From 4d8ec5f3b65dd64fa785192dc7ab2807916a05b2 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Thu, 19 Aug 2010 08:06:16 +0200 Subject: ALSA: hda - Add support for IDT 92HD89XX codecs Just added new codec ids. These are almost compatible with existing ones. Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3f861bd1bf..95148e58026 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -6303,6 +6303,21 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, {} /* terminator */ }; -- cgit v1.2.3-70-g09d2 From 274714f55c023c683a6b2deedfb2209a9457f4ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Aug 2010 08:11:53 +0200 Subject: ALSA: hda - Fix build error with CONFIG_PROC_FS=n hdmi_eld_update_pcm_info() must be always compiled in. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 803b298f741..26c3ade7358 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -596,6 +596,8 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) } EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free); +#endif /* CONFIG_PROC_FS */ + /* update PCM info based on ELD */ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, struct hda_pcm_stream *codec_pars) @@ -644,5 +646,3 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info); - -#endif /* CONFIG_PROC_FS */ -- cgit v1.2.3-70-g09d2 From 9c77b846ec8b4e0c7107dd7f820172462dc84a61 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 18 Aug 2010 19:33:43 -0400 Subject: ALSA: intel8x0: Mute External Amplifier by default for ThinkPad X31 BugLink: https://bugs.launchpad.net/bugs/619439 This ThinkPad model needs External Amplifier muted for audible playback, so set the inv_eapd quirk for it. Reported-and-tested-by: Dennis Bell Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6433e65c950..46774924957 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1774,6 +1774,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP/Compaq nx7010", .type = AC97_TUNE_MUTE_LED }, + { + .subvendor = 0x1014, + .subdevice = 0x0534, + .name = "ThinkPad X31", + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x1014, .subdevice = 0x1f00, -- cgit v1.2.3-70-g09d2 From 4f34760787c3751a3146f0eecdc79c3e97b94962 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Aug 2010 09:41:59 +0200 Subject: ALSA: hda - Fix conflict of sticky PCM parameter in HDMI codecs Intel and Nvidia HDMI codec drivers have own implementations of sticky PCM parameters. Now HD-audio core part already has it, thus both setups conflict. The fix is simply remove the part in patch_intelhdmi.c and patch_nvhdmi.c and simply call snd_hda_codec_setup_stream() as usual. Reported-and-tested-by: Stephen Warren Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 21 +-------------------- sound/pci/hda/patch_intelhdmi.c | 8 -------- sound/pci/hda/patch_nvhdmi.c | 8 -------- 3 files changed, 1 insertion(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2bc0f07cf33..afd6022a96a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -707,8 +707,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; - int tag; - int fmt; int pinctl; int new_pinctl = 0; int i; @@ -745,24 +743,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, return -EINVAL; } - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index d382d3c81c0..36a9b83a617 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -69,20 +69,12 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } -static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, .ops = { .open = hdmi_pcm_open, .prepare = intel_hdmi_playback_pcm_prepare, - .cleanup = intel_hdmi_playback_pcm_cleanup, }, }; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index f636870dc71..69b950d527c 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -326,13 +326,6 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, return 0; } -static int nvhdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -350,7 +343,6 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = { .ops = { .open = hdmi_pcm_open, .prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89, - .cleanup = nvhdmi_playback_pcm_cleanup, }, }; -- cgit v1.2.3-70-g09d2 From 3f50ac6a0ec80a83a1a033fe5004fb319ad72db7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Aug 2010 09:44:36 +0200 Subject: ALSA: hda - Fix stream and channel-ids codec-bus wide The new sticky PCM parameter introduced the delayed clean-ups of stream- and channel-id tags. In the current implementation, this check (adding dirty flag) and actual clean-ups are done only for the codec chip. However, with HD-audio architecture, multiple codecs can be on a single bus, and the controller assign stream- and channel-ids in the bus-wide. In this patch, the stream-id and channel-id are checked over all codecs connected to the corresponding bus. Together with it, the mutex is moved to struct hda_bus, as this becomes also bus-wide. Reported-and-tested-by: Stephen Warren Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++++++------------- sound/pci/hda/hda_codec.h | 2 +- 2 files changed, 21 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index dd8fb86c842..3827092cc1d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -589,6 +589,7 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); + mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); snprintf(bus->workq_name, sizeof(bus->workq_name), @@ -1068,7 +1069,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); - mutex_init(&codec->prepare_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); @@ -1213,6 +1213,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format) { + struct hda_codec *c; struct hda_cvt_setup *p; unsigned int oldval, newval; int i; @@ -1253,10 +1254,12 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p->dirty = 0; /* make other inactive cvts with the same stream-tag dirty */ - for (i = 0; i < codec->cvt_setups.used; i++) { - p = snd_array_elem(&codec->cvt_setups, i); - if (!p->active && p->stream_tag == stream_tag) - p->dirty = 1; + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + p = snd_array_elem(&c->cvt_setups, i); + if (!p->active && p->stream_tag == stream_tag) + p->dirty = 1; + } } } EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); @@ -1306,12 +1309,16 @@ static void really_cleanup_stream(struct hda_codec *codec, /* clean up the all conflicting obsolete streams */ static void purify_inactive_streams(struct hda_codec *codec) { + struct hda_codec *c; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); - if (p->dirty) - really_cleanup_stream(codec, p); + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + struct hda_cvt_setup *p; + p = snd_array_elem(&c->cvt_setups, i); + if (p->dirty) + really_cleanup_stream(c, p); + } } } @@ -3502,11 +3509,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream) { int ret; - mutex_lock(&codec->prepare_mutex); + mutex_lock(&codec->bus->prepare_mutex); ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); if (ret >= 0) purify_inactive_streams(codec); - mutex_unlock(&codec->prepare_mutex); + mutex_unlock(&codec->bus->prepare_mutex); return ret; } EXPORT_SYMBOL_HDA(snd_hda_codec_prepare); @@ -3515,9 +3522,9 @@ void snd_hda_codec_cleanup(struct hda_codec *codec, struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { - mutex_lock(&codec->prepare_mutex); + mutex_lock(&codec->bus->prepare_mutex); hinfo->ops.cleanup(hinfo, codec, substream); - mutex_unlock(&codec->prepare_mutex); + mutex_unlock(&codec->bus->prepare_mutex); } EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4303353feda..62c70224010 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -648,6 +648,7 @@ struct hda_bus { struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; struct mutex cmd_mutex; + struct mutex prepare_mutex; /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; @@ -826,7 +827,6 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; - struct mutex prepare_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ -- cgit v1.2.3-70-g09d2 From 6f0ef6ea1d11ef242de584e345355b0de756fcb2 Mon Sep 17 00:00:00 2001 From: Jerone Young Date: Mon, 23 Aug 2010 08:34:36 +0200 Subject: ALSA: hda - Add support for Lenovo S10-3t This patch adds quirk for the Lenovo S10-3t so the headphone & microphone jacks will now work. Signed-off-by: Jerone Young Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c424952a734..5cdb80edbd7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3059,6 +3059,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} -- cgit v1.2.3-70-g09d2 From 70bf043b137aa9ff2711b16532774465e07a8f47 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Mon, 23 Aug 2010 08:54:02 +0200 Subject: ASoC: i.MX ssi: use SSI_STCCR in synchronous mode In synchronous mode the SSI_SRCCR values are ignored. Instead SSI_STCCR must be used for both receiving and transmitting. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index a11daa1e905..c81da05a4f1 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -254,6 +254,9 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, dma_data = &ssi->dma_params_rx; } + if (ssi->flags & IMX_SSI_SYN) + reg = SSI_STCCR; + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; -- cgit v1.2.3-70-g09d2 From dbbcbc073ad3132bfbc410b11546b2fb4bdf2568 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 23 Aug 2010 08:14:35 +0200 Subject: ALSA: hda - Add Sony VAIO quirk for ALC269 The attached patch enables playback on a Sony VAIO machine. BugLink: http://launchpad.net/bugs/618271 Signed-off-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a4dd04524e4..627bf996336 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14467,6 +14467,7 @@ static const struct alc_fixup alc269_fixups[] = { static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), {} }; -- cgit v1.2.3-70-g09d2 From 708fafb3c54039caa5dadc8e9d2dfd999f88f190 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 27 Aug 2010 10:34:27 +0800 Subject: ASoC: soc-core: fix debugfs_pop_time file permissions I think this is a typo, debugfs_pop_time should not be executable. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 844ae8221a3..acc91daa1c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -251,7 +251,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644, codec->debugfs_codec_root, &codec->pop_time); if (!codec->debugfs_pop_time) -- cgit v1.2.3-70-g09d2 From 60f1deb595c08687a96157a6a3ce08ef34142362 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Sat, 28 Aug 2010 19:52:24 +1200 Subject: ALSA: asihpi - Return hw error directly from oustream_write. If hw error is ignored, status is updated with invalid info. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3b441344822..22c5fc62553 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static u32 outstream_get_space_available(struct hpi_hostbuffer_status - *status) +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { return status->size_in_bytes - (status->host_index - status->dSP_index); @@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao, /* write it */ phm->function = HPI_OSTREAM_WRITE; hw_message(pao, phm, phr); + + if (phr->error) + return; + /* update status information that the DSP would typically * update (and will update next time the DSP * buffer update task reads data from the host BBM buffer) -- cgit v1.2.3-70-g09d2 From 3182c8a72b31414e043184a97b0d5d3c0d590168 Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Sat, 28 Aug 2010 13:25:33 +0900 Subject: sound: oss: fix uninitialized spinlock The spinlock lock in sound_timer.c is used without initialization. Signed-off-by: Akinobu Mita Signed-off-by: Takashi Iwai --- sound/oss/sound_timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index f0f0c19fbff..48cda6c4c25 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -26,7 +26,7 @@ static unsigned long prev_event_time; static volatile unsigned long usecs_per_tmr; /* Length of the current interval */ static struct sound_lowlev_timer *tmr; -static spinlock_t lock; +static DEFINE_SPINLOCK(lock); static unsigned long tmr2ticks(int tmr_value) { -- cgit v1.2.3-70-g09d2 From 7a28826ac73d31a379d93785d8fbd24ab492b0bd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 27 Aug 2010 22:02:15 +0200 Subject: ALSA: pcm: add more format names There were some new formats added in commit 15c0cee6c809 "ALSA: pcm: Define G723 3-bit and 5-bit formats". That commit increased SNDRV_PCM_FORMAT_LAST as well. My concern is that there are a couple places which do: for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { if (dummy->pcm_hw.formats & (1ULL << i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } I haven't tested these but it looks like if "i" were equal to SNDRV_PCM_FORMAT_G723_24 or higher then we might read past the end of the array. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815dfbdc..204af48c5cc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = { FORMAT(S18_3BE), FORMAT(U18_3LE), FORMAT(U18_3BE), + FORMAT(G723_24), + FORMAT(G723_24_1B), + FORMAT(G723_40), + FORMAT(G723_40_1B), }; const char *snd_pcm_format_name(snd_pcm_format_t format) { + if (format >= ARRAY_SIZE(snd_pcm_format_names)) + return "Unknown"; return snd_pcm_format_names[format]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); -- cgit v1.2.3-70-g09d2 From 048e78a5bc22c27410cb5ca9680c3c7ac400607f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 2 Sep 2010 08:35:47 +0200 Subject: ALSA: hda - Add a new hp-laptop model for Conexant 5066, tested on HP G60 This new model adds the following functionality to HP G60: - Automute of internal speakers - Autoswitch of internal/external mics - Remove SPDIF not physically present BugLink: http://launchpad.net/bugs/587388 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 57 ++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ce46fa1e643..37c6aad5e59 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5051 Conexant 5066 ============= laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5cdb80edbd7..4f0619908a3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -116,6 +116,7 @@ struct conexant_spec { unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; + unsigned int hp_laptop:1; unsigned int ext_mic_present; unsigned int recording; @@ -2299,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_hp_laptop_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 3); +} + + /* toggle input of built-in digital mic and mic jack appropriately order is: external mic -> dock mic -> interal mic */ static void cxt5066_thinkpad_automic(struct hda_codec *codec) @@ -2407,6 +2420,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_hp_laptop_automic(codec); + break; + } +} + /* unsolicited event for jack sensing */ static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) { @@ -2989,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { { } /* end */ }; + +static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -3004,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_ideapad_automic(codec); else if (spec->thinkpad) cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3031,6 +3068,7 @@ enum { CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3041,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_HP_LAPTOP] = "hp-laptop", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -3052,6 +3091,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), @@ -3116,6 +3156,23 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_HP_LAPTOP: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_hp_laptop; + spec->num_init_verbs++; + spec->hp_laptop = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; + case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- cgit v1.2.3-70-g09d2 From 7b6717e144de6592e614fd7fc3b914b6bf686a9d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 2 Sep 2010 17:13:15 +0800 Subject: ALSA: usb-audio: Assume first control interface is for audio For devices with more than one control interface, let's assume the first one contains the audio controls. Unfortunately, there is no field in any of the descriptors to tell us whether a control interface is for audio or MIDI controls, so a better check is not easy to implement. On a composite device with audio and MIDI functions, for example, the code currently overwrites chip->ctrl_intf, causing operations on the control interface to fail if they are issued after the device probe. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 9feb00c831a..b443a33d31c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -465,7 +465,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } - chip->ctrl_intf = alts; + /* + * For devices with more than one control interface, we assume the + * first contains the audio controls. We might need a more specific + * check here in the future. + */ + if (!chip->ctrl_intf) + chip->ctrl_intf = alts; if (err > 0) { /* create normal USB audio interfaces */ -- cgit v1.2.3-70-g09d2 From a2acad8298a42b7be684a32fafaf83332bba9c2b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Sep 2010 10:53:11 +0200 Subject: ALSA: usb-audio: fix detection of vendor-specific device protocol settings The Audio Class v2 support code in 2.6.35 added checks for the bInterfaceProtocol field. However, there are devices (usually those detected by vendor-specific quirks) that do not have one of the predefined values in this field, which made the driver reject them. To fix this regression, restore the old behaviour, i.e., assume that a device with an unknown bInterfaceProtocol field (other than UAC_VERSION_2) has more or less UAC-v1-compatible descriptors. [compile warning fixes by tiwai] Signed-off-by: Clemens Ladisch Cc: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 9 +++++---- sound/usb/clock.c | 3 +-- sound/usb/endpoint.c | 11 ++++++----- sound/usb/format.c | 22 ++++++++++------------ sound/usb/mixer.c | 10 +++++++++- sound/usb/pcm.c | 3 +-- 6 files changed, 32 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index b443a33d31c..32e4be8a187 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } switch (protocol) { + default: + snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n", + protocol); + /* fall through */ + case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1 = control_header; @@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) break; } - - default: - snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); - return -EINVAL; } return 0; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b853f8df794..7754a103454 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); case UAC_VERSION_2: return set_sample_rate_v2(chip, iface, alts, fmt, rate); } - - return -EINVAL; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 1a701f1e8f5..ef0a07e3484 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* get audio formats */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + case UAC_VERSION_1: { struct uac1_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno, as->bTerminalLink); continue; } - - default: - snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", - dev->devnum, iface_no, altno, protocol); - continue; } /* get format type */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 3a1375459c0..69148212aa7 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, u64 pcm_formats; switch (protocol) { - case UAC_VERSION_1: { + case UAC_VERSION_1: + default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; @@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format <<= 1; break; } - - default: - return -EINVAL; } pcm_formats = 0; @@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * audio class v2 uses class specific EP0 range requests for that. */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); @@ -392,10 +394,6 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, /* fp->channels is already set in this case */ ret = parse_audio_format_rates_v2(chip, fp); break; - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } if (fp->channels < 1) { @@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -456,10 +458,6 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, ret = parse_audio_format_rates_v2(chip, fp); break; } - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } return ret; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c166db0057d..3ed3901369c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, } host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol; + switch (get_iface_desc(host_iface)->bInterfaceProtocol) { + case UAC_VERSION_1: + default: + mixer->protocol = UAC_VERSION_1; + break; + case UAC_VERSION_2: + mixer->protocol = UAC_VERSION_2; + break; + } if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3634cedf930..3b5135c9306 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return init_pitch_v1(chip, iface, alts, fmt); case UAC_VERSION_2: return init_pitch_v2(chip, iface, alts, fmt); } - - return -EINVAL; } /* -- cgit v1.2.3-70-g09d2 From 4d155641c81203440da64c4633b4efaab75f63b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Sep 2010 11:58:30 +0200 Subject: ALSA: hda - Add quirk for Lenovo T400s Lenovo T400s requires the quirk to make automatic HP/mic switching working. Reported-by: Frank Becker Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f0619908a3..71f9d6475b0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3094,6 +3094,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", 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, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), -- cgit v1.2.3-70-g09d2 From 4c25b93223340deff73381cc47f9244fb379a74d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Sep 2010 13:37:10 +0200 Subject: ALSA: virtuoso: work around missing reset in the Xonar DS Windows driver For the WM8776 chip, this driver uses a different sample format and more features than the Windows driver. When rebooting from Linux into Windows, the latter driver does not reset the chip but assumes all its registers have their default settings, so we get garbled sound or, if the output happened to be muted before rebooting, no sound. To make that driver happy, hook our driver's cleanup function into the shutdown notifier and ensure that the chip gets reset. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Nathan Schagen Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_lib.c | 21 ++++++++++++++++++--- sound/pci/oxygen/virtuoso.c | 1 + sound/pci/oxygen/xonar_wm87x6.c | 1 + 4 files changed, 21 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 6147216af74..a3409edcfb5 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -155,6 +155,7 @@ void oxygen_pci_remove(struct pci_dev *pci); int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); int oxygen_pci_resume(struct pci_dev *pci); #endif +void oxygen_pci_shutdown(struct pci_dev *pci); /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index fad03d64e3a..7e93cf88443 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -519,16 +519,21 @@ static void oxygen_init(struct oxygen *chip) } } -static void oxygen_card_free(struct snd_card *card) +static void oxygen_shutdown(struct oxygen *chip) { - struct oxygen *chip = card->private_data; - spin_lock_irq(&chip->reg_lock); chip->interrupt_mask = 0; chip->pcm_running = 0; oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); spin_unlock_irq(&chip->reg_lock); +} + +static void oxygen_card_free(struct snd_card *card) +{ + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); if (chip->irq >= 0) free_irq(chip->irq, chip); flush_scheduled_work(); @@ -778,3 +783,13 @@ int oxygen_pci_resume(struct pci_dev *pci) } EXPORT_SYMBOL(oxygen_pci_resume); #endif /* CONFIG_PM */ + +void oxygen_pci_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); + chip->model.cleanup(chip); +} +EXPORT_SYMBOL(oxygen_pci_shutdown); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index f03a2f2cffe..06c863e86e3 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -95,6 +95,7 @@ static struct pci_driver xonar_driver = { .suspend = oxygen_pci_suspend, .resume = oxygen_pci_resume, #endif + .shutdown = oxygen_pci_shutdown, }; static int __init alsa_card_xonar_init(void) diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index dbc4b89d74e..0b89932fb8c 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -193,6 +193,7 @@ static void xonar_ds_init(struct oxygen *chip) static void xonar_ds_cleanup(struct oxygen *chip) { xonar_disable_output(chip); + wm8776_write(chip, WM8776_RESET, 0); } static void xonar_ds_suspend(struct oxygen *chip) -- cgit v1.2.3-70-g09d2 From fe6ce80ae25953d95ebaf9bce27b585218cda25c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Sep 2010 13:38:49 +0200 Subject: ALSA: virtuoso: fix setting of Xonar DS line-in/mic-in controls The Line and Mic inputs cannot be used at the same time, so the driver has to automatically disable one of them if both are set. However, it forgot to notify userspace about this change, so the mixer state would be inconsistent. To fix this, check if the other control gets muted, and send a notification event in this case. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Nathan Schagen Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 0b89932fb8c..b82c1cfa96f 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -53,6 +53,8 @@ struct xonar_wm87x6 { struct xonar_generic generic; u16 wm8776_regs[0x17]; u16 wm8766_regs[0x10]; + struct snd_kcontrol *line_adcmux_control; + struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; }; @@ -604,6 +606,7 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; struct xonar_wm87x6 *data = chip->model_data; + struct snd_kcontrol *other_ctl; unsigned int mux_bit = ctl->private_value; u16 reg; int changed; @@ -611,8 +614,18 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = data->wm8776_regs[WM8776_ADCMUX]; if (value->value.integer.value[0]) { - reg &= ~0x003; reg |= mux_bit; + /* line-in and mic-in are exclusive */ + mux_bit ^= 3; + if (reg & mux_bit) { + reg &= ~mux_bit; + if (mux_bit == 1) + other_ctl = data->line_adcmux_control; + else + other_ctl = data->mic_adcmux_control; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &other_ctl->id); + } } else reg &= ~mux_bit; changed = reg != data->wm8776_regs[WM8776_ADCMUX]; @@ -964,7 +977,13 @@ static int xonar_ds_mixer_init(struct oxygen *chip) err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; + if (!strcmp(ctl->id.name, "Line Capture Switch")) + data->line_adcmux_control = ctl; + else if (!strcmp(ctl->id.name, "Mic Capture Switch")) + data->mic_adcmux_control = ctl; } + if (!data->line_adcmux_control || !data->mic_adcmux_control) + return -ENXIO; BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { ctl = snd_ctl_new1(&lc_controls[i], chip); -- cgit v1.2.3-70-g09d2 From 76195fb096ca6db2f8bbaffb96e3025aaf1649a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 08:27:02 +0200 Subject: ALSA: usb - Release capture substream URBs properly Due to the wrong "return" in the loop, a capture substream won't be released at disconnection properly if the device is capture only and has no playback substream. This caused Oops occasionally at the device reconnection. Reported-by: Kim Minhyoung Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 32e4be8a187..4eabafa5b03 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -126,7 +126,7 @@ static void snd_usb_stream_disconnect(struct list_head *head) for (idx = 0; idx < 2; idx++) { subs = &as->substream[idx]; if (!subs->num_formats) - return; + continue; snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; } -- cgit v1.2.3-70-g09d2 From a769cbcf60cee51f4431c0938acd39e7e5b76b8d Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 7 Sep 2010 14:36:22 -0500 Subject: ALSA: hda - Add errata initverb sequence for CS42xx codecs Add init verb sequence for errata ER880C3 http://www.cirrus.com/en/pubs/errata/ER880C3.pdf Signed-off-by: Brian Austin Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 50 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4ef5efaaaef..488fd9ade1b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -972,6 +972,53 @@ static struct hda_verb cs_coef_init_verbs[] = { {} /* terminator */ }; +/* Errata: CS4207 rev C0/C1/C2 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf + * + * 6. At high temperature (TA > +85°C), the digital supply current (IVD) + * may be excessive (up to an additional 200 μA), which is most easily + * observed while the part is being held in reset (RESET# active low). + * + * Root Cause: At initial powerup of the device, the logic that drives + * the clock and write enable to the S/PDIF SRC RAMs is not properly + * initialized. + * Certain random patterns will cause a steady leakage current in those + * RAM cells. The issue will resolve once the SRCs are used (turned on). + * + * Workaround: The following verb sequence briefly turns on the S/PDIF SRC + * blocks, which will alleviate the issue. + */ + +static struct hda_verb cs_errata_init_verbs[] = { + {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ + {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x9999}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0xa412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0009}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Rx: D0 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Tx: D0 */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0x2412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0000}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0008}, + {0x11, AC_VERB_SET_PROC_STATE, 0x00}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */ + /*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */ + + {} /* terminator */ +}; + /* SPDIF setup */ static void init_digital(struct hda_codec *codec) { @@ -991,6 +1038,9 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; + /* init_verb sequence for C0/C1/C2 errata*/ + snd_hda_sequence_write(codec, cs_errata_init_verbs); + snd_hda_sequence_write(codec, cs_coef_init_verbs); if (spec->gpio_mask) { -- cgit v1.2.3-70-g09d2 From 080dc7bc2562615a5be0a705a9d1a8c24eb198d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 08:38:41 +0200 Subject: ALSA: hda - Enable PC-beep for EeePC with ALC269 codec EeePC 1001HAG has a similar problem like other ASUS machine, which doesn't set the codec SSID properly for indicating the beep capability. To enable PC-beep again, put this to the whitelist. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 627bf996336..bcbf9160ed8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5334,6 +5334,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), + SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; -- cgit v1.2.3-70-g09d2 From e4ee8dd8afcbcbe502fa8a3d3af6eb09c96dd806 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 09:58:12 +0200 Subject: ALSA: msnd-classic: Fix invalid cfg parameter The driver doesn't probe the device properly because of left-over cfg[] that isn't used at all for msnd-classic device. This is only for msnd- pinnacle. Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 5f3e68401f9..91d6023a63e 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -764,9 +764,9 @@ static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +#ifndef MSND_CLASSIC static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; -#ifndef MSND_CLASSIC /* Extra Peripheral Configuration (Default: Disable) */ static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -894,7 +894,11 @@ static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) struct snd_card *card; struct snd_msnd *chip; - if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) { + if (has_isapnp(idx) +#ifndef MSND_CLASSIC + || cfg[idx] == SNDRV_AUTO_PORT +#endif + ) { printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); return -ENODEV; } -- cgit v1.2.3-70-g09d2 From 27f7ad53829f79e799a253285318bff79ece15bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Sep 2010 09:13:45 +0200 Subject: ALSA: seq/oss - Fix double-free at error path of snd_seq_oss_open() The error handling in snd_seq_oss_open() has several bad codes that do dereferecing released pointers and double-free of kmalloc'ed data. The object dp is release in free_devinfo() that is called via private_free callback. The rest shouldn't touch this object any more. The patch changes delete_port() to call kfree() in any case, and gets rid of unnecessary calls of destructors in snd_seq_oss_open(). Fixes CVE-2010-3080. Reported-and-tested-by: Tavis Ormandy Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 685712276ac..69cd7b3c362 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -281,13 +281,10 @@ snd_seq_oss_open(struct file *file, int level) return 0; _error: - snd_seq_oss_writeq_delete(dp->writeq); - snd_seq_oss_readq_delete(dp->readq); snd_seq_oss_synth_cleanup(dp); snd_seq_oss_midi_cleanup(dp); - delete_port(dp); delete_seq_queue(dp->queue); - kfree(dp); + delete_port(dp); return rc; } @@ -350,8 +347,10 @@ create_port(struct seq_oss_devinfo *dp) static int delete_port(struct seq_oss_devinfo *dp) { - if (dp->port < 0) + if (dp->port < 0) { + kfree(dp); return 0; + } debug_printk(("delete_port %i\n", dp->port)); return snd_seq_event_port_detach(dp->cseq, dp->port); -- cgit v1.2.3-70-g09d2 From 122661b67899980f1372812d907e73ebcfb3d037 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 14:57:04 +0200 Subject: ALSA: hda - Fix wrong HP pin detection in snd_hda_parse_pin_def_config() snd_hda_parse_pin_def_config() has some workaround for re-assigning some pins declared as headphones to line-outs. This didn't work properly for some cases because it used memmove() stupidly wrongly. Reference: Novell bnc#637263 https://bugzilla.novell.com/show_bug.cgi?id=637263 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3827092cc1d..14829210ef0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4536,7 +4536,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_outs--; memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); - memmove(sequences_hp + i - 1, sequences_hp + i, + memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } } -- cgit v1.2.3-70-g09d2 From a7a13d0676335a7dc9dd72264cca02606e43aaba Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 9 Sep 2010 00:11:41 +0200 Subject: ALSA: rawmidi: fix the get next midi device ioctl If we pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then the "next device" should be -1. This function just returns device + 1. But the main thing is that "device + 1" can lead to a (harmless) integer overflow and that annoys static analysis tools. [fix the case for device == SNDRV_RAWMIDI_DEVICE by tiwai] Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index eb68326c37d..a7868ad4d53 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -829,6 +829,8 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; + if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ + device = SNDRV_RAWMIDI_DEVICES - 1; mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_RAWMIDI_DEVICES) { -- cgit v1.2.3-70-g09d2 From cea310e8f8702226f982f09386cfd3c5793c5e2f Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Fri, 10 Sep 2010 16:29:56 -0700 Subject: ALSA: hda_intel: ALSA HD Audio patch for Intel Patsburg DeviceIDs This patch adds the Intel Patsburg (PCH) HD Audio Controller DeviceIDs. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1053fff4bd0..34940a07905 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -126,6 +126,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH10}," "{Intel, PCH}," "{Intel, CPT}," + "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2749,6 +2750,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH }, /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, + /* PBG */ + { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v1.2.3-70-g09d2 From 2ca9cac965e81da4b74f2dcec4b87ebfd106b357 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Fri, 10 Sep 2010 15:47:55 +0200 Subject: ALSA: hda - Add quirk for Toshiba C650D using a Conexant CX20585 Add a quirk for laptop Toshiba Satellite C650D to have proper external HP and external Mic support. Signed-off-by: Anisse Astier Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71f9d6475b0..972e7c453b3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3092,6 +3092,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), -- cgit v1.2.3-70-g09d2 From 147fcf1c211f1a87bf4d0711b7e9637f3d6ce080 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 11 Sep 2010 22:10:59 -0700 Subject: sound: Remove pr_ uses of KERN_ Signed-off-by: Joe Perches Acked-by: Mark Brown Acked-by: Geoff Levand Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 2 +- sound/soc/s3c24xx/s3c-dma.c | 3 +-- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 2f12da4da56..581a670e826 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -579,7 +579,7 @@ static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream, rate * delay_ms / 1000) * substream->runtime->channels; - pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", + pr_debug("%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", __func__, delay_ms, rate, diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 1b61c23ff30..f1b1bc4bacf 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -94,8 +94,7 @@ static void s3c_dma_enqueue(struct snd_pcm_substream *substream) if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); + pr_debug("%s: corrected dma len %ld\n", __func__, len); } ret = s3c2410_dma_enqueue(prtd->params->channel, -- cgit v1.2.3-70-g09d2 From 3894335876a6257ac46e14845bd37ae6fb0f7c87 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 14 Sep 2010 10:48:59 -0600 Subject: ALSA: patch_nvhdmi.c: Fix supported sample rate list. 22050 isn't a valid HDMI sample rate. 32000 is. Signed-off-by: Stephen Warren Acked-By: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 69b950d527c..baa108b9d6a 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -84,7 +84,7 @@ static struct hda_verb nvhdmi_basic_init_7x[] = { #else /* support all rates and formats */ #define SUPPORTED_RATES \ - (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 -- cgit v1.2.3-70-g09d2 From 145a902bfeb1f89a41165bd2d1e633ce070bcb73 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Sep 2010 10:07:53 +0200 Subject: ALSA: HDA: Enable internal speaker on Dell M101z BugLink: http://launchpad.net/bugs/640254 In some cases a magic processing coefficient is needed to enable the internal speaker on Dell M101z. According to Realtek, this processing coefficient is only present on ALC269vb. Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bcbf9160ed8..a1312a6c8af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14453,6 +14453,7 @@ static void alc269_auto_init(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, + ALC269_FIXUP_DELL_M101Z, }; static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = { @@ -14464,11 +14465,20 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_SONY_VAIO] = { .verbs = alc269_sony_vaio_fixup_verbs }, + [ALC269_FIXUP_DELL_M101Z] = { + .verbs = (const struct hda_verb[]) { + /* Enables internal speaker */ + {0x20, AC_VERB_SET_COEF_INDEX, 13}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, + {} + } + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), {} }; -- cgit v1.2.3-70-g09d2 From 8699a0b657b43fa6401537dfe345bee7aa8115ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 22:52:32 +0200 Subject: ALSA: pcm - Fix unbalanced pm_qos_request The pm_qos_request isn't freed properly when OSS PCM emulation is used because it skips snd_pcm_hw_free() call but directly releases the stream. This resulted in Oops later. Tested-by: Simon Kirby Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 134fc6c2e08..d4eb2ef8078 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1992,6 +1992,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) substream->ops->close(substream); substream->hw_opened = 0; } + if (pm_qos_request_active(&substream->latency_pm_qos_req)) + pm_qos_remove_request(&substream->latency_pm_qos_req); if (substream->pcm_release) { substream->pcm_release(substream); substream->pcm_release = NULL; -- cgit v1.2.3-70-g09d2 From 901d46d5a8eb821b03ca9e8cf005beb0c92f31ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 23:06:50 +0200 Subject: ALSA: pcm - Fix race with proc files The PCM proc files may open a race against substream close, which can end up with an Oops. Use the open_mutex to protect for it. Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 204af48c5cc..ac242a377ae 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -372,14 +372,17 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - return; + goto unlock; } snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access)); snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format)); @@ -398,20 +401,25 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames); } #endif + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - return; + goto unlock; } snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode)); snd_iprintf(buffer, "period_step: %u\n", runtime->period_step); @@ -421,24 +429,29 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold); snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size); snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary); + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; struct snd_pcm_status status; int err; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } memset(&status, 0, sizeof(status)); err = snd_pcm_status(substream, &status); if (err < 0) { snd_iprintf(buffer, "error %d\n", err); - return; + goto unlock; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); @@ -452,6 +465,8 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); + unlock: + mutex_unlock(&substream->pcm->open_mutex); } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -- cgit v1.2.3-70-g09d2 From cbfa5184cc5f58627f08c7fad225424f565b439d Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 17 Sep 2010 12:30:11 +0200 Subject: ASoC: fix clkdev API usage in sh/migor.c The clkdev API doesn't use .name and .id members of struct clk for clock lookup. Instead clocks should be added to a lookup list. Without this patch audio om the Migo-R board fails silently. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/migor.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index b823a5c9b9b..87e2b7fcbf1 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -12,6 +12,7 @@ #include #include +#include #include #include @@ -40,12 +41,12 @@ static struct clk_ops siumckb_clk_ops = { }; static struct clk siumckb_clk = { - .name = "siumckb_clk", - .id = -1, .ops = &siumckb_clk_ops, .rate = 0, /* initialised at run-time */ }; +static struct clk_lookup *siumckb_lookup; + static int migor_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -180,6 +181,13 @@ static int __init migor_init(void) if (ret < 0) return ret; + siumckb_lookup = clkdev_alloc(&siumckb_clk, "siumckb_clk", NULL); + if (!siumckb_lookup) { + ret = -ENOMEM; + goto eclkdevalloc; + } + clkdev_add(siumckb_lookup); + /* Port number used on this machine: port B */ migor_snd_device = platform_device_alloc("soc-audio", 1); if (!migor_snd_device) { @@ -200,12 +208,15 @@ static int __init migor_init(void) epdevadd: platform_device_put(migor_snd_device); epdevalloc: + clkdev_drop(siumckb_lookup); +eclkdevalloc: clk_unregister(&siumckb_clk); return ret; } static void __exit migor_exit(void) { + clkdev_drop(siumckb_lookup); clk_unregister(&siumckb_clk); platform_device_unregister(migor_snd_device); } -- cgit v1.2.3-70-g09d2 From 0f9f1ee9d1412d45a22bfd69dfd4d4324b506e9e Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Tue, 21 Sep 2010 17:05:46 +1000 Subject: ALSA: hda - Add Dell Latitude E6400 model quirk BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/643891 Set the Dell Latitude E6400 (1028:0233) SSID to use AD1984_DELL_DESKTOP Cc: stable@kernel.org Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b697fd2a6f8..10bbbaf6ebc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3641,6 +3641,7 @@ static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), + SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP), {} }; -- cgit v1.2.3-70-g09d2 From 0873a5ae747847ee55a63db409dff3476e45bcd9 Mon Sep 17 00:00:00 2001 From: "Erik J. Staab" Date: Wed, 22 Sep 2010 11:07:41 +0200 Subject: ALSA: oxygen: fix analog capture on Claro halo cards On the HT-Omega Claro halo card, the ADC data must be captured from the second I2S input. Using the default first input, which isn't connected to anything, would result in silence. Signed-off-by: Erik J. Staab Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 289cb4dacfc..6c0a11adb2a 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -543,6 +543,10 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF; break; } if (id->driver_data == MODEL_MERIDIAN || -- cgit v1.2.3-70-g09d2 From d47372e852391d0c6553dfbc7c4c56b89b527e13 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 21 Sep 2010 15:03:26 +0100 Subject: ASoC: Fix soc-cache buffer overflow bug Make sure we stay within the cache boundaries when updating the register cache. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index adbc68ce905..15d2779074e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -203,8 +203,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = (value >> 8) & 0xff; data[2] = value & 0xff; - if (!snd_soc_codec_volatile_register(codec, reg)) - reg_cache[reg] = value; + if (!snd_soc_codec_volatile_register(codec, reg) + && reg < codec->driver->reg_cache_size) + reg_cache[reg] = value; if (codec->cache_only) { codec->cache_sync = 1; -- cgit v1.2.3-70-g09d2 From 0077ca0b5c986477e33451b797b6e7dc92a8bbc0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Sep 2010 18:47:40 +0100 Subject: ASoC: Fix multi-componentism Spot the build testing. Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 15d2779074e..f6b0d2829ea 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -204,7 +204,7 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) - && reg < codec->driver->reg_cache_size) + && reg < codec->reg_cache_size) reg_cache[reg] = value; if (codec->cache_only) { -- cgit v1.2.3-70-g09d2 From 01fdf1801e349302fce5d9865470a7100a2d9b74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Sep 2010 09:09:42 +0200 Subject: ALSA: hda - Fix auto-parse of SPDIF input of Realtek codecs The SPDIF in audio widget must be searched through the list as the widget that contains the given pin as the connection source. The current code was implemented in a reverse way. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1312a6c8af..a432e6efd19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1594,12 +1594,22 @@ static void alc_auto_parse_digital(struct hda_codec *codec) } if (spec->autocfg.dig_in_pin) { - hda_nid_t dig_nid; - err = snd_hda_get_connections(codec, - spec->autocfg.dig_in_pin, - &dig_nid, 1); - if (err > 0) - spec->dig_in_nid = dig_nid; + dig_nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, dig_nid++) { + unsigned int wcaps = get_wcaps(codec, dig_nid); + if (get_wcaps_type(wcaps) != AC_WID_AUD_IN) + continue; + if (!(wcaps & AC_WCAP_DIGITAL)) + continue; + if (!(wcaps & AC_WCAP_CONN_LIST)) + continue; + err = get_connection_index(codec, dig_nid, + spec->autocfg.dig_in_pin); + if (err >= 0) { + spec->dig_in_nid = dig_nid; + break; + } + } } } -- cgit v1.2.3-70-g09d2 From e68d3b316ab7b02a074edc4f770e6a746390cb7d Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Sat, 25 Sep 2010 11:07:27 -0400 Subject: ALSA: sound/pci/rme9652: prevent reading uninitialized stack memory The SNDRV_HDSP_IOCTL_GET_CONFIG_INFO and SNDRV_HDSP_IOCTL_GET_CONFIG_INFO ioctls in hdspm.c and hdsp.c allow unprivileged users to read uninitialized kernel stack memory, because several fields of the hdsp{m}_config_info structs declared on the stack are not altered or zeroed before being copied back to the user. This patch takes care of it. Signed-off-by: Dan Rosenberg Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 1 + sound/pci/rme9652/hdspm.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b92adef8e81..d6fa7bfd9aa 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4609,6 +4609,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (err < 0) return err; + memset(&info, 0, sizeof(info)); spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 547b713d720..0c98ef9156d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4127,6 +4127,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO: + memset(&info, 0, sizeof(info)); spin_lock_irq(&hdspm->lock); info.pref_sync_ref = hdspm_pref_sync_ref(hdspm); info.wordclock_sync_check = hdspm_wc_sync_check(hdspm); -- cgit v1.2.3-70-g09d2 From 5591bf07225523600450edd9e6ad258bb877b779 Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Tue, 28 Sep 2010 14:18:20 -0400 Subject: ALSA: prevent heap corruption in snd_ctl_new() The snd_ctl_new() function in sound/core/control.c allocates space for a snd_kcontrol struct by performing arithmetic operations on a user-provided size without checking for integer overflow. If a user provides a large enough size, an overflow will occur, the allocated chunk will be too small, and a second user-influenced value will be written repeatedly past the bounds of this chunk. This code is reachable by unprivileged users who have permission to open a /dev/snd/controlC* device (on many distros, this is group "audio") via the SNDRV_CTL_IOCTL_ELEM_ADD and SNDRV_CTL_IOCTL_ELEM_REPLACE ioctls. Signed-off-by: Dan Rosenberg Cc: Signed-off-by: Takashi Iwai --- sound/core/control.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 070aab49019..45a818002d9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -31,6 +31,7 @@ /* max number of user-defined controls */ #define MAX_USER_CONTROLS 32 +#define MAX_CONTROL_COUNT 1028 struct snd_kctl_ioctl { struct list_head list; /* list of all ioctls */ @@ -195,6 +196,10 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, if (snd_BUG_ON(!control || !control->count)) return NULL; + + if (control->count > MAX_CONTROL_COUNT) + return NULL; + kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL); if (kctl == NULL) { snd_printk(KERN_ERR "Cannot allocate control instance\n"); -- cgit v1.2.3-70-g09d2 From e913b146493993c8ac33561655c590e58b500c6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Sep 2010 22:59:12 +0200 Subject: ALSA: i2c/other/ak4xx-adda: Fix a compile warning with CONFIG_PROCFS=n Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4xxx-adda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 1adb8a3c2b6..42d7844ecd0 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -900,7 +900,7 @@ static int proc_init(struct snd_akm4xxx *ak) return 0; } #else /* !CONFIG_PROC_FS */ -static int proc_init(struct snd_akm4xxx *ak) {} +static int proc_init(struct snd_akm4xxx *ak) { return 0; } #endif int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) -- cgit v1.2.3-70-g09d2 From d4cfa4d12f46e2520f4c1d1a92e891ce068b7464 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 10 Oct 2010 19:33:52 +0200 Subject: OSS: soundcard: locking bug in sound_ioctl() We shouldn't return directly here because we're still holding the &soundcard_mutex. This bug goes all the way back to the start of git. It's strange that no one has complained about it as a runtime bug. CC: stable@kernel.org Signed-off-by: Dan Carpenter Acked-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/oss/soundcard.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 92aa762ffb7..07f803e6d20 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -391,11 +391,11 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - return audio_ioctl(dev, file, cmd, p); + ret = audio_ioctl(dev, file, cmd, p); break; case SND_DEV_MIDIN: - return MIDIbuf_ioctl(dev, file, cmd, p); + ret = MIDIbuf_ioctl(dev, file, cmd, p); break; } -- cgit v1.2.3-70-g09d2 From 9b2167d59f38691b86430ce559c7fa9d4f973b1f Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Wed, 6 Oct 2010 15:45:46 +1100 Subject: ALSA: hda - Add another HP DV6 quirk BugLink: https://bugs.launchpad.net/bugs/653420 Add another HP DV6 notebook (103c:363e) to use STAC_HP_DV5. Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 95148e58026..c16c5ba0fda 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1747,6 +1747,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv6", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x363e, + "HP DV6", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- cgit v1.2.3-70-g09d2