From f0cdcf3ab6c62b3f774a2af15dfa01988e7a9b02 Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Fri, 30 Mar 2012 00:13:02 +0800 Subject: [PATCH 01/24] ASoC: sgtl5000: Enable VAG when DAC/ADC up As manual described, VAG is an internal voltage reference of DAC/ADC, So enabled it before DAC/ADC up. One more thing should care about is VAG fully ramped down requires 400ms, wait it to avoid pop. Signed-off-by: Zeng Zhaoming Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d1926266fe00..8e92fb88ed09 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } /* - * using codec assist to small pop, hp_powerup or lineout_powerup - * should stay setting until vag_powerup is fully ramped down, - * vag fully ramped down require 400ms. + * As manual described, ADC/DAC only works when VAG powerup, + * So enabled VAG before ADC/DAC up. + * In power down case, we need wait 400ms when vag fully ramped down. */ -static int small_pop_event(struct snd_soc_dapm_widget *w, +static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { @@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), @@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), }; @@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ From cd1506736f3a77429f619ede817a119a7ff5f7e5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:17 -0600 Subject: [PATCH 02/24] ASoC: tegra: ensure clocks are enabled when touching registers Debugfs files could be accessed any time, so explicitly enable clocks when reading registers to generate debugfs file content. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 4 ++++ sound/soc/tegra/tegra_spdif.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33509de52540..2d98c925c0aa 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused) struct tegra_i2s *i2s = s->private; int i; + clk_enable(i2s->clk_i2s); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_i2s_read(i2s, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(i2s->clk_i2s); + return 0; } diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index 475428cf270e..9ff2c601445f 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused) struct tegra_spdif *spdif = s->private; int i; + clk_enable(spdif->clk_spdif_out); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_spdif_read(spdif, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(spdif->clk_spdif_out); + return 0; } From e95cee0e36c671db2804f2763b547a86930061ea Mon Sep 17 00:00:00 2001 From: Martin Jansa Date: Mon, 2 Apr 2012 10:24:08 +0200 Subject: [PATCH 03/24] ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro * fixes sound/soc/pxa/pxa2xx-i2s.c:86:2: error: implicit declaration of function 'IOMEM' [-Werror=implicit-function-declaration] sound/soc/pxa/pxa2xx-i2s.c:86:2: error: initializer element is not constant after 23019a733bb83c8499f192fb428b7e6e81c95a34 removed IOMEM definition from arch/arm/mach-pxa/include/mach/hardware.h Signed-off-by: Martin Jansa Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd51e55f..d08583790d23 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include From 6c284903731eae12ae62aa138f479d48ccbcf1d1 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 3 Apr 2012 09:45:43 +0300 Subject: [PATCH 04/24] MAINTAINERS: Add missing ASoC OMAP co-maintainer Peter Ujfalusi has been co-maintaining sound/soc/omap/ for years but was missing from this MAINTAINERS entry. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index eecf3441ac21..85c599b4392a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4807,6 +4807,7 @@ F: arch/arm/mach-omap2/clockdomain2xxx_3xxx.c F: arch/arm/mach-omap2/clockdomain44xx.c OMAP AUDIO SUPPORT +M: Peter Ujfalusi M: Jarkko Nikula L: alsa-devel@alsa-project.org (subscribers-only) L: linux-omap@vger.kernel.org From fef9516425cb3a03a4a95b4de3cf8c575521df9a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Apr 2012 12:06:24 +0100 Subject: [PATCH 05/24] MAINTAINERS: Don't list everyone working on Wolfson drivers Rather than listing every single person who works on the drivers include the mailing list where they can all be found. Leave myself as a human contact. Signed-off-by: Mark Brown --- MAINTAINERS | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 85c599b4392a..5190cf25fd8d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7463,8 +7463,7 @@ F: include/linux/wm97xx.h WOLFSON MICROELECTRONICS DRIVERS M: Mark Brown -M: Ian Lartey -M: Dimitris Papastamos +L: patches@opensource.wolfsonmicro.com T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices From 1f99e44cf059d2ed43c5a0724fa738b83800f725 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 4 Apr 2012 23:28:01 -0700 Subject: [PATCH 06/24] ASoC: ak4642: fixup: mute needs +1 step ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute. But current settings didn't care +1 step for mute. This patch adds it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f8e10ced244a..b3e24f289421 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -140,7 +140,7 @@ * min : 0xFE : -115.0 dB * mute: 0xFF */ -static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { From 00792ac4e0d88e82fc489a5e1c4d4435125a301c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 5 Apr 2012 09:45:51 -0300 Subject: [PATCH 07/24] ASoC: imx-audmux: Fix ssi port numbers in sysfs Doing a 'cat /sys/kernel/debug/audmux/ssi7' causes the following oops to be printed by the kernel: Uhandled fault: external abort on non-linefetch (0x008) at 0xf53b003c Internal error: : 8 [#1] PREEMPT Modules linked in: CPU: 0 Not tainted (3.3.0-00033-gecc726e-dirty #307) PC is at audmux_read_file+0x68/0x2f4 LR is at clk_enable+0x3c/0x48 pc : [] lr : [] psr: a0000013 sp : c3ad3f38 ip : c30a4000 fp : 00000003 r10: 00001000 r9 : be83fb00 r8 : c3ad3f80 r7 : c3ad3f80 r6 : 00000007 r5 : 00031010 r4 : c30a5000 r3 : f53b0000 r2 : 0000003c r1 : 380fa100 r0 : c068dda0 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 0005317f Table: 83034000 DAC: 00000015 Process cat (pid: 1042, stack limit = 0xc3ad2270) Stack: (0xc3ad3f38 to 0xc3ad4000) 3f20: c3139180 00000000 3f40: c3bc6500 00001000 be83fb00 c3ad3f80 00001000 c3ad2000 00000000 c0095f3c 3f60: 00000003 c3bc6508 c3bc6500 be83fb00 00000000 00000000 00001000 c0096010 3f80: 00000000 00000000 b6fe2050 00000000 00001000 be83fb00 00000003 00000003 3fa0: c000eb88 c000e9e0 00001000 be83fb00 00000003 be83fb00 00001000 00000000 3fc0: 00001000 be83fb00 00000003 00000003 00000001 00000001 00000000 00000003 3fe0: 000bec8c be83fae0 0000f808 b6ea8d5c 60000010 00000003 7dff7ede 749bedf1 [] (audmux_read_file+0x68/0x2f4) from [] (vfs_read+0xb0/0x144) [] (vfs_read+0xb0/0x144) from [] (sys_read+0x40/0x70) [] (sys_read+0x40/0x70) from [] (ret_fast_syscall+0x0/0x2c) Code: e1a02186 e2822004 e3500000 e7935186 (e7937002) ---[ end trace 4d046e31309023de ]--- Fix the ssi port numbers in sysfs to fix this problem. Reported-by: Joan Carles Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/imx/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 601df809a26a..912a342ef776 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -158,7 +158,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 1; i < 8; i++) { + for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) From 66bb2a7f835a28a9405f3f6571fbf34156e6bc1e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 5 Apr 2012 10:57:51 -0300 Subject: [PATCH 08/24] ASoC: imx-audmux: Check for NULL pointer Check for NULL pointer before accessing it. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/imx/imx-audmux.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 912a342ef776..0fe66c3dde12 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -79,6 +79,9 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + if (!audmux_base) + return -ENOSYS; + if (audmux_clk) clk_prepare_enable(audmux_clk); From 3fec6b6d5a53d37194735268b9e220f75ca37f19 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 12:28:01 -0600 Subject: [PATCH 09/24] ASoC: set idle_bias_off=1 for all platform DAPM contexts The ASoC core currently defaults to using STANDBY rather than OFF for idle ASoC platform devices, which causes a permanent pm_runtime_get() on them. This keeps the device active unnecessarily. This can be especially problematic when the ASoC platform device and DAI device are the same device. The distinction between OFF and STANDBY is likely not relevant for ASoC platform drivers, since they aren't analog devices. So, solve this issue by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this turns out to be a problem, this value could be sourced from the snd_soc_platform_driver, similarly to soc_probe_codec(). Note: Prior to this change, this caused a large (10) runtime_active count for the Tegra I2S controller even when not in use, and a leak in that value as streams were started and stopped. This change probably hides a bug. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4deebc0801a..8d2ebf502df4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1087,6 +1087,8 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + platform->dapm.idle_bias_off = 1; + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { From 8abe05c6eb358967f16bce8a02c88d57c82cfbd6 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 23:11:16 -0600 Subject: [PATCH 10/24] ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed the prototype of tegra_i2s_debug_add, but didn't update the dummy inline used when !CONFIG_DEBUG_FS. Fix that. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown Cc: # 3.3 --- sound/soc/tegra/tegra_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 2d98c925c0aa..e53349912b2e 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -116,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) { } From 156d14da4cfc4fe01b705d6e2d22e44c0a2dbecd Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 9 Apr 2012 10:16:32 +0200 Subject: [PATCH 11/24] sound: sound/oss/msnd_pinnacle.c: add vfrees At the point of this error-handling code, HAVE_DSPCODEH may be undefined, so free INITCODE and PERMCODE as done elsewhere. A jump and label are introduced to avoid code duplication. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/oss/msnd_pinnacle.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index 2c79d60a725f..536c4c0514d3 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1294,6 +1294,8 @@ static int __init calibrate_adc(WORD srate) static int upload_dsp_code(void) { + int ret = 0; + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); #ifndef HAVE_DSPCODEH INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE); @@ -1312,7 +1314,8 @@ static int upload_dsp_code(void) memcpy_toio(dev.base, PERMCODE, PERMCODESIZE); if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) { printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); - return -ENODEV; + ret = -ENODEV; + goto out; } #ifdef HAVE_DSPCODEH printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); @@ -1320,12 +1323,13 @@ static int upload_dsp_code(void) printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); #endif +out: #ifndef HAVE_DSPCODEH vfree(INITCODE); vfree(PERMCODE); #endif - return 0; + return ret; } #ifdef MSND_CLASSIC From 38be95dd3d314bd393a26f6e441ae2c57ef7f064 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 9 Apr 2012 10:16:35 +0200 Subject: [PATCH 12/24] ALSA: sound/isa/sscape.c: add missing resource-release code At the point of this error-handling code, both regions and the dma have been allocated, so free it as done in previous and subsequent error-handling code. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b4a6aa960f4b..8490f59709bb 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1019,13 +1019,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } /* From fae3d88a5c56c3f836e95c4516da883a48612437 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Tue, 10 Apr 2012 17:00:35 +0800 Subject: [PATCH 13/24] ALSA: hda - hide HDMI/ELD printks unless snd.debug=2 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Also remove two warnings when CONFIG_SND_DEBUG is not set: sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’: sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable] sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable] Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- include/sound/core.h | 10 ++++++++++ sound/pci/hda/hda_eld.c | 6 +++--- sound/pci/hda/patch_hdmi.c | 9 ++++----- 3 files changed, 17 insertions(+), 8 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index b6e0f57d451d..bc056687f647 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -325,6 +325,13 @@ void release_and_free_resource(struct resource *res); /* --- */ +/* sound printk debug levels */ +enum { + SND_PR_ALWAYS, + SND_PR_DEBUG, + SND_PR_VERBOSE, +}; + #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) __printf(4, 5) void __snd_printk(unsigned int level, const char *file, int line, @@ -354,6 +361,8 @@ void __snd_printk(unsigned int level, const char *file, int line, */ #define snd_printd(fmt, args...) \ __snd_printk(1, __FILE__, __LINE__, fmt, ##args) +#define _snd_printd(level, fmt, args...) \ + __snd_printk(level, __FILE__, __LINE__, fmt, ##args) /** * snd_BUG - give a BUG warning message and stack trace @@ -383,6 +392,7 @@ void __snd_printk(unsigned int level, const char *file, int line, #else /* !CONFIG_SND_DEBUG */ #define snd_printd(fmt, args...) do { } while (0) +#define _snd_printd(level, fmt, args...) do { } while (0) #define snd_BUG() do { } while (0) static inline int __snd_bug_on(int cond) { diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index b58b4b1687fa..4c054f4486b9 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - printk(KERN_INFO "HDMI: supports coding type %s:" + _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" " channels = %d, rates =%s%s\n", cea_audio_coding_type_names[a->format], a->channels, @@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e) { int i; - printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 540cd13f7f15..83f345f3c961 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -757,8 +757,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) struct hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int pin_nid; - int pd = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); int pin_idx; struct hda_jack_tbl *jack; @@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) pin_nid = jack->nid; jack->jack_dirty = 1; - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pd, eldv); + codec->addr, pin_nid, + !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx < 0) @@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) if (eld->monitor_present) eld_valid = !!(present & AC_PINSENSE_ELDV); - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld_valid); From 912093bc7c08f59e97faed2c0269e1e5429dcd58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Apr 2012 14:03:41 +0200 Subject: [PATCH 14/24] ALSA: hda/realtek - Add a few ALC882 model strings back Since there are still many Acer models that might not be covered by the current fixup table, let's add back a few typical model names so that user can test the fixup without recompiling. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 4 +++- sound/pci/hda/patch_realtek.c | 10 +++++++++- 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index d97d992ced14..03f7897c6414 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -43,7 +43,9 @@ ALC680 ALC882/883/885/888/889 ====================== - N/A + acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G + acer-aspire-8930g Acer Aspire 8330G/6935G + acer-aspire Acer Aspire others ALC861/660 ========== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9917e55d6f11..e7b2b839a539 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5399,6 +5399,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc882_fixup_models[] = { + {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, + {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, + {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {} +}; + /* * BIOS auto configuration */ @@ -5439,7 +5446,8 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, + alc882_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); From 038d4fef376bc494d4f11072d2ab248414b7d568 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Apr 2012 17:18:12 +0200 Subject: [PATCH 15/24] ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co Add GPIO1 setup explicitly for Acer Aspire 493x & co. This could be set by alc_auto_init_amp(), but it's safer to set it more explicitly in the fixup table. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7b2b839a539..4eec2150312b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5269,7 +5269,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x16, 0x99130111 }, /* CLFE speaker */ { 0x17, 0x99130112 }, /* surround speaker */ { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC882_FIXUP_ACER_ASPIRE_8930G] = { .type = ALC_FIXUP_PINS, @@ -5312,7 +5314,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC885_FIXUP_MACPRO_GPIO] = { .type = ALC_FIXUP_FUNC, From fe97da1f7001ca0f572358462606eb3d1bde3f23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Apr 2012 08:00:19 +0200 Subject: [PATCH 16/24] ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G It's compatible with 8930G. Using the same fixup gives the proper 5.1 sound back. Reported-and-tested-by: Dany Martineau Cc: [v3.3+] 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 4eec2150312b..d25a6f90a37b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5363,6 +5363,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), + SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), From 29ebe40284c75a5888c601872059fca7e258528d Mon Sep 17 00:00:00 2001 From: Josh Boyer Date: Thu, 12 Apr 2012 13:55:36 -0400 Subject: [PATCH 17/24] ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines A user reported that setting model=imac24 used to allow sound to work on their Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models to auto-parser" removed this model option. All Mac machines are now explicitly handled with a quirk and the auto-parser. This adds a quirk for the device found on the Mac Pro 5,1 machines. This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559 [sorted the new entry in the ID number order by tiwai] Reported-by: Gabriel Somlo Signed-off-by: Josh Boyer 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 d25a6f90a37b..8f4a48463fad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5389,6 +5389,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), From 7d7eb9ea314e992413620610b4d09c9cd5fa8959 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 12 Apr 2012 22:11:25 +0200 Subject: [PATCH 18/24] ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace). In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the 'for (;;)' loop, if the 'badness' value returned from fill_and_eval_dacs() is negative, then we'll return from the function without freeing the memory we allocated for 'best_cfg', thus leaking. Fix the leak by kfree()'ing the memory when badness is negative. While I was there I also noticed some trailing whitespace in the function that I removed (along with all other trailing whitespace in the file) - it didn't seem worth-while to do that as two patches, so I hope it's OK that I just did it all as one patch. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8f4a48463fad..2508f8109f11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3398,8 +3398,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) for (;;) { badness = fill_and_eval_dacs(codec, fill_hardwired, fill_mio_first); - if (badness < 0) + if (badness < 0) { + kfree(best_cfg); return badness; + } debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", cfg->line_out_type, fill_hardwired, fill_mio_first, badness); @@ -3434,7 +3436,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; fill_hardwired = true; continue; - } + } if (cfg->hp_outs > 0 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { cfg->speaker_outs = cfg->line_outs; @@ -3448,7 +3450,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_HP_OUT; fill_hardwired = true; continue; - } + } break; } @@ -4423,7 +4425,7 @@ static int alc_parse_auto_config(struct hda_codec *codec, static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; - static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; + static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } @@ -6093,7 +6095,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), @@ -6310,7 +6312,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec, { if (action == ALC_FIXUP_ACT_PRE_PROBE) codec->no_jack_detect = 1; -} +} static const struct alc_fixup alc861_fixups[] = { [ALC861_FIXUP_FSC_AMILO_PI1505] = { @@ -6728,7 +6730,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1), From f2ec52d4c3698c995c89c579c34d818eab589d8b Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 17 Apr 2012 17:03:42 -0700 Subject: [PATCH 19/24] ALSA: fix core/vmaster.c kernel-doc warning Fix kernel-doc warning in sound/core/vmaster.c: Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 14a286a7bf2b..857586135d18 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -419,6 +419,7 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master); * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control * @kcontrol: vmaster kctl element * @hook: the hook function + * @private_data: the private_data pointer to be saved * * Adds the given hook to the vmaster control element so that it's called * at each time when the value is changed. From 118cb4a408e1c4021ac85d6c05da66bb6f57e556 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 07:33:27 +0200 Subject: [PATCH 20/24] ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1 Through the transition to the auto-parser, the support for Quanta/Gericom KN1 got broken. There are two problems behind it: - This machine doesn't like the default COEF setup for ALC260 we take now as default - BIOS doesn't set the pins correctly at all; especially the machine uses only the pin 0x0f for both headphone and speaker This patch adds the fixup as a workaround for these issues. Reported-and-tested-by: Uros Vampl Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++++++--- 1 file changed, 45 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2508f8109f11..e65e35433055 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1445,6 +1445,13 @@ enum { ALC_FIXUP_ACT_BUILD, }; +static void alc_apply_pincfgs(struct hda_codec *codec, + const struct alc_pincfg *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} + static void alc_apply_fixup(struct hda_codec *codec, int action) { struct alc_spec *spec = codec->spec; @@ -1478,9 +1485,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) snd_printdd(KERN_INFO "hda_codec: %s: " "Apply pincfg for %s\n", codec->chip_name, modelname); - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, - cfg->val); + alc_apply_pincfgs(codec, cfg); break; case ALC_FIXUP_VERBS: if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) @@ -4861,6 +4866,7 @@ enum { ALC260_FIXUP_GPIO1_TOGGLE, ALC260_FIXUP_REPLACER, ALC260_FIXUP_HP_B1900, + ALC260_FIXUP_KN1, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -4888,6 +4894,36 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, } } +static void alc260_fixup_kn1(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct alc_pincfg pincfgs[] = { + { 0x0f, 0x02214000 }, /* HP/speaker */ + { 0x12, 0x90a60160 }, /* int mic */ + { 0x13, 0x02a19000 }, /* ext mic */ + { 0x18, 0x01446000 }, /* SPDIF out */ + /* disable bogus I/O pins */ + { 0x10, 0x411111f0 }, + { 0x11, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + }; + + switch (action) { + case ALC_FIXUP_ACT_PRE_PROBE: + alc_apply_pincfgs(codec, pincfgs); + break; + case ALC_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_NONE; + break; + } +} + static const struct alc_fixup alc260_fixups[] = { [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, @@ -4938,7 +4974,11 @@ static const struct alc_fixup alc260_fixups[] = { .v.func = alc260_fixup_gpio1_toggle, .chained = true, .chain_id = ALC260_FIXUP_COEF, - } + }, + [ALC260_FIXUP_KN1] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_kn1, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4948,6 +4988,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), + SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} From 3e843196c697ee2c319d96e861980fb4c3e04e24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 12:04:03 +0200 Subject: [PATCH 21/24] ALSA: hda/sigmatel - Fix inverted mute LED While refactoring the mute-LED handling for HP laptops, I messed up the polarity check in a wrong way. The red (or the mute-LED if any) should appear in the muted state, corresponding to GPIO on. Reported-by: Mikko Vinni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 33a9946b492c..4742cac26aa9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5063,12 +5063,11 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) if (spec->gpio_led_polarity) muted = !muted; - /*polarity defines *not* muted state level*/ if (!spec->vref_mute_led_nid) { if (muted) - spec->gpio_data &= ~spec->gpio_led; /* orange */ + spec->gpio_data |= spec->gpio_led; else - spec->gpio_data |= spec->gpio_led; /* white */ + spec->gpio_data &= ~spec->gpio_led; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } else { From 590b4775d6b628c7ad215fd0335a0a787032e2dd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Apr 2012 00:00:27 -0700 Subject: [PATCH 22/24] ALSA: workaround: change the timing of alsa_sound_last_init() Current alsa_sound_last_init() was called as __initcall(). So, on current ALSA, only devices that had been properly registered at this point were shown. So, it will show "No soundcards found" if driver requests probe deferment. it's often misleading. This patch delays the timing of alsa_sound_last_init() as workaround. Signed-off-by: Kuninori Morimoto Reviwed-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/last.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/last.c b/sound/last.c index bdd0857b8871..7ffc182e0844 100644 --- a/sound/last.c +++ b/sound/last.c @@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void) return 0; } -__initcall(alsa_sound_last_init); +late_initcall_sync(alsa_sound_last_init); From ca3649de026ff95c6f2847e8d096cf2f411c02b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 15:15:25 +0200 Subject: [PATCH 23/24] ALSA: hda/conexant - Don't set HP pin-control bit unconditionally Some output pins on Conexant chips have no HP control bit, but the auto-parser initializes these pins unconditionally with PIN_HP. Check the pin-capability and avoid the HP bit if not supported. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d29d6d377904..f52c9ef3cc8c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3951,9 +3951,14 @@ static void cx_auto_init_output(struct hda_codec *codec) int i; mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids); - for (i = 0; i < cfg->hp_outs; i++) + for (i = 0; i < cfg->hp_outs; i++) { + unsigned int val = PIN_OUT; + if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) & + AC_PINCAP_HP_DRV) + val |= AC_PINCTL_HP_EN; snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + } mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins); From d70f363222ef373c2037412f09a600357cfa1c7a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 15:18:08 +0200 Subject: [PATCH 24/24] ALSA: hda/conexant - Set up the missing docking-station pins ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the docking-station ports, but BIOS doesn't initialize for these pins. Thus, like the former X200, we need to set up the pins manually in the driver. The odd part is that the same PCI SSID is used for X200 and T400, thus we need to prepare individual fixup tables for cx5051 and others. Bugzilla entries: https://bugzilla.redhat.com/show_bug.cgi?id=808559 https://bugzilla.redhat.com/show_bug.cgi?id=806217 https://bugzilla.redhat.com/show_bug.cgi?id=810697 Reported-by: Josh Boyer Reported-by: Jens Taprogge Tested-by: Jens Taprogge Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 32 +++++++++++++++++++++++++++----- 1 file changed, 27 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f52c9ef3cc8c..58b5de4a6eed 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4367,8 +4367,10 @@ static void apply_pin_fixup(struct hda_codec *codec, enum { CXT_PINCFG_LENOVO_X200, + CXT_PINCFG_LENOVO_TP410, }; +/* ThinkPad X200 & co with cxt5051 */ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4376,15 +4378,33 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { {} }; -static const struct cxt_pincfg *cxt_pincfg_tbl[] = { - [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, +/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */ +static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = { + { 0x19, 0x042110ff }, /* HP (seq# overridden) */ + { 0x1a, 0x21a190f0 }, /* dock-mic */ + { 0x1c, 0x212140ff }, /* dock-HP */ + {} }; -static const struct snd_pci_quirk cxt_fixups[] = { +static const struct cxt_pincfg *cxt_pincfg_tbl[] = { + [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410, +}; + +static const struct snd_pci_quirk cxt5051_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; +static const struct snd_pci_quirk cxt5066_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + {} +}; + /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches * can be created (bko#42825) */ @@ -4421,11 +4441,13 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f15051: add_cx5051_fake_mutes(codec); + apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl); + break; + default: + apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl); break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); - err = cx_auto_search_adcs(codec); if (err < 0) return err;