From 8ac60a6866e58b861f2a15689c6513faf1602a3d Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 29 May 2013 20:01:19 +0200 Subject: ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 030f53c96ec..756c204b62d 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 0, 0x7f, 0x19, hl_tlv), + 0, 0x7f, 0x19, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), -- cgit v1.2.3-70-g09d2 From 7d8acf2cba81d7c64842b5dac0d7b3dae16f0378 Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 29 May 2013 20:01:20 +0200 Subject: ASoC: cs42l52: fix hp_gain_enum shift value. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 756c204b62d..987f728718c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -262,7 +262,7 @@ static const char * const hp_gain_num_text[] = { }; static const struct soc_enum hp_gain_enum = - SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5, ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); static const char * const beep_pitch_text[] = { -- cgit v1.2.3-70-g09d2 From 9e43088bb015397930d6c9ea5edba92abc0dc655 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 May 2013 18:38:46 +0100 Subject: ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index dfd997aaadf..19e0b2048af 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3836,7 +3836,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) ret); } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); - return IRQ_HANDLED; + goto out; } } else if (!(reg & WM8958_MICD_STS)) { snd_soc_jack_report(wm8994->micdet[0].jack, 0, -- cgit v1.2.3-70-g09d2 From 7afce3f5e56e9cb97cf1f35832bf8e8dde08cc45 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 May 2013 13:42:27 +0100 Subject: ASoC: wm8994: Ensure microphone detection state is reset on removal Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 19e0b2048af..29e95f93d48 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3842,6 +3842,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, 0, SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + wm8994->mic_detecting = true; goto out; } -- cgit v1.2.3-70-g09d2 From 056790923e1c4eed5d8cc502e1092944a2b23025 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 1 Jun 2013 23:13:53 +0100 Subject: ASoC: pcm: Require both CODEC and CPU support when declaring stream caps When declaring playback and capture capabilities check for both CODEC side and CPU side support rather than only checking for CODEC side support. While it is unusual some CPUs do have unidirectional DAIs. Reported-by: Fabio Estevam Tested-by: Fabio Estevam Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8eefa49..a9fddf0fea1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2011,9 +2011,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- cgit v1.2.3-70-g09d2 From ee4b7c7fe0c50b97d074f9185dba9558d9440c21 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Jun 2013 14:51:30 +0100 Subject: ASoC: arizona: Correct AEC loopback enable Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d3939ee..100fdadda56 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ba38f067966..88ad7db52dd 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5110_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), -- cgit v1.2.3-70-g09d2 From 4616274d3382fa7698536d61b351e63cf0ce27f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 5 Jun 2013 19:36:11 +0100 Subject: ASoC: dapm: Treat DAI widgets like AIF widgets for power Even though they are virtual widgets DAI widgets still get counted for the DAPM context power management so we can't just use the active state to check if they should be powered as they may not be part of a complete path. Instead split them into input and output widgets and do the same power checks as we perform on AIFs. Reported-by: Stephen Warren Tested-by: Stephen Warren Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 ++- sound/soc/soc-dapm.c | 49 +++++++++++++++++++++++++----------------------- sound/soc/soc-pcm.c | 7 ++++++- 3 files changed, 34 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d4609029f01..385c6329a96 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -450,7 +450,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ - snd_soc_dapm_dai, /* link to DAI structure */ + snd_soc_dapm_dai_in, /* link to DAI structure */ + snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8b..c7051c457b7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai] = 3, + [snd_soc_dapm_dai_in] = 3, + [snd_soc_dapm_dai_out] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -92,7 +93,8 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, - [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_dai_in] = 10, + [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, @@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_out: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) -{ - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) - return w->active; - - return dapm_generic_check_power(w); -} - /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); @@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; - case snd_soc_dapm_dai: - w->power_check = dapm_dai_check_power; - break; default: w->power_check = dapm_always_on_check_power; break; @@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.reg = SND_SOC_NOPM; if (dai->driver->playback.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_in; template.name = dai->driver->playback.stream_name; template.sname = dai->driver->playback.stream_name; @@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } if (dai->driver->capture.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_out; template.name = dai->driver->capture.stream_name; template.sname = dai->driver->capture.stream_name; @@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { - if (dai_w->id != snd_soc_dapm_dai) + switch (dai_w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } dai = dai_w->priv; @@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (w->dapm != dai_w->dapm) continue; - if (w->id == snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: continue; + default: + break; + } if (!w->sname) continue; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a9fddf0fea1..ccb6be4d658 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Create any new FE <--> BE connections */ for (i = 0; i < list->num_widgets; i++) { - if (list->widgets[i]->id != snd_soc_dapm_dai) + switch (list->widgets[i]->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, list->widgets[i], stream); -- cgit v1.2.3-70-g09d2 From 2894770ec17ff732f911c8495ae0504f06a5dad5 Mon Sep 17 00:00:00 2001 From: Andreas Irestål Date: Wed, 5 Jun 2013 08:49:47 +0200 Subject: ASoC: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is prevented. This problem was introduced in Linux 3.5 Signed-off-by: Andreas Irestål Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d60b7c..1514bf845e4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, break; } - - if (found) - snd_soc_dapm_sync(widget->dapm); } - ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); - mutex_unlock(&widget->codec->mutex); + + if (found) + snd_soc_dapm_sync(widget->dapm); + + ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); return ret; } -- cgit v1.2.3-70-g09d2 From dd6c5cd8fedddc9605209098e2fa4e82c7af22aa Mon Sep 17 00:00:00 2001 From: Al Viro Date: Wed, 5 Jun 2013 14:07:08 -0400 Subject: snd_pcm_link(): fix a leak... in case when snd_pcm_stream_linked(substream) is true, we end up leaking group. Signed-off-by: Al Viro --- sound/core/pcm_native.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ccfa383f1fd..f9281815595 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1649,6 +1649,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) } if (!snd_pcm_stream_linked(substream)) { substream->group = group; + group = NULL; spin_lock_init(&substream->group->lock); INIT_LIST_HEAD(&substream->group->substreams); list_add_tail(&substream->link_list, &substream->group->substreams); @@ -1663,8 +1664,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) _nolock: snd_card_unref(substream1->pcm->card); fput_light(file, fput_needed); - if (res < 0) - kfree(group); + kfree(group); return res; } -- cgit v1.2.3-70-g09d2 From d81bf8cf549f7a6656a64924672a42c101d17026 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 11 Jun 2013 11:06:48 +0200 Subject: ALSA: hda - Headset mic support for three more machines They need these quirks to have headset mic support. BugLink: https://bugs.launchpad.net/bugs/1189363 Tested-by: Shawn Wang Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 02e22b4458d..af9e71bec8b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3483,6 +3483,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3494,6 +3495,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), -- cgit v1.2.3-70-g09d2 From 342cda29343a6272c630f94ed56810a76740251b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 15 Jun 2013 11:21:09 +0200 Subject: ALSA: usb-audio: work around Android accessory firmware bug When the Android firmware enables the audio interfaces in accessory mode, it always declares in the control interface's baInterfaceNr array that interfaces 0 and 1 belong to the audio function. However, the accessory interface itself, if also enabled, already is at index 0 and shifts the actual audio interface numbers to 1 and 2, which prevents the PCM streaming interface from being seen by the host driver. To get the PCM interface interface to work, detect when the descriptors point to the (for this driver useless) accessory interface, and redirect to the correct one. Reported-by: Jeremy Rosen Tested-by: Jeremy Rosen Cc: Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 1a033177b83..64952e2d3ed 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -147,14 +147,32 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return -EINVAL; } + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + + /* + * Android with both accessory and audio interfaces enabled gets the + * interface numbers wrong. + */ + if ((chip->usb_id == USB_ID(0x18d1, 0x2d04) || + chip->usb_id == USB_ID(0x18d1, 0x2d05)) && + interface == 0 && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + altsd->bInterfaceSubClass == USB_SUBCLASS_VENDOR_SPEC) { + interface = 2; + iface = usb_ifnum_to_if(dev, interface); + if (!iface) + return -EINVAL; + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + } + if (usb_interface_claimed(iface)) { snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, interface); return -EINVAL; } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { -- cgit v1.2.3-70-g09d2 From 6ab982e8cf8e5760da407ccdc4abc815bea23179 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Jun 2013 10:19:49 +0200 Subject: ALSA: hda - Fix pin configurations for MacBook Air 4,2 MacBook Air 4,2 requires the whole default pin configuration table to be overridden by the driver, as usual, as Apple's machines don't set up properly after boot. Otherwise mic won't work, and other ill effect may happen. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59381 Reported-and-tested-by: Peter John Hartman Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index bd8d46cca2b..cccaf9c7a7b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -58,6 +58,7 @@ enum { CS420X_GPIO_23, CS420X_MBP101, CS420X_MBP81, + CS420X_MBA42, CS420X_AUTO, /* aliases */ CS420X_IMAC27_122 = CS420X_GPIO_23, @@ -346,6 +347,7 @@ static const struct hda_model_fixup cs420x_models[] = { { .id = CS420X_APPLE, .name = "apple" }, { .id = CS420X_MBP101, .name = "mbp101" }, { .id = CS420X_MBP81, .name = "mbp81" }, + { .id = CS420X_MBA42, .name = "mba42" }, {} }; @@ -361,6 +363,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), + SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; @@ -414,6 +417,20 @@ static const struct hda_pintbl mbp101_pincfgs[] = { {} /* terminator */ }; +static const struct hda_pintbl mba42_pincfgs[] = { + { 0x09, 0x012b4030 }, /* HP */ + { 0x0a, 0x400000f0 }, + { 0x0b, 0x90100120 }, /* speaker */ + { 0x0c, 0x400000f0 }, + { 0x0d, 0x90a00110 }, /* mic */ + { 0x0e, 0x400000f0 }, + { 0x0f, 0x400000f0 }, + { 0x10, 0x400000f0 }, + { 0x12, 0x400000f0 }, + { 0x15, 0x400000f0 }, + {} /* terminator */ +}; + static void cs420x_fixup_gpio_13(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -482,6 +499,12 @@ static const struct hda_fixup cs420x_fixups[] = { .chained = true, .chain_id = CS420X_GPIO_13, }, + [CS420X_MBA42] = { + .type = HDA_FIXUP_PINS, + .v.pins = mba42_pincfgs, + .chained = true, + .chain_id = CS420X_GPIO_13, + }, }; static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid) -- cgit v1.2.3-70-g09d2 From 36691e1be6ec551eef4a5225f126a281f8c051c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Jun 2013 10:25:02 +0200 Subject: ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310 Just like the previous fix for LogitechHD Webcam c270 in commit 11e7064f35bb87da8f427d1aa4bbd8b7473a3993, c310 model also requires the same workaround for avoiding the kernel warning. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59741 Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e5c7f9f20fd..d5438083fd6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -885,6 +885,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ case USB_ID(0x046d, 0x0991): -- cgit v1.2.3-70-g09d2