mixer_write(cm, 0, 0); - synchronize_irq(cm->irq); - free_irq(cm->irq, cm); } diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 52fae4a7cfdd..fbf1124f7c79 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1910,7 +1910,8 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) outl(0, ES_REG(ensoniq, CONTROL)); /* switch everything off */ outl(0, ES_REG(ensoniq, SERIAL)); /* clear serial interface */ #endif - synchronize_irq(ensoniq->irq); + if (ensoniq->irq >= 0) + synchronize_irq(ensoniq->irq); pci_set_power_state(ensoniq->pci, 3); __hw_end: #ifdef CHIP1370 diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 1a314fa99c45..84fac1fbf103 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1488,7 +1488,6 @@ static int es1938_suspend(struct pci_dev *pci, pm_message_t state) outb(0x00, SLIO_REG(chip, IRQCONTROL)); /* disable irqs */ if (chip->irq >= 0) { - synchronize_irq(chip->irq); free_irq(chip->irq, chip); chip->irq = -1; } @@ -1578,10 +1577,8 @@ static int snd_es1938_free(struct es1938 *chip) snd_es1938_free_gameport(chip); - if (chip->irq >= 0) { - synchronize_irq(chip->irq); + if (chip->irq >= 0) free_irq(chip->irq, chip); - } pci_release_regions(chip->pci); pci_disable_device(chip->pci); kfree(chip); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 138379896061..1bf298d214b9 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2475,7 +2475,8 @@ static inline void snd_es1968_free_gameport(struct es1968 *chip) { } static int snd_es1968_free(struct es1968 *chip) { if (chip->io_port) { - synchronize_irq(chip->irq); + if (chip->irq >= 0) + synchronize_irq(chip->irq); outw(1, chip->io_port + 0x04); /* clear WP interrupts */ outw(0, chip->io_port + ESM_PORT_HOST_IRQ); /* disable IRQ */ } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bc3867e1945c..b3a618eb42cd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1682,7 +1682,6 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) snd_hda_suspend(chip->bus, state); azx_stop_chip(chip); if (chip->irq >= 0) { - synchronize_irq(chip->irq); free_irq(chip->irq, chip); chip->irq = -1; } @@ -1738,10 +1737,8 @@ static int azx_free(struct azx *chip) azx_stop_chip(chip); } - if (chip->irq >= 0) { - synchronize_irq(chip->irq); + if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); - } if (chip->msi) pci_disable_msi(chip->pci); if (chip->remap_addr) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 38e93ca12e27..29d449d73c98 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2490,10 +2490,9 @@ static int snd_ice1712_free(struct snd_ice1712 *ice) outb(0xff, ICEREG(ice, IRQMASK)); /* --- */ __hw_end: - if (ice->irq >= 0) { - synchronize_irq(ice->irq); + if (ice->irq >= 0) free_irq(ice->irq, ice); - } + if (ice->port) pci_release_regions(ice->pci); snd_ice1712_akm4xxx_free(ice); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6b15e1cbfe4f..13ea94f317cb 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2153,10 +2153,8 @@ static int snd_vt1724_free(struct snd_ice1712 *ice) outb(0xff, ICEREG1724(ice, IRQMASK)); /* --- */ __hw_end: - if (ice->irq >= 0) { - synchronize_irq(ice->irq); + if (ice->irq >= 0) free_irq(ice->irq, ice); - } pci_release_regions(ice->pci); snd_ice1712_akm4xxx_free(ice); pci_disable_device(ice->pci); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 47485afcab5f..048d99e25ab0 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2468,7 +2468,7 @@ static int snd_intel8x0_free(struct intel8x0 *chip) pci_write_config_dword(chip->pci, 0x4c, val); } /* --- */ - synchronize_irq(chip->irq); + __hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -2517,7 +2517,6 @@ static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state) chip->sdm_saved = igetbyte(chip, ICHREG(SDM)); if (chip->irq >= 0) { - synchronize_irq(chip->irq); free_irq(chip->irq, chip); chip->irq = -1; } diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index cadda8d6b70f..15db810d5893 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -986,7 +986,8 @@ static int snd_intel8x0_free(struct intel8x0m *chip) for (i = 0; i < chip->bdbars_count; i++) iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS); /* --- */ - synchronize_irq(chip->irq); + if (chip->irq >= 0) + synchronize_irq(chip->irq); __hw_end: if (chip->bdbars.area) snd_dma_free_pages(&chip->bdbars); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index a753dae65ab6..a536c59fbea1 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2542,10 +2542,8 @@ static int snd_m3_free(struct snd_m3 *chip) vfree(chip->suspend_mem); #endif - if (chip->irq >= 0) { - synchronize_irq(chip->irq); + if (chip->irq >= 0) free_irq(chip->irq, chip); - } if (chip->iobase) pci_release_regions(chip->pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index f84f6a128ee9..897697d43506 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -410,10 +410,8 @@ static void oxygen_card_free(struct snd_card *card) oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); spin_unlock_irq(&chip->reg_lock); - if (chip->irq >= 0) { + if (chip->irq >= 0) free_irq(chip->irq, chip); - synchronize_irq(chip->irq); - } flush_scheduled_work(); chip->model->cleanup(chip); mutex_destroy(&chip->mutex); diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 742f1180c39e..df2007e3be7c 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1194,7 +1194,6 @@ static int sis_suspend(struct pci_dev *pci, pm_message_t state) /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. */ if (sis->irq >= 0) { - synchronize_irq(sis->irq); free_irq(sis->irq, sis); sis->irq = -1; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index a756be661f9a..b585cc3e4c47 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2236,7 +2236,7 @@ static int snd_via82xx_free(struct via82xx *chip) /* disable interrupts */ for (i = 0; i < chip->num_devs; i++) snd_via82xx_channel_reset(chip, &chip->devs[i]); - synchronize_irq(chip->irq); + if (chip->irq >= 0) free_irq(chip->irq, chip); __end_hw: diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index f5df1c79bee1..31f64ee39882 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1075,7 +1075,7 @@ static int snd_via82xx_free(struct via82xx_modem *chip) /* disable interrupts */ for (i = 0; i < chip->num_devs; i++) snd_via82xx_channel_reset(chip, &chip->devs[i]); - synchronize_irq(chip->irq); + __end_hw: if (chip->irq >= 0) free_irq(chip->irq, chip); From d80fd0935e2c177ae58d85cb736684ff6c00314d Mon Sep 17 00:00:00 2001 From: Peter Lienig Date: Tue, 22 Apr 2008 17:05:07 +0200 Subject: [PATCH 227/250] [ALSA] ice1712 - Add Terrasoniq TS88 support Added the support of Terrasonq TS88. Signed-off-by: Peter Lienig Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ews.c | 15 +++++++++++++++ sound/pci/ice1712/ews.h | 4 +++- 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 064760d2a027..013fc4f04822 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -238,6 +238,7 @@ static void snd_ice1712_ews_cs8404_spdif_write(struct snd_ice1712 *ice, unsigned case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_CS8404], &bits, 1) != 1) goto _error; @@ -433,6 +434,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: ice->num_total_dacs = 8; ice->num_total_adcs = 8; break; @@ -475,6 +477,8 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: + err = snd_i2c_device_create(ice->i2c, "CS8404", ICE1712_EWS88MT_CS8404_ADDR, &spec->i2cdevs[EWS_I2C_CS8404]); @@ -518,6 +522,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: case ICE1712_SUBDEVICE_EWS88D: /* set up CS8404 */ ice->spdif.ops.open = ews88_open_spdif; @@ -547,6 +552,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: err = snd_ice1712_akm4xxx_init(ak, &akm_ews88mt, &akm_ews88mt_priv, ice); break; case ICE1712_SUBDEVICE_EWX2496: @@ -973,6 +979,7 @@ static int __devinit snd_ice1712_ews_add_controls(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: case ICE1712_SUBDEVICE_DMX6FIRE: err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) @@ -992,6 +999,7 @@ static int __devinit snd_ice1712_ews_add_controls(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: + case ICE1712_SUBDEVICE_TS88: err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_ews88mt_input_sense, ice)); if (err < 0) return err; @@ -1048,6 +1056,13 @@ struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = { .chip_init = snd_ice1712_ews_init, .build_controls = snd_ice1712_ews_add_controls, }, + { + .subvendor = ICE1712_SUBDEVICE_TS88, + .name = "terrasoniq TS88", + .model = "phase88", + .chip_init = snd_ice1712_ews_init, + .build_controls = snd_ice1712_ews_add_controls, + }, { .subvendor = ICE1712_SUBDEVICE_EWS88D, .name = "TerraTec EWS88D", diff --git a/sound/pci/ice1712/ews.h b/sound/pci/ice1712/ews.h index e4ed1b475b08..1c443718af03 100644 --- a/sound/pci/ice1712/ews.h +++ b/sound/pci/ice1712/ews.h @@ -30,7 +30,8 @@ "{TerraTec,EWS 88MT},"\ "{TerraTec,EWS 88D},"\ "{TerraTec,DMX 6Fire},"\ - "{TerraTec,Phase 88}," + "{TerraTec,Phase 88}," \ + "{terrasoniq,TS 88}," #define ICE1712_SUBDEVICE_EWX2496 0x3b153011 #define ICE1712_SUBDEVICE_EWS88MT 0x3b151511 @@ -38,6 +39,7 @@ #define ICE1712_SUBDEVICE_EWS88D 0x3b152b11 #define ICE1712_SUBDEVICE_DMX6FIRE 0x3b153811 #define ICE1712_SUBDEVICE_PHASE88 0x3b155111 +#define ICE1712_SUBDEVICE_TS88 0x3b157c11 /* entry point */ extern struct snd_ice1712_card_info snd_ice1712_ews_cards[]; From 7a22323b231fe5d47804f98f31a70eb34c6104a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Apr 2008 17:08:52 +0200 Subject: [PATCH 228/250] [ALSA] soc - Support PXA3xx AC97 The PXA3xx does not support the use of interrupts during reset and access to the GPIO status requires similar handling to that for PXA27x. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/pxa2xx-ac97.c | 29 +++++++++++++++++++++++++++-- 1 file changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1092d58e8523..97ec2d90547c 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -61,7 +61,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -111,7 +111,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -134,6 +134,9 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 100; +#endif gsr_bits = 0; #ifdef CONFIG_PXA27x @@ -144,6 +147,11 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) GCR |= GCR_WARM_RST; pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); udelay(500); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts */ + GCR |= GCR_WARM_RST; + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); #else GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); @@ -159,6 +167,16 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 1000; + + /* Hold CLKBPB for 100us */ + GCR = 0; + GCR = GCR_CLKBPB; + udelay(100); + GCR = 0; +#endif + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -170,6 +188,13 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts on PXA3xx */ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + + GCR = GCR_WARM_RST | GCR_COLD_RST; + while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(10); #else GCR = GCR_COLD_RST; GCR |= GCR_CDONE_IE|GCR_SDONE_IE; From 815c1be320fd51e5981c007f737aca410707baf8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Apr 2008 17:09:49 +0200 Subject: [PATCH 229/250] [ALSA] pxa2xx-ac97: Support PXA3xx AC97 Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 71fbf8d7ee82..5b3274b465eb 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -112,6 +112,16 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigne static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) { /* First, try cold reset */ +#ifdef CONFIG_PXA3xx + int timeout; + + /* Hold CLKBPB for 100us */ + GCR = 0; + GCR = GCR_CLKBPB; + udelay(100); + GCR = 0; +#endif + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -123,6 +133,14 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); +#elif defined(CONFIG_PXA3xx) + timeout = 1000; + /* Can't use interrupts on PXA3xx */ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + + GCR = GCR_WARM_RST | GCR_COLD_RST; + while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(10); #else GCR = GCR_COLD_RST; GCR |= GCR_CDONE_IE|GCR_SDONE_IE; @@ -143,6 +161,12 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) GCR |= GCR_WARM_RST; pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); udelay(500); +#elif defined(CONFIG_PXA3xx) + timeout = 100; + /* Can't use interrupts */ + GCR |= GCR_WARM_RST; + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); #else GCR |= GCR_WARM_RST|GCR_PRIRDY_IEN|GCR_SECRDY_IEN; wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); From 6b9a9b329640b7e8143df7b2782884ea758650f7 Mon Sep 17 00:00:00 2001 From: Tim Niemeyer Date: Tue, 22 Apr 2008 17:10:23 +0200 Subject: [PATCH 230/250] [ALSA] soc - neo1973_wm8753 - Fix module unload Signed-off-by: Tim Niemeyer Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/neo1973_wm8753.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 6ee115ceb011..962cc20b1af5 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -659,6 +659,7 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { + i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } From ebf029da38829ede6b53ac8a5ad45b149064ea16 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Apr 2008 17:28:11 +0200 Subject: [PATCH 231/250] [ALSA] Fix possible races at free_irq in PCI drivers The irq handler of PCI drivers must be released before releasing other resources since the handler for a shared irq can be still called and may access the freed resource again. Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 5 ++--- sound/pci/cs46xx/cs46xx_lib.c | 6 +++--- sound/pci/echoaudio/echoaudio.c | 7 ++++--- sound/pci/emu10k1/emu10k1_main.c | 15 ++++++++------- sound/pci/emu10k1/emu10k1x.c | 8 ++++---- sound/pci/intel8x0m.c | 8 ++------ sound/pci/korg1212/korg1212.c | 1 - sound/pci/nm256/nm256.c | 4 +--- sound/pci/trident/trident_main.c | 4 ++-- 9 files changed, 26 insertions(+), 32 deletions(-) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 3818249fcc81..ecbe79b67e43 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1114,6 +1114,8 @@ static int snd_ca0106_free(struct snd_ca0106 *chip) * So we can fix: snd-malloc: Memory leak? pages not freed = 8 */ } + if (chip->irq >= 0) + free_irq(chip->irq, chip); // release the data #if 1 if (chip->buffer.area) @@ -1123,9 +1125,6 @@ static int snd_ca0106_free(struct snd_ca0106 *chip) // release the i/o port release_and_free_resource(chip->res_port); - // release the irq - if (chip->irq >= 0) - free_irq(chip->irq, chip); pci_disable_device(chip->pci); kfree(chip); return 0; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 87ddffcd9d89..e214e567dec8 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2772,6 +2772,9 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) if (chip->irq >= 0) free_irq(chip->irq, chip); + if (chip->active_ctrl) + chip->active_ctrl(chip, -chip->amplifier); + for (idx = 0; idx < 5; idx++) { struct snd_cs46xx_region *region = &chip->region.idx[idx]; if (region->remap_addr) @@ -2779,9 +2782,6 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) release_and_free_resource(region->resource); } - if (chip->active_ctrl) - chip->active_ctrl(chip, -chip->amplifier); - #ifdef CONFIG_SND_CS46XX_NEW_DSP if (chip->dsp_spos_instance) { cs46xx_dsp_spos_destroy(chip); diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 90ec090792ba..e16dc92e82fb 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1852,15 +1852,16 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { DE_INIT(("Stop DSP...\n")); - if (chip->comm_page) { + if (chip->comm_page) rest_in_peace(chip); - snd_dma_free_pages(&chip->commpage_dma_buf); - } DE_INIT(("Stopped.\n")); if (chip->irq >= 0) free_irq(chip->irq, chip); + if (chip->comm_page) + snd_dma_free_pages(&chip->commpage_dma_buf); + if (chip->dsp_registers) iounmap(chip->dsp_registers); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 9a9b977d3cf1..abde5b901884 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1249,11 +1249,6 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) if (emu->port) { /* avoid access to already used hardware */ snd_emu10k1_fx8010_tram_setup(emu, 0); snd_emu10k1_done(emu); - /* remove reserved page */ - if (emu->reserved_page) { - snd_emu10k1_synth_free(emu, (struct snd_util_memblk *)emu->reserved_page); - emu->reserved_page = NULL; - } snd_emu10k1_free_efx(emu); } if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { @@ -1262,6 +1257,14 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); + if (emu->irq >= 0) + free_irq(emu->irq, emu); + /* remove reserved page */ + if (emu->reserved_page) { + snd_emu10k1_synth_free(emu, + (struct snd_util_memblk *)emu->reserved_page); + emu->reserved_page = NULL; + } if (emu->memhdr) snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) @@ -1273,8 +1276,6 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) #ifdef CONFIG_PM free_pm_buffer(emu); #endif - if (emu->irq >= 0) - free_irq(emu->irq, emu); if (emu->port) pci_release_regions(emu->pci); if (emu->card_capabilities->ca0151_chip) /* P16V */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 341f34e19f3c..491a4a50f869 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -754,13 +754,13 @@ static int snd_emu10k1x_free(struct emu10k1x *chip) // disable audio outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); - // release the i/o port - release_and_free_resource(chip->res_port); - - // release the irq + /* release the irq */ if (chip->irq >= 0) free_irq(chip->irq, chip); + // release the i/o port + release_and_free_resource(chip->res_port); + // release the DMA if (chip->dma_buffer.area) { snd_dma_free_pages(&chip->dma_buffer); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 15db810d5893..faf674e671ac 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -985,18 +985,15 @@ static int snd_intel8x0_free(struct intel8x0m *chip) /* reset channels */ for (i = 0; i < chip->bdbars_count; i++) iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS); - /* --- */ + __hw_end: if (chip->irq >= 0) - synchronize_irq(chip->irq); - __hw_end: + free_irq(chip->irq, chip); if (chip->bdbars.area) snd_dma_free_pages(&chip->bdbars); if (chip->addr) pci_iounmap(chip->pci, chip->addr); if (chip->bmaddr) pci_iounmap(chip->pci, chip->bmaddr); - if (chip->irq >= 0) - free_irq(chip->irq, chip); pci_release_regions(chip->pci); pci_disable_device(chip->pci); kfree(chip); @@ -1018,7 +1015,6 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state) snd_pcm_suspend_all(chip->pcm[i]); snd_ac97_suspend(chip->ac97); if (chip->irq >= 0) { - synchronize_irq(chip->irq); free_irq(chip->irq, chip); chip->irq = -1; } diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 10c713d9ac49..f4c85b52bde3 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2102,7 +2102,6 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) snd_korg1212_TurnOffIdleMonitor(korg1212); if (korg1212->irq >= 0) { - synchronize_irq(korg1212->irq); snd_korg1212_DisableCardInterrupts(korg1212); free_irq(korg1212->irq, korg1212); korg1212->irq = -1; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 7ac654e381da..7efb838d18a6 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1439,7 +1439,7 @@ static int snd_nm256_free(struct nm256 *chip) snd_nm256_capture_stop(chip); if (chip->irq >= 0) - synchronize_irq(chip->irq); + free_irq(chip->irq, chip); if (chip->cport) iounmap(chip->cport); @@ -1447,8 +1447,6 @@ static int snd_nm256_free(struct nm256 *chip) iounmap(chip->buffer); release_and_free_resource(chip->res_cport); release_and_free_resource(chip->res_buffer); - if (chip->irq >= 0) - free_irq(chip->irq, chip); pci_disable_device(chip->pci); kfree(chip->ac97_regs); diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 71138ff9b310..bbcee2c09ae4 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3676,6 +3676,8 @@ static int snd_trident_free(struct snd_trident *trident) else if (trident->device == TRIDENT_DEVICE_ID_SI7018) { outl(0, TRID_REG(trident, SI_SERIAL_INTF_CTRL)); } + if (trident->irq >= 0) + free_irq(trident->irq, trident); if (trident->tlb.buffer.area) { outl(0, TRID_REG(trident, NX_TLBC)); if (trident->tlb.memhdr) @@ -3685,8 +3687,6 @@ static int snd_trident_free(struct snd_trident *trident) vfree(trident->tlb.shadow_entries); snd_dma_free_pages(&trident->tlb.buffer); } - if (trident->irq >= 0) - free_irq(trident->irq, trident); pci_release_regions(trident->pci); pci_disable_device(trident->pci); kfree(trident); From 409203074e9f3c423cdc7c38f984ce24ae261556 Mon Sep 17 00:00:00 2001 From: Tim Niemeyer Date: Tue, 22 Apr 2008 18:26:59 +0200 Subject: [PATCH 232/250] [ALSA] soc - s3c24xx - Improve diagnostic output Add some debug messages for suspend/resume and to add a clear prefix to s3c24xx-i2s and s3c24xx-pcm. Signed-off-by: Tim Niemeyer Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 ++++- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 301002cd3fc8..cb68b9ced601 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -46,7 +46,7 @@ #define S3C24XX_I2S_DEBUG 0 #if S3C24XX_I2S_DEBUG -#define DBG(x...) printk(KERN_DEBUG x) +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) #else #define DBG(x...) #endif @@ -414,6 +414,8 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) int s3c24xx_i2s_suspend(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai) { + DBG("Entered %s\n", __func__); + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); @@ -427,6 +429,7 @@ int s3c24xx_i2s_suspend(struct platform_device *pdev, int s3c24xx_i2s_resume(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai) { + DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 40112e2b1ecb..49580fb481d5 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -39,7 +39,7 @@ #define S3C24XX_PCM_DEBUG 0 #if S3C24XX_PCM_DEBUG -#define DBG(x...) printk(KERN_DEBUG x) +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x) #else #define DBG(x...) #endif From d8ed061a9fb2ab1f4bd90b5c30f4dc98b9c2085b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Apr 2008 18:27:22 +0200 Subject: [PATCH 233/250] [ALSA] soc - s3c24xx - Declare suspend and resume static Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index cb68b9ced601..ce935a71099b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -411,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) } #ifdef CONFIG_PM -int s3c24xx_i2s_suspend(struct platform_device *pdev, +static int s3c24xx_i2s_suspend(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -426,7 +426,7 @@ int s3c24xx_i2s_suspend(struct platform_device *pdev, return 0; } -int s3c24xx_i2s_resume(struct platform_device *pdev, +static int s3c24xx_i2s_resume(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai) { DBG("Entered %s\n", __func__); From a0b8f7d89b8de0cc79999b9fdd3a303912f3b2a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Apr 2008 19:39:49 +0200 Subject: [PATCH 234/250] [ALSA] hda - Fix model for Acer Aspire 5720z Set the proper model=acer for Acer Aspire 5720z with ALC268 codec. ALSA bug#3550: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3550 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 732515dcc99d..cdda64b02f46 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10404,6 +10404,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { }; static struct snd_pci_quirk alc268_cfg_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), From 5bae4199522c56110d33e6fd925c052bc5ae36a1 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Wed, 23 Apr 2008 10:50:32 +0200 Subject: [PATCH 235/250] pcsp - Don't build pcspkr when snd-pcsp is enabled - Update CREDITS with the pc-speaker driver authors. - Prevent pcspkr from being built together with snd-pcsp. Both pcspkr and snd-pcsp use the same platform driver name "pcspkr". Signed-off-by: Stas Sergeev --- CREDITS | 8 ++++++++ drivers/input/misc/Kconfig | 1 + 2 files changed, 9 insertions(+) diff --git a/CREDITS b/CREDITS index da0a56e23bee..8fec7b3f96d5 100644 --- a/CREDITS +++ b/CREDITS @@ -403,6 +403,8 @@ D: Linux CD and Support Giveaway List N: Erik Inge Bolsø E: knan@mo.himolde.no D: Misc kernel hacks +D: Updated PC speaker driver for 2.3 +S: Norway N: Andreas E. Bombe E: andreas.bombe@munich.netsurf.de @@ -3116,6 +3118,12 @@ S: Post Office Box 64132 S: Sunnyvale, California 94088-4132 S: USA +N: Stas Sergeev +E: stsp@users.sourceforge.net +D: PCM PC-Speaker driver +D: misc fixes +S: Russia + N: Simon Shapiro E: shimon@i-Connect.Net W: http://www.-i-Connect.Net/~shimon diff --git a/drivers/input/misc/Kconfig b/drivers/input/misc/Kconfig index c5263d63aca3..92b683411d5a 100644 --- a/drivers/input/misc/Kconfig +++ b/drivers/input/misc/Kconfig @@ -15,6 +15,7 @@ if INPUT_MISC config INPUT_PCSPKR tristate "PC Speaker support" depends on ALPHA || X86 || MIPS || PPC_PREP || PPC_CHRP || PPC_PSERIES + depends on SND_PCSP=n help Say Y here if you want the standard PC Speaker to be used for bells and whistles. From 05808ecc45802c1b533f42ed701a132d4c949034 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Apr 2008 13:50:08 +0200 Subject: [PATCH 236/250] [ALSA] hda - Fix Thinkpad X300 digital mic TP X300 digital mic requires additional init verbs with magic COEFs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f486eb16a386..e0a605adde42 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3740,6 +3740,9 @@ static struct hda_verb ad1984a_thinkpad_verbs[] = { {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, /* internal mic - dmic */ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, { } /* end */ }; From 40efc15fc637cff22cf9c4f02c63f3f398320f83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:09:31 +0200 Subject: [PATCH 237/250] [ALSA] soc - s3c24xx-i2s - Use linux/io.h Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ce935a71099b..340f36e6979c 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -32,7 +33,6 @@ #include #include -#include #include #include #include From 0015e7d1e2b09443ac76573a2fb886854aa1ca15 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:09:57 +0200 Subject: [PATCH 238/250] [ALSA] soc - s3c24xx-i2s - Fix tab/space breakage Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/s3c24xx-i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 340f36e6979c..50e06f0777f1 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -159,10 +159,10 @@ static void s3c24xx_snd_rxctrl(int on) * DMA engine will simply freeze randomly. */ - iisfcon &= ~S3C2410_IISFCON_RXENABLE; - iisfcon &= ~S3C2410_IISFCON_RXDMA; - iiscon |= S3C2410_IISCON_RXIDLE; - iiscon &= ~S3C2410_IISCON_RXDMAEN; + iisfcon &= ~S3C2410_IISFCON_RXENABLE; + iisfcon &= ~S3C2410_IISFCON_RXDMA; + iiscon |= S3C2410_IISCON_RXIDLE; + iiscon &= ~S3C2410_IISCON_RXDMAEN; iismod &= ~S3C2410_IISMOD_RXMODE; writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); From 0fe564a564922465ec3c483cee0e3dc6b368d879 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:10:28 +0200 Subject: [PATCH 239/250] [ALSA] soc - s3c24xx-i2s - Add missing spaces Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 50e06f0777f1..4ebcd6a8bf28 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -387,7 +387,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) if (s3c24xx_i2s.regs == NULL) return -ENXIO; - s3c24xx_i2s.iis_clk=clk_get(&pdev->dev, "iis"); + s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); if (s3c24xx_i2s.iis_clk == NULL) { DBG("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); From 1bfcd361461f25be7d6d180a8da30d02bc124046 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:12:19 +0200 Subject: [PATCH 240/250] [ALSA] soc - corgi - Fix checkpatch warnings Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/corgi.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1a70a6ac98ce..7f32a1167572 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -297,21 +297,19 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL)); + snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL)); if (err < 0) return err; } /* Add corgi specific widgets */ - for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - } /* Set up corgi specific audio path audio_map */ - for(i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_sync_endpoints(codec); return 0; @@ -353,7 +351,8 @@ static int __init corgi_init(void) { int ret; - if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky())) + if (!(machine_is_corgi() || machine_is_shepherd() || + machine_is_husky())) return -ENODEV; corgi_snd_device = platform_device_alloc("soc-audio", -1); From 29e36e49bdb7f24ca7cc0fb980fab2c407a8a2c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:13:04 +0200 Subject: [PATCH 241/250] [ALSA] soc - poodle - Fix checkpatch warnings Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/poodle.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4fbf8bba9627..7e830b218943 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -257,21 +257,19 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL)); + snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL)); if (err < 0) return err; } /* Add poodle specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - } /* Set up poodle specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_sync_endpoints(codec); return 0; From 22cd630285b6a12a50f02dfb23c531f151be5499 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:13:33 +0200 Subject: [PATCH 242/250] [ALSA] soc - spitz - Fix checkpatch warnings Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/spitz.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ecca39033fcc..d8b8372db00e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -313,15 +313,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) } /* Add spitz specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); - } /* Set up spitz specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_sync_endpoints(codec); return 0; From b32432e3f2d386d9563669c8cfdeaa473bfd8572 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:14:18 +0200 Subject: [PATCH 243/250] [ALSA] soc - pxa2xx-pcm - Fix checkpatch warnings Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/pxa2xx-pcm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index daeaa4c8b876..01ad7bf716b7 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -64,8 +64,8 @@ static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) if (dcsr & DCSR_ENDINTR) { snd_pcm_period_elapsed(substream); } else { - printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", - prtd->params->name, dma_ch, dcsr ); + printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", + prtd->params->name, dma_ch, dcsr); } } @@ -84,8 +84,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; + if (!dma) + return 0; /* this may get called several times by oss emulation * with different params */ @@ -363,7 +363,6 @@ struct snd_soc_platform pxa2xx_soc_platform = { .pcm_new = pxa2xx_pcm_new, .pcm_free = pxa2xx_pcm_free_dma_buffers, }; - EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); MODULE_AUTHOR("Nicolas Pitre"); From d454aee9be72472ee18b5397fda2c673f40a1e69 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:16:46 +0200 Subject: [PATCH 244/250] [ALSA] soc - wm8731 - Clean up checkpatch warnings Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8731.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 9c33fe874928..0cf9265fca8f 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -110,7 +110,7 @@ static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - wm8731_write_reg_cache (codec, reg, value); + wm8731_write_reg_cache(codec, reg, value); if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -154,8 +154,10 @@ static int wm8731_add_controls(struct snd_soc_codec *codec) int err, i; for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { - if ((err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0) + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8731_snd_controls[i], + codec, NULL)); + if (err < 0) return err; } @@ -221,15 +223,13 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec) { int i; - for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - } /* set up audio path interconnects */ - for(i = 0; intercon[i][0] != NULL; i++) { + for (i = 0; intercon[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]); - } snd_soc_dapm_new_widgets(codec); return 0; @@ -589,7 +589,7 @@ pcm_err: static struct snd_soc_device *wm8731_socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * WM8731 2 wire address is determined by GPIO5 @@ -651,7 +651,7 @@ err: static int wm8731_i2c_detach(struct i2c_client *client) { - struct snd_soc_codec* codec = i2c_get_clientdata(client); + struct snd_soc_codec *codec = i2c_get_clientdata(client); i2c_detach_client(client); kfree(codec->reg_cache); kfree(client); @@ -709,7 +709,7 @@ static int wm8731_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_paths); wm8731_socdev = socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; @@ -734,7 +734,7 @@ static int wm8731_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8731_i2c_driver); #endif kfree(codec->private_data); @@ -749,7 +749,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { .suspend = wm8731_suspend, .resume = wm8731_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); MODULE_DESCRIPTION("ASoC WM8731 driver"); From 42f3030f0cac474fc3232c8028b97f54b985718c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:17:12 +0200 Subject: [PATCH 245/250] [ALSA] soc - wm8750 - Clean up checkpatch warnings Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8750.c | 27 +++++++++++++-------------- 1 file changed, 13 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 77a857b997a2..16cd5d4d5ad9 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -110,7 +110,7 @@ static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - wm8750_write_reg_cache (codec, reg, value); + wm8750_write_reg_cache(codec, reg, value); if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -257,7 +257,8 @@ static int wm8750_add_controls(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL)); + snd_soc_cnew(&wm8750_snd_controls[i], + codec, NULL)); if (err < 0) return err; } @@ -478,15 +479,13 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec) { int i; - for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); - } /* set up audio path audio_mapnects */ - for(i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_new_widgets(codec); return 0; @@ -714,8 +713,8 @@ static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) } #define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -784,7 +783,8 @@ static int wm8750_resume(struct platform_device *pdev) if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); codec->dapm_state = SNDRV_CTL_POWER_D0; - schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(1000)); } return 0; @@ -864,7 +864,7 @@ pcm_err: around */ static struct snd_soc_device *wm8750_socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * WM8731 2 wire address is determined by GPIO5 @@ -979,8 +979,8 @@ static int wm8750_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_paths); wm8750_socdev = socdev; INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work); - -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; @@ -1025,7 +1025,7 @@ static int wm8750_remove(struct platform_device *pdev) run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8750_i2c_driver); #endif kfree(codec->private_data); @@ -1040,7 +1040,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = { .suspend = wm8750_suspend, .resume = wm8750_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); MODULE_DESCRIPTION("ASoC WM8750 driver"); From 24c053e755f2f77d9c9d9a9250ca1132eae280e7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Apr 2008 15:26:45 +0200 Subject: [PATCH 246/250] [ALSA] soc - ac97 - Clean up checkpatch warnings Also change some if (x == NULL) to if (!x). Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/ac97.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 242130cf1abd..2a1ffe396908 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -40,7 +40,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream) } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) struct snd_soc_codec_dai ac97_dai = { .name = "AC97 HiFi", @@ -86,7 +87,7 @@ static int ac97_soc_probe(struct platform_device *pdev) printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + if (!socdev->codec) return -ENOMEM; codec = socdev->codec; mutex_init(&codec->mutex); @@ -102,17 +103,17 @@ static int ac97_soc_probe(struct platform_device *pdev) /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if(ret < 0) + if (ret < 0) goto err; /* add codec as bus device for standard ac97 */ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus); - if(ret < 0) + if (ret < 0) goto bus_err; memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97); - if(ret < 0) + if (ret < 0) goto bus_err; ret = snd_soc_register_card(socdev); @@ -135,7 +136,7 @@ static int ac97_soc_remove(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - if(codec == NULL) + if (!codec) return 0; snd_soc_free_pcms(socdev); @@ -145,11 +146,10 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; } -struct snd_soc_codec_device soc_codec_dev_ac97= { +struct snd_soc_codec_device soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .remove = ac97_soc_remove, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); MODULE_DESCRIPTION("Soc Generic AC97 driver"); From 73bdd2ad7aac70456494c4a1d93f99fe88184dba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Apr 2008 17:08:58 +0200 Subject: [PATCH 247/250] [ALSA] pcsp - Fix dependency in Kconfig Added the proper dependency to Kconfig for snd-pcsp driver. Signed-off-by: Takashi Iwai --- sound/drivers/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 78648c4e9e73..fe85af1c5693 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -7,6 +7,7 @@ menu "Generic devices" config SND_PCSP tristate "Internal PC speaker support" depends on X86_PC && HIGH_RES_TIMERS + depends on INPUT help If you don't have a sound card in your computer, you can include a driver for the PC speaker which allows it to act like a primitive From efd89d9dcf75ab0a31b200db0ae4ae19cad25e48 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Wed, 23 Apr 2008 17:16:38 +0200 Subject: [PATCH 248/250] [ALSA] pcsp: fix wording in DEBUG_PAGEALLOC warning Signed-off-by: Stas Sergeev (fixed invalid KERN_WARNING by tiwai) Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index d8f96219fd36..59203511e77d 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -147,12 +147,8 @@ static int __devinit alsa_card_pcsp_init(struct device *dev) #ifdef CONFIG_DEBUG_PAGEALLOC /* Well, CONFIG_DEBUG_PAGEALLOC makes the sound horrible. Lets alert */ - printk(KERN_WARNING - "PCSP: Warning, CONFIG_DEBUG_PAGEALLOC is enabled!\n" - "You have to disable it if you want to use the PC-Speaker " - "driver.\n" - "Unless it is disabled, enjoy the horrible, distorted " - "and crackling noise.\n"); + printk(KERN_WARNING "PCSP: CONFIG_DEBUG_PAGEALLOC is enabled, " + "which may make the sound noisy.\n"); #endif return 0; From b415ed45f4db9f8365daac84cf2518642a174dc0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Apr 2008 17:47:28 +0200 Subject: [PATCH 249/250] [ALSA] Define MPU401 registers in sound/mpu401_uart.h Define some MPU401 registers in sound/mpu401_uart.h so that other drivers can refer to them. Signed-off-by: Takashi Iwai --- include/sound/mpu401.h | 15 +++++++++++++++ sound/drivers/mpu401/mpu401_uart.c | 10 ++++------ 2 files changed, 19 insertions(+), 6 deletions(-) diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index d45218b44dfe..68b634b75068 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -102,6 +102,21 @@ struct snd_mpu401 { #define MPU401C(mpu) (mpu)->cport #define MPU401D(mpu) (mpu)->port +/* + * control register bits + */ +/* read MPU401C() */ +#define MPU401_RX_EMPTY 0x80 +#define MPU401_TX_FULL 0x40 + +/* write MPU401C() */ +#define MPU401_RESET 0xff +#define MPU401_ENTER_UART 0x3f + +/* read MPU401D() */ +#define MPU401_ACK 0xfe + + /* */ diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index dd6ec4266732..18cca2457d44 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -49,12 +49,10 @@ static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu); */ -#define snd_mpu401_input_avail(mpu) (!(mpu->read(mpu, MPU401C(mpu)) & 0x80)) -#define snd_mpu401_output_ready(mpu) (!(mpu->read(mpu, MPU401C(mpu)) & 0x40)) - -#define MPU401_RESET 0xff -#define MPU401_ENTER_UART 0x3f -#define MPU401_ACK 0xfe +#define snd_mpu401_input_avail(mpu) \ + (!(mpu->read(mpu, MPU401C(mpu)) & MPU401_RX_EMPTY)) +#define snd_mpu401_output_ready(mpu) \ + (!(mpu->read(mpu, MPU401C(mpu)) & MPU401_TX_FULL)) /* Build in lowlevel io */ static void mpu401_write_port(struct snd_mpu401 *mpu, unsigned char data, From 3a841d519f91463361bbbe7addc24a0c1b2e9f99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Apr 2008 17:47:28 +0200 Subject: [PATCH 250/250] [ALSA] ice1724 - Fix IRQ lock-up with MPU access The sound boards with VT1724 and compatible chips may lock up when MPU401 is accessed together with the PCM streaming. This patch fixes the problem. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 99 ++++++++++++++++++++++++++++------ sound/pci/ice1712/prodigy192.c | 4 -- 2 files changed, 83 insertions(+), 20 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 13ea94f317cb..4490422fb930 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -222,6 +222,32 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice) return data; } +/* + * MPU401 accessor + */ +static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu, + unsigned long addr) +{ + /* fix status bits to the standard position */ + /* only RX_EMPTY and TX_FULL are checked */ + if (addr == MPU401C(mpu)) + return (inb(addr) & 0x0c) << 4; + else + return inb(addr); +} + +static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu, + unsigned char data, unsigned long addr) +{ + if (addr == MPU401C(mpu)) { + if (data == MPU401_ENTER_UART) + outb(0x01, addr); + /* what else? */ + } else + outb(data, addr); +} + + /* * Interrupt handler */ @@ -230,24 +256,53 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) { struct snd_ice1712 *ice = dev_id; unsigned char status; + unsigned char status_mask = + VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM; int handled = 0; +#ifdef CONFIG_SND_DEBUG + int timeout = 0; +#endif while (1) { status = inb(ICEREG1724(ice, IRQSTAT)); + status &= status_mask; if (status == 0) break; - - handled = 1; - /* these should probably be separated at some point, - * but as we don't currently have MPU support on the board - * I will leave it - */ - if ((status & VT1724_IRQ_MPU_RX)||(status & VT1724_IRQ_MPU_TX)) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt(irq, ice->rmidi[0]->private_data); - outb(status & (VT1724_IRQ_MPU_RX|VT1724_IRQ_MPU_TX), ICEREG1724(ice, IRQSTAT)); - status &= ~(VT1724_IRQ_MPU_RX|VT1724_IRQ_MPU_TX); +#ifdef CONFIG_SND_DEBUG + if (++timeout > 10) { + printk(KERN_ERR + "ice1724: Too long irq loop, status = 0x%x\n", + status); + break; } +#endif + handled = 1; + if (status & VT1724_IRQ_MPU_TX) { + if (ice->rmidi[0]) + snd_mpu401_uart_interrupt_tx(irq, + ice->rmidi[0]->private_data); + else /* disable TX to be sure */ + outb(inb(ICEREG1724(ice, IRQMASK)) | + VT1724_IRQ_MPU_TX, + ICEREG1724(ice, IRQMASK)); + /* Due to mysterical reasons, MPU_TX is always + * generated (and can't be cleared) when a PCM + * playback is going. So let's ignore at the + * next loop. + */ + status_mask &= ~VT1724_IRQ_MPU_TX; + } + if (status & VT1724_IRQ_MPU_RX) { + if (ice->rmidi[0]) + snd_mpu401_uart_interrupt(irq, + ice->rmidi[0]->private_data); + else /* disable RX to be sure */ + outb(inb(ICEREG1724(ice, IRQMASK)) | + VT1724_IRQ_MPU_RX, + ICEREG1724(ice, IRQMASK)); + } + /* ack MPU irq */ + outb(status, ICEREG1724(ice, IRQSTAT)); if (status & VT1724_IRQ_MTPCM) { /* * Multi-track PCM @@ -2236,10 +2291,7 @@ static int __devinit snd_vt1724_create(struct snd_card *card, } /* unmask used interrupts */ - if (! (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401)) - mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX; - else - mask = 0; + mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX; outb(mask, ICEREG1724(ice, IRQMASK)); /* don't handle FIFO overrun/underruns (just yet), * since they cause machine lockups @@ -2373,14 +2425,29 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (! c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { + struct snd_mpu401 *mpu; if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, ICEREG1724(ice, MPU_CTRL), - MPU401_INFO_INTEGRATED, + (MPU401_INFO_INTEGRATED | + MPU401_INFO_TX_IRQ), ice->irq, 0, &ice->rmidi[0])) < 0) { snd_card_free(card); return err; } + mpu = ice->rmidi[0]->private_data; + mpu->read = snd_vt1724_mpu401_read; + mpu->write = snd_vt1724_mpu401_write; + /* unmask MPU RX/TX irqs */ + outb(inb(ICEREG1724(ice, IRQMASK)) & + ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX), + ICEREG1724(ice, IRQMASK)); +#if 0 /* for testing */ + /* set watermarks */ + outb(VT1724_MPU_RX_FIFO | 0x1, + ICEREG1724(ice, MPU_FIFO_WM)); + outb(0x1, ICEREG1724(ice, MPU_FIFO_WM)); +#endif } } diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 25ceb67a9c16..48d3679292a7 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -812,10 +812,6 @@ struct snd_ice1712_card_info snd_vt1724_prodigy192_cards[] __devinitdata = { .build_controls = prodigy192_add_controls, .eeprom_size = sizeof(prodigy71_eeprom), .eeprom_data = prodigy71_eeprom, - /* the current MPU401 code loops infinitely - * when opening midi device - */ - .no_mpu401 = 1, }, { } /* terminator */ };