December 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Tune the accessory detection rates for WM8996
by Mark Brown 14 Dec '11

14 Dec '11

Use longer intervals when the microphone is not inserted to increase robustness against leisurely insertion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 18 ++++++++++++------ 1 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8f88f5a..da7acae 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2572,8 +2572,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) SND_JACK_BTN_0); snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - WM8996_MICD_RATE_MASK); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + WM8996_MICD_RATE_MASK | + 9 << WM8996_MICD_BIAS_STARTTIME_SHIFT); return; } @@ -2590,8 +2592,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) /* Increase poll rate to give better responsiveness * for buttons */ snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 5 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); } else { dev_dbg(codec->dev, "Mic button up\n"); snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); @@ -2639,8 +2643,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) * responsiveness. */ snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 7 << WM8996_MICD_RATE_SHIFT); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 7 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); } } } -- 1.7.7.3

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Re: [alsa-devel] MAYA44 USB audio device no inputs control
by Kamil Sobolak 14 Dec '11

14 Dec '11

Hi! I attached two logs in zip folder. Please check this. Thanks! Kamil Sobolak

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[alsa-devel] [PATCH 1/8] ASoC: Remove cache default for volatile wm9081 reset register
by Mark Brown 14 Dec '11

14 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 1f2672b..a6bab39 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -31,7 +31,6 @@ #include "wm9081.h" static struct reg_default wm9081_reg[] = { - { 0, 0x9081 }, /* R0 - Software Reset */ { 2, 0x00B9 }, /* R2 - Analogue Lineout */ { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ { 4, 0x0001 }, /* R4 - VMID Control */ -- 1.7.7.3

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Re: [alsa-devel] [PATCH 5/5] ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
by Dong Aisheng-B29396 14 Dec '11

14 Dec '11

> -----Original Message----- > From: linux-kernel-owner(a)vger.kernel.org [mailto:linux-kernel- > owner(a)vger.kernel.org] On Behalf Of Shawn Guo > Sent: Wednesday, December 14, 2011 3:48 PM > To: Liam Girdwood; Mark Brown > Cc: Dong Aisheng; alsa-devel(a)alsa-project.org; linux- > kernel(a)vger.kernel.org; linux-arm-kernel(a)lists.infradead.org; Shawn Guo > Subject: [PATCH 5/5] ASoC: mxs: correct 'direction' of > device_prep_dma_cyclic > > The commit 49920bc (dmaengine: add new enum dma_transfer_direction) > changes the type of parameter 'direction' of device_prep_dma_cyclic from > dma_data_direction to dma_transfer_direction. > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > --- > sound/soc/mxs/mxs-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Regards Dong Aisheng

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[alsa-devel] [PATCH] ALSA: hda: remove unused quirk for inverted mute led
by Gustavo Maciel Dias Vieira 14 Dec '11

14 Dec '11

Commit b99a776d0b17ae0f3a54e86009887a00ac4889d0 removed all effects of the STAC92HD83* model quirk "hp". However, it left the model selection and documentation behind, confusing users with inverted mute leds. Completely remove this quirk and its documentation. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo(a)sagui.org> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/patch_sigmatel.c | 4 ---- 2 files changed, 0 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index bda5469..c8c5454 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -338,7 +338,6 @@ STAC92HD83* mic-ref Reference board with power management for ports dell-s14 Dell laptop dell-vostro-3500 Dell Vostro 3500 laptop - hp HP laptops with (inverted) mute-LED hp-dv7-4000 HP dv-7 4000 auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8d2f9d..2034f33 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -96,7 +96,6 @@ enum { STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, STAC_DELL_VOSTRO_3500, - STAC_92HD83XXX_HP, STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, STAC_92HD83XXX_MODELS @@ -1691,7 +1690,6 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", [STAC_DELL_VOSTRO_3500] = "dell-vostro-3500", - [STAC_92HD83XXX_HP] = "hp", [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", }; @@ -1706,8 +1704,6 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "unknown Dell", STAC_DELL_S14), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028, "Dell Vostro 3500", STAC_DELL_VOSTRO_3500), - SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, - "HP", STAC_92HD83XXX_HP), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1657, -- 1.7.7.3

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[alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 14 Dec '11

14 Dec '11

Hi, Back to 3.0, the sound on my soundcard [1] started to be choppy, I reported it [2] and it got fixed (thanks to Takashi Iwai!). However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): I've got similar choppy sound again. I bisected the few commits that happened on sound/pci/hda/patch_realtek.c, and finally found the villain: "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" Reverting it from v3.1 fixes the problem. master (94956ee) also suffers of the problem, but I couldn't test without that commit because reverting it fails, too much changes happened. Thanks for your attention, and if you need any extra information (although many are already available in my previous report [2]) or testing, please let me know. Regards, Colomban [1] Realtek ALC889 on an MSI H55M-E33 [2] bugs.debian.org/cgi-bin/bugreport.cgi?bug=635362, http://mailman.alsa-project.org/pipermail/alsa-devel/2011-July/042250.html (though I can't find my original message here) [3] I first saw the problem in Debian's 3.0.0-6 which includes 3.0 up to .8 -- but the debugging I did was on vanilla kernel from git://github.com/torvalds/linux.git.

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Re: [alsa-devel] [PATCH] ALSA HDA: Add support for new AMD products
by Takashi Iwai 14 Dec '11

14 Dec '11

At Wed, 14 Dec 2011 16:10:27 +0800, Andiry Xu wrote: > > This patch adds HDMI audio support for new AMD products. As HW default > disable snoop, force non-snoop mode in HD audio driver. > > Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> Thanks, applied now. Also I committed a clean up patch to merge both VIA and AMD-HDMI non-snoop check in a single place. Takashi > --- > sound/pci/hda/hda_intel.c | 13 +++++++++++++ > 1 files changed, 13 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index d1582dd..9f7c901a 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -487,6 +487,7 @@ enum { > AZX_DRIVER_SCH, > AZX_DRIVER_ATI, > AZX_DRIVER_ATIHDMI, > + AZX_DRIVER_ATIHDMI_NS, > AZX_DRIVER_VIA, > AZX_DRIVER_SIS, > AZX_DRIVER_ULI, > @@ -533,6 +534,7 @@ static char *driver_short_names[] __devinitdata = { > [AZX_DRIVER_SCH] = "HDA Intel MID", > [AZX_DRIVER_ATI] = "HDA ATI SB", > [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", > + [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", > [AZX_DRIVER_VIA] = "HDA VIA VT82xx", > [AZX_DRIVER_SIS] = "HDA SIS966", > [AZX_DRIVER_ULI] = "HDA ULI M5461", > @@ -2678,6 +2680,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, > > chip->single_cmd = single_cmd; > chip->snoop = hda_snoop; > + if (chip->driver_type == AZX_DRIVER_ATIHDMI_NS) > + chip->snoop = 0; > > if (bdl_pos_adj[dev] < 0) { > switch (chip->driver_type) { > @@ -2776,6 +2780,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, > chip->capture_streams = ULI_NUM_CAPTURE; > break; > case AZX_DRIVER_ATIHDMI: > + case AZX_DRIVER_ATIHDMI_NS: > chip->playback_streams = ATIHDMI_NUM_PLAYBACK; > chip->capture_streams = ATIHDMI_NUM_CAPTURE; > break; > @@ -3037,6 +3042,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > { PCI_DEVICE(0x1002, 0xaa48), > .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0x9902), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaaa0), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaaa8), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaab0), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > /* VIA VT8251/VT8237A */ > { PCI_DEVICE(0x1106, 0x3288), > .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, > -- > 1.7.4.1 > >

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[alsa-devel] start_threshold and stream controls for PCM streams
by Nallasellan, Singaravelan 14 Dec '11

14 Dec '11

Hi, Would like to know how start_threshold is expected to be used in single_threaded audio player? I assume that in blocking implementation, the write call will block till the ring buffer becomes empty. Is it right? If the audio player has to wait for the write call to be blocked, I guess it is possible that the delay to service the stream controls will be delayed as long as the buffer duration. If so, how can I send stream control operations for the stream when the stream is blocked for the buffer? What are the option to service stream controls immediately? Thanks in advance. Thanks -Sing

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[alsa-devel] [PATCH] ALSA: HDA: Use LPIB Position fix for Intel SCH Poulsbo
by David Henningsson 14 Dec '11

14 Dec '11

Hi Takashi, I just got another position_fix=1 report for the same vendor/deviceID [1] as I sent you the other day, and I assume Eric Piel (a month ago on this list) have the same chipset as well, although I cannot see his alsa-info. I think you have it, can you verify? So...can we consider quirking the entire chip to use LPIB, like in the attached patch (untested)? If so I'll send another patch later to remove existing quirks for this chipset. // David [1] http://www.alsa-project.org/db/?f=b7421521b8d8a4db3842ee14cdb54b4f440cae5e

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[alsa-devel] [PATCH] ASoC: Remove unused extern declarations for sh4_hac_dai and sh7760_soc_platform
by Axel Lin 14 Dec '11

14 Dec '11

Both sh4_hac_dai and sh7760_soc_platform are changed to static by multi-component patch and they are not used in sh7760-ac97.c now. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/sh/sh7760-ac97.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index c62ae68..df651e8 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -16,10 +16,6 @@ #define IPSEL 0xFE400034 -/* platform specific structs can be declared here */ -extern struct snd_soc_dai_driver sh4_hac_dai[2]; -extern struct snd_soc_platform_driver sh7760_soc_platform; - static struct snd_soc_dai_link sh7760_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", -- 1.7.5.4

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