From 31794bc37bf2db84f085da52b72bfba65739b2d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Feb 2012 22:00:47 -0800 Subject: [PATCH 01/11] ASoC: wm8962: Fix sidetone enumeration texts The sidetone enumeration texts have left and right swapped. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 29c4b02c4790..0ac228b7dc04 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *st_text[] = { "None", "Right", "Left" }; +static const char *st_text[] = { "None", "Left", "Right" }; static const struct soc_enum str_enum = SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); From ef8d60fb79614a86a82720dc2402631dbcafb315 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 10:12:38 +0100 Subject: [PATCH 02/11] ALSA: hda/realtek - Fix surround output regression on Acer Aspire 5935 The previous fix for the speaker on Acer Aspire 59135 introduced another problem for surround outputs. It changed the connections on the line-in/mic pins for limiting the routes, but it left the modified connections. Thus wrong connection indices were written when set to 4ch or 6ch mode. This patch fixes it by restoring the right connections just after parsing the tree but before the initialization. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 389a28a21fa9..3647baa9bfed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4441,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { + /* fake the connections during parsing the tree */ hda_nid_t conn1[2] = { 0x0c, 0x0d }; hda_nid_t conn2[2] = { 0x0e, 0x0f }; snd_hda_override_conn_list(codec, 0x14, 2, conn1); snd_hda_override_conn_list(codec, 0x15, 2, conn1); snd_hda_override_conn_list(codec, 0x18, 2, conn2); snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } else if (action == ALC_FIXUP_ACT_PROBE) { + /* restore the connections */ + hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; + snd_hda_override_conn_list(codec, 0x14, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 5, conn); } } From e555cf363167f09efae96d32a363e24c4de16b7b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 20 Feb 2012 20:14:16 -0800 Subject: [PATCH 03/11] ASoC: ak4642: fixup HeadPhone L/R dapm settings Current ak4642 driver had wrong dapm settings for headphone L/R. If you select headphone L, and select R after that, headphone L setting was removed by R settings. This patch fixes it up. It provides just "Headphone Enable" to user side Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5ef70b5d27e4..278c0a0575f5 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), - - SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; -static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { - SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), -}; +static const struct snd_kcontrol_new ak4642_headphone_control = + SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), @@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("LINEOUT"), - SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, + &ak4642_headphone_control), - SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], @@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ - {"HPOUTL", NULL, "HPOUTL Mixer"}, - {"HPOUTR", NULL, "HPOUTR Mixer"}, + {"HPOUTL", NULL, "HPL Out"}, + {"HPOUTR", NULL, "HPR Out"}, {"LINEOUT", NULL, "LINEOUT Mixer"}, - {"HPOUTL Mixer", "DACH", "DAC"}, - {"HPOUTR Mixer", "DACH", "DAC"}, + {"HPL Out", NULL, "Headphone Enable"}, + {"HPR Out", NULL, "Headphone Enable"}, + + {"Headphone Enable", "Switch", "DACH"}, + + {"DACH", NULL, "DAC"}, + {"LINEOUT Mixer", "DACL", "DAC"}, }; From cb74eb15ac88d6aacf7e58db1d8f8dadee710fd9 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Tue, 21 Feb 2012 21:26:31 +0000 Subject: [PATCH 04/11] ALSA: snd-usb-caiaq: Fix the return of XRUN Commit 3702b08 added a lock, but did not account for the case of SNDRV_PCM_POS_XRUN, which would get immediately overwritten. This could be bundled into one if-else-if statement, but the goto helps to clarify the 'exceptional' case. Thanks to Andreas Pape for spotting this. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2cf87f5afed4..fde9a7a29cb6 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) spin_lock(&dev->spinlock); - if (dev->input_panic || dev->output_panic) + if (dev->input_panic || dev->output_panic) { ptr = SNDRV_PCM_POS_XRUN; + goto unlock; + } if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ptr = bytes_to_frames(sub->runtime, @@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); +unlock: spin_unlock(&dev->spinlock); return ptr; } From 7679e42ec833ed70aa34790a5f39dcb7e5bda4fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Feb 2012 15:52:56 +0000 Subject: [PATCH 05/11] ASoC: dapm: Check for bias level when powering down Recent enhancements in the bias management means that we might not be in standby when the CODEC is idle and can have active widgets without being in full power mode but the shutdown functionality assumes these things. Add checks for the bias level at each stage so that we don't do transitions other than the ON->PREPARE->STANDBY->OFF ones that the drivers are expecting. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/soc-dapm.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f55ded4047f..1315663c1c09 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); + if (dapm->bias_level == SND_SOC_BIAS_ON) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_STANDBY); } } @@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&codec->dapm, + SND_SOC_BIAS_OFF); } } From 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7 Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Thu, 23 Feb 2012 15:43:18 +0100 Subject: [PATCH 06/11] ASoC: i.MX SSI: Fix DSP_A format. According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects whether the most significant or the less significant part of the data word written to the FIFO is transmitted. As DSP_A is the same as DSP_B with a data offset of 1 bit, it doesn't make any sense to remove TXBIT0 bit here. Signed-off-by: Javier Martin Acked-by: Sascha Hauer Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/imx/imx-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 01d1f749cf02..b6adbed6e506 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; break; } From 068b939431486f524438330b0848a8222e33d421 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 25 Feb 2012 11:13:16 +0100 Subject: [PATCH 07/11] ALSA: hda/realtek - Fix resume of multiple input sources When there are multiple input sources, the driver wrongly overwrites with the value of the last input source on other slots at resume. Thus the primary input source may be shown wrongly. Reported-and-tested-by: Julian Sikorski Cc: [v3.1+] 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 3647baa9bfed..4fe2d5960a04 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3797,7 +3797,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec) else nums = spec->num_adc_nids; for (c = 0; c < nums; c++) - alc_mux_select(codec, 0, spec->cur_mux[c], true); + alc_mux_select(codec, c, spec->cur_mux[c], true); } /* add mic boosts if needed */ From 87c9e7d7027643bf248b396c15c804456e967fcd Mon Sep 17 00:00:00 2001 From: Alban Bedel Date: Sat, 25 Feb 2012 16:15:57 +0100 Subject: [PATCH 08/11] ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3 opl3->private_data was set even if opl3 could not be created. Signed-off-by: Alban Bedel Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 95ffa6a9db6e..496f14c1a731 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto out_err; + opl3->private_data = chip; } - opl3->private_data = chip; - sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->ctrl_io, chip->irq); From 7bff172a352a2fbe9856bba517d71a2072aab041 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Feb 2012 09:41:17 +0100 Subject: [PATCH 09/11] ALSA: hda - Always set HP pin in unsol handler for STAC/IDT codecs A bug report with an old Sony laptop showed that we can't rely on BIOS setting the pins of headphones but the driver should set always by itself. Cc: 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 6345df131a00..9dbb5735d778 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4629,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (presence) + if (1 /*presence*/) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions From 3868137ea41866773e75d9ac4b9988dcc361ff1d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Feb 2012 15:00:58 +0100 Subject: [PATCH 10/11] ALSA: hda - Add a fake mute feature Some codecs don't supply the mute amp-capabilities although the lowest volume gives the mute. It'd be handy if the parser provides the mute mixers in such a case. This patch adds an extension amp-cap bit (which is used only in the driver) to represent the min volume = mute state. Also modified the amp cache code to support the fake mute feature when this bit is set but the real mute bit is unset. In addition, conexant cx5051 parser uses this new feature to implement the missing mute controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 ++++++-- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/patch_conexant.c | 22 +++++++++++++++++++++- 3 files changed, 30 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c2c65f63bf06..0ae6eb20b13b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - parm |= val; + if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && + (info->amp_caps & AC_AMPCAP_MIN_MUTE)) + ; /* set the zero value as a fake mute */ + else + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute) + if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc0d464..f0f1943a4b2c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -298,6 +298,9 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a7a5733aa4d2..ca117cf5132f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + if (!(query_amp_caps(codec, nid, hda_dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) break; } return 0; @@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; +/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches + * can be created (bko#42825) + */ +static void add_cx5051_fake_mutes(struct hda_codec *codec) +{ + static hda_nid_t out_nids[] = { + 0x10, 0x11, 0 + }; + hda_nid_t *p; + + for (p = out_nids; *p; p++) + snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, + AC_AMPCAP_MIN_MUTE | + query_amp_caps(codec, *p, HDA_OUTPUT)); +} + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; + case 0x14f15051: + add_cx5051_fake_mutes(codec); + break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); From e49a3434f1bc64dc49ff3a56e416bb5894868dde Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Mar 2012 18:14:41 +0100 Subject: [PATCH 11/11] ALSA: hda - Kill hyphenated names Kill hyphens from "Line-Out" name strings, as suggested by Mark Brown. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_cirrus.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 6 +++--- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0ae6eb20b13b..684307372d73 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5118,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5177,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line-Out", + return fill_audio_out_name(codec, nid, cfg, "Line Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index bc5a993d1146..c83ccdba1e5a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, "Front Speaker", "Surround Speaker", "Bass Speaker" }; static const char * const line_outs[] = { - "Front Line-Out", "Surround Line-Out", "Bass Line-Out" + "Front Line Out", "Surround Line Out", "Bass Line Out" }; fix_volume_caps(codec, dac); @@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, if (num_ctls > 1) name = line_outs[idx]; else - name = "Line-Out"; + name = "Line Out"; break; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca117cf5132f..d29d6d377904 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4fe2d5960a04..f286bb8fda13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -802,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -1856,7 +1856,7 @@ static const char * const alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "Line-Out Playback Volume", + "Line Out Playback Volume", "CLFE Playback Volume", "Bass Speaker Playback Volume", "PCM Playback Volume", @@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", - "Line-Out Playback Switch", + "Line Out Playback Switch", "CLFE Playback Switch", "Bass Speaker Playback Switch", "PCM Playback Switch",