From 77c4d5cdb81d25a45fbdfb84dd3348121219a072 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Thu, 2 Dec 2010 22:45:45 -0500 Subject: [PATCH 01/14] ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture BugLink: https://launchpad.net/bugs/595482 The original reporter states that audible playback from the internal speaker is inaudible despite the hardware being properly detected. To work around this symptom, he uses the model=lg quirk to properly enable both playback, capture, and jack sense. Another user corroborates this workaround on separate hardware. Add this PCI SSID to the quirk table to enable it for further LG P1 Expresses. Reported-and-tested-by: Philip Peitsch Tested-by: nikhov Cc: [2.6.32+] Signed-off-by: Daniel T Chen 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 8fddc9d08726..2d7d7de8498a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4595,6 +4595,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), + SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG), SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), From 3a253445e327d0295bbe51bcbec5f44f86b54a8c Mon Sep 17 00:00:00 2001 From: John Baboval Date: Thu, 2 Dec 2010 11:21:31 -0500 Subject: [PATCH 02/14] ALSA: hda - Fix ThinkPad T410[s] docking station line-out On the docking station for the Lenovo T410 and T410s, the line-out doesn't work. The trouble seems to be that it generates a plug event, but then doesn't report that the jack is connected. So automute mutes the jack when you plug something into it. The following patch (next message) fixes it. Signed-off-by: John Baboval Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 846d1ead47fd..e652b34b1bd9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2116,8 +2116,8 @@ static void cxt5066_update_speaker(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int pinctl; - snd_printdd("CXT5066: update speaker, hp_present=%d\n", - spec->hp_present); + snd_printdd("CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n", + spec->hp_present, spec->cur_eapd); /* Port A (HP) */ pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0; @@ -2125,11 +2125,20 @@ static void cxt5066_update_speaker(struct hda_codec *codec) pinctl); /* Port D (HP/LO) */ - pinctl = ((spec->hp_present & 2) && spec->cur_eapd) - ? spec->port_d_mode : 0; - /* Mute if Port A is connected on Thinkpad */ - if (spec->thinkpad && (spec->hp_present & 1)) - pinctl = 0; + if (spec->dell_automute) { + /* DELL AIO Port Rule: PortA> PortD> IntSpk */ + pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) + ? PIN_OUT : 0; + } else if (spec->thinkpad) { + if (spec->cur_eapd) + pinctl = spec->port_d_mode; + /* Mute dock line-out if Port A (laptop HP) is present */ + if (spec->hp_present& 1) + pinctl = 0; + } else { + pinctl = ((spec->hp_present & 2) && spec->cur_eapd) + ? spec->port_d_mode : 0; + } snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); @@ -2137,14 +2146,6 @@ static void cxt5066_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - - if (spec->dell_automute) { - /* DELL AIO Port Rule: PortA > PortD > IntSpk */ - pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) - ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - } } /* turn on/off EAPD (+ mute HP) as a master switch */ From ef61d4e6d88da80dc2b417cf8ad3c77aa94f0c8f Mon Sep 17 00:00:00 2001 From: Manoj Iyer Date: Fri, 3 Dec 2010 18:43:55 -0600 Subject: [PATCH 03/14] ALSA: hda - Enable jack sense for Thinkpad Edge 13 Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13. Reference: http://launchpad.net/bugs/685015 Signed-off-by: Manoj Iyer Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e652b34b1bd9..4ab5ea9f0530 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3110,6 +3110,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD), From dd5a089edfa51a74692604b4b427953d8e16bc35 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 5 Dec 2010 08:43:14 -0500 Subject: [PATCH 04/14] ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic BugLink: https://launchpad.net/bugs/685161 The reporter of the bug states that he must use position_fix=1 to enable capture for the internal microphone, so set it for his machine's PCI SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build. Reported-and-tested-by: Ralph Wabel Cc: Signed-off-by: Daniel T Chen 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 21aa9b0e28f6..b030c8eba21f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2296,6 +2296,7 @@ static int azx_dev_free(struct snd_device *device) */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), From 3fcc0afbb9c93f3599ba03273e59915670b6c2c2 Mon Sep 17 00:00:00 2001 From: Uk Kim Date: Sun, 5 Dec 2010 17:32:16 +0900 Subject: [PATCH 05/14] ASoC: Fix off by one error in WM8994 EQ register bank size Signed-off-by: Uk Kim Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- include/linux/mfd/wm8994/pdata.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 5c51f367c061..add8a1b8bcf0 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -29,7 +29,7 @@ struct wm8994_ldo_pdata { #define WM8994_CONFIGURE_GPIO 0x8000 #define WM8994_DRC_REGS 5 -#define WM8994_EQ_REGS 19 +#define WM8994_EQ_REGS 20 /** * DRC configurations are specified with a label and a set of register From ed8cc471d75365f8590c76f580def899d58028c0 Mon Sep 17 00:00:00 2001 From: Uk Kim Date: Sun, 5 Dec 2010 17:26:07 +0900 Subject: [PATCH 06/14] ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit. Signed-off-by: Uk Kim Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 19ca782ac970..0e24092722c3 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -293,7 +293,7 @@ SOC_DOUBLE_R("Speaker Switch", SOC_DOUBLE_R("Speaker ZC Switch", WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT, 7, 1, 0), -SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 0, 3, 7, 0, +SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0, spkboost_tlv), SOC_ENUM("Speaker Reference", speaker_ref), SOC_ENUM("Speaker Mode", speaker_mode), From 1dcb4f38e5bc28dfce0f8c7eef184a090b03bfc7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 6 Dec 2010 16:48:03 +0800 Subject: [PATCH 07/14] ASoC: Hold client_mutex while calling snd_soc_instantiate_cards() As the comments of snd_soc_instantiate_cards() said, snd_soc_instantiate_cards() must be called with client_mutex. Signed-off-by: Axel Lin 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 441285ade024..02ae7bea3b50 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3072,7 +3072,9 @@ int snd_soc_register_dais(struct device *dev, pr_debug("Registered DAI '%s'\n", dai->name); } + mutex_lock(&client_mutex); snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); return 0; err: From 681e36924788aeea2717c07cc42a21c9c86d7559 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 7 Dec 2010 20:56:30 +0800 Subject: [PATCH 08/14] ASoC: Fix resource leak if soc_register_ac97_dai_link failed Properly free the resources in the case of soc_register_ac97_dai_link failure. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 02ae7bea3b50..85b7d548f167 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1619,12 +1619,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #ifdef CONFIG_SND_SOC_AC97_BUS /* register any AC97 codecs */ for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); - goto probe_dai_err; - } + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + goto probe_dai_err; } + } #endif card->instantiated = 1; From 6b464321d276e448d478c99202c19d83f2bd25f4 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Tue, 7 Dec 2010 19:23:07 +0900 Subject: [PATCH 09/14] ASoC: WM8580: Debug BCLK and sample size In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32, rather than WM8580_AIF_LENGTH_24. Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index a2e0ed59b376..879dff2714dd 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -491,16 +491,16 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, paifa |= 0x8; break; case SNDRV_PCM_FORMAT_S20_3LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_20; break; case SNDRV_PCM_FORMAT_S24_LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_24; break; case SNDRV_PCM_FORMAT_S32_LE: - paifa |= 0x10; - paifb |= WM8580_AIF_LENGTH_24; + paifa |= 0x0; + paifb |= WM8580_AIF_LENGTH_32; break; default: return -EINVAL; From 2a7b1a00206895cfa444fd83477dca67a88a9d25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Dec 2010 15:32:38 +0000 Subject: [PATCH 10/14] ASoC: Correct WM8962 interrupt mask register read Fix mismerge from the out of tree BSP where this support was developed. Signed-off-by: Mark Brown --- 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 e8092745a207..1304ca91a11c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3339,7 +3339,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) int mask; int active; - mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); + mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; From 4b0dbdb17f846a8887e5f7fbeea2deb0703236bd Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 7 Dec 2010 18:41:35 +0200 Subject: [PATCH 11/14] ALSA: hda - Do not wrongly restrict min_channels based on ELD Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, it wrongly assumes that the bits 0-2 of the first byte of CEA Short Audio Descriptors mean a supported number of channels. In reality, they mean the maximum number of channels (as per CEA-861-D 7.5.2). This means that the channel count can only be used to restrict max_channels, not min_channels. Restricting min_channels causes us to deny opening the device in stereo mode if the sink only has SADs that declare larger numbers of channels (like Primare SP32 AV Processor does). Fix that by not restricting min_channels based on ELD information. Signed-off-by: Anssi Hannula Reported-by: Jean-Yves Avenard Tested-by: Jean-Yves Avenard Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 ---- sound/pci/hda/patch_hdmi.c | 1 - 2 files changed, 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index cb0c23a6b473..47ef8aa4a844 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -601,13 +601,10 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->rates = 0; pcm->formats = 0; pcm->maxbps = 0; - pcm->channels_min = -1; pcm->channels_max = 0; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; - if (a->channels < pcm->channels_min) - pcm->channels_min = a->channels; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, /* restrict the parameters by the values the codec provides */ pcm->rates &= codec_pars->rates; pcm->formats &= codec_pars->formats; - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3e49aa5b9ec..31df7747990d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, return -ENODEV; } else { /* fallback to the codec default */ - hinfo->channels_min = codec_pars->channels_min; hinfo->channels_max = codec_pars->channels_max; hinfo->rates = codec_pars->rates; hinfo->formats = codec_pars->formats; From 3dc86429032910bdf762adeb2969112bb303924c Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 7 Dec 2010 20:56:19 +0200 Subject: [PATCH 12/14] ALSA: hda - Always allow basic audio irrespective of ELD info Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, according to CEA-861-D no SAD is needed for basic audio (32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a basic audio flag in the CEA EDID Extension. The flag is not present in ELD. However, as all audio capable sinks are required to support basic audio, we can assume it to be always available. Fix allowed audio formats with sinks that have SADs (Short Audio Descriptors) which do not completely overlap with the basic audio formats (there are no reports of affected devices so far) by always assuming that basic audio is supported. Reported-by: Stephen Warren Signed-off-by: Anssi Hannula Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 47ef8aa4a844..009031fae2ba 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -598,21 +598,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, { int i; - pcm->rates = 0; - pcm->formats = 0; - pcm->maxbps = 0; - pcm->channels_max = 0; + /* assume basic audio support (the basic audio flag is not in ELD; + * however, all audio capable sinks are required to support basic + * audio) */ + pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + pcm->formats = SNDRV_PCM_FMTBIT_S16_LE; + pcm->maxbps = 16; + pcm->channels_max = 2; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { - if (a->sample_bits & AC_SUPPCM_BITS_16) { - pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE; - if (pcm->maxbps < 16) - pcm->maxbps = 16; - } if (a->sample_bits & AC_SUPPCM_BITS_20) { pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; if (pcm->maxbps < 20) From 0bbaee3a58c379c4f7bab9635c71d7bad9c422a2 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 7 Dec 2010 21:19:23 +0200 Subject: [PATCH 13/14] ALSA: hda - Reset sample sizes and max bitrates when reading ELD When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc() is called for every SAD (Short Audio Descriptor) in the ELD data. For LPCM coding type SAD defines the supported sample sizes. For several other coding types (such as AC-3), a maximum bitrate is defined. The maximum bitrate and sample size fields are not always cleared. Therefore, if a device is unplugged and a different one is plugged in, and the coding types of some SAD positions differ between the devices, the old max_bitrate or sample_bits values will persist if the new SADs do not define those values. The leftover max_bitrate and sample_bits do not cause any issues other than wrongly showing up in eld#X.Y procfs file and kernel log. Fix that by always clearing sample_bits and max_bitrate when reading SADs. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 009031fae2ba..4a663471dadc 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->channels = GRAB_BITS(buf, 0, 0, 3); a->channels++; + a->sample_bits = 0; + a->max_bitrate = 0; + a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: @@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, case AUDIO_CODING_TYPE_LPCM: val = GRAB_BITS(buf, 2, 0, 3); - a->sample_bits = 0; for (i = 0; i < 3; i++) if (val & (1 << i)) a->sample_bits |= cea_sample_sizes[i + 1]; From 8a96b1e02029aa512199b1b6d281dcede9ed81f1 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 9 Dec 2010 07:17:27 +0100 Subject: [PATCH 14/14] ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work BugLink: http://launchpad.net/497546 Confirmed that the ideapad model works better than the current quirk for Dell Vostro 320. Cc: stable@kernel.org (2.6.35+) Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4ab5ea9f0530..76bd58a0e2b6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3096,8 +3096,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", - CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),