July 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3] ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
by Dylan Reid 20 Jul '12

20 Jul '12

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is enabled in open, but is disabled in hdmi_init_pin which is called during system resume. This causes a system suspend/resume during playback to mute HDMI/DP. Enabling the pin in prepare instead of open allows calling snd_pcm_prepare after a system resume to restore audio. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0b4a1ea..641408d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -876,7 +876,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - int pinctl; /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); @@ -912,11 +911,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl | PIN_OUT); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1153,11 +1147,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } -- 1.7.11.2.138.g2b53359

2 1

Re: [alsa-devel] [regression] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 20 Jul '12

20 Jul '12

On Thu, Jul 19, 2012 at 12:29 AM, Oleksij Rempel (fishor) <bug-track(a)fisher-privat.net> wrote: > On 19.07.2012 08:44, Dylan Reid wrote: >> >> On Wed, Jul 18, 2012 at 10:26 PM, Oleksij Rempel (fishor) >> <bug-track(a)fisher-privat.net> wrote: >>> >>> On 17.07.2012 07:26, Dylan Reid wrote: >>>> >>>> >>>> On Sat, Jul 14, 2012 at 6:36 AM, Oleksij Rempel (fishor) >>>> <bug-track(a)fisher-privat.net> wrote: >>>>> >>>>> >>>>> dmesg is in attachment >>>> >>>> >>>> Thanks Oleksij, >>>> >>>> I didn't get much from that though. What is your powersave timeout >>>> set to? >>> >>> >>> >>> I do not know, it should be some default value. >>> >>> >>>> If you have time could you try adding the following and send >>>> me the output, I have two guesses about where it is spending it's time >>>> (no guesses about why yet though). I've got a couple of systems with >>>> a 269, I'll try to get some tests run on those tomorrow. >>> >>> >>> >>> result is attached. >>> >> Great, thanks Oleksij, >> >> I tried this on an ACER AC700 which also has the ALC269VB and could >> not get anything like this to happen. So I'm sorry to keep bothering >> you for more information. I attached the alsa-info.sh output, if you >> see any interesting differences let me know. >> >> From the output it looks like something is opening the driver about >> three times a second, do you know what is doing that? It appears to >> be doing what it should be in this case, although it is a little hard >> to tell. If you could strace the process doing all the opens that >> might provide some good info. >> >> Thanks again for the help. >> Dylan > > > > Ok, i did some more testing compared the difference between battery and > non-battery mode. > > first of all on non-battery mode, this part of code is not executed. I have > no messages about "power_up cancel sync". > Then i found some difference in syslog. > > Here is what pulse audio say if i reproduce this issue: > [pulseaudio] module-alsa-card.c: Failed to open hctl/mixer for jack > detection Interesting, I'll try to get pulseaudio installed on my system to test tomorrow. Does it give any more info if you turn up the verbosity? (set log-level to debug in /etc/pulse/daemon.conf)

1 1

Re: [alsa-devel] Selecting Audio formats
by Clemens Ladisch 19 Jul '12

19 Jul '12

Harman Kalra wrote: > Can you please explain difference between signed and unsigned PCM > formats, Signed PCM formats have a sign, unsigned don't. S16 allows values in the range -32768..32767, U16, 0..65535. > in what cases does unsigned is used, In practice, never. It was historically used by some 8-bit cards. > which is more efficient (memory and sound clarity) I refer the honourable gentleman to the reply I gave in the last mail. >> If you tried to set MPEG format, ALSA would inform you that the device >> does not support this. > > But when I queried the hardware regarding the supported audio > formats using the api " snd_pcm_format_name", it responded with a list of > formats, including , A-law, u-law-, IMA_ADPCM, MPEG, GSM, so what does it > mean. It means that the sample format identified by the constant SND_PCM_FORMAT_MPEG is indeed named "MPEG". This has nothing to do with what formats some particular hardware supports. > What would be audio sample file size in case of A-law, will it be > same as U8. I refer the honourable gentleman to the reply I gave in the last mail. Regards, Clemens

1 0

[alsa-devel] Selecting Audio formats
by Harman Kalra 19 Jul '12

19 Jul '12

Hello Everybody I am a newbie to Linux Sound Programming, so few of my queries may sound foolish, but kindly address them. *I want to capture a video on my beagle board Xm rev C (TI DM3730). *So as per as the audio capturing is concerned, I am very much confused about the audio formats, like which format should I choose to get a better sound quality in the least possible memory requirement. Since so many PCM formats such as S8, U8, S16_LE, U16_LE, A-law, u-law-, IMA_ADPCM, MPEG, GSM, etc are there so which format should I opt for. Also 1. What is difference between signed and unsigned PCM format, how does it can affect the sound quality. 2. If I go for MPEG format, does it mean that audio compression would handled at my audio port level and my DSP processor would not come into picture. 3. In reference to my above query i.e. 2, if audio compression is handled at audio port level, so at the time of audio/video sync up, how would be my processor be able to time sync both audio and video, since audio is not being compressed by the DSP. 4. If I opt for A-law or u-law, what could be benefits/disadvantages of using these formats. Would the audio sample be in compressed form in this case, if yes, how would audio/video sync up will happen in this case. If possible kindly provide me a link where I can find differences between these various formats, pros and cons of each of them. Thanks & Regards Harman

2 1

[alsa-devel] SGTL5000: fix max Mic Volume in SGTL5000_CHIP_MIC_CTRL
by Simon Brenner 19 Jul '12

19 Jul '12

Hi all, I have noticed a wrong value for the "max" field of the macro SOC_SINGLE_TLV() in the Linux kernel file <kernel_root>/sound/soc/codecs/sgtl5000.c (approx. line 402, sgtl5000_snd_controls[]): SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, 0, 4, 0, mic_gain_tlv) The value "4" should be a "3" (see datasheet "Low Power Stereo Codec with Headphone Amp" by Freescale, page 40). Regards, -Simon.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
by Dylan Reid 19 Jul '12

19 Jul '12

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is enabled in open, but is disabled in hdmi_init_pin which is called during system resume. This causes a system suspend/resume during playback to mute HDMI/DP. Enabling the pin in prepare instead of open allows calling snd_pcm_prepare after a system resume to restore audio. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0b4a1ea..764b975 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -914,9 +914,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl | PIN_OUT); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1153,11 +1150,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } -- 1.7.11.2.138.g2b53359

2 2

[alsa-devel] [PATCH] ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
by Dylan Reid 19 Jul '12

19 Jul '12

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is enabled in open, but is disabled in hdmi_init_pin which is called during system resume. This causes a system suspend/resume during playback to mute HDMI/DP. Enabling the pin in prepare allows calling snd_pcm_prepare after a system resume to restore audio. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0b4a1ea..c53f99c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1153,11 +1153,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } -- 1.7.11.2.138.g2b53359

3 3

Re: [alsa-devel] [regression] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 19 Jul '12

19 Jul '12

On Wed, Jul 18, 2012 at 10:26 PM, Oleksij Rempel (fishor) <bug-track(a)fisher-privat.net> wrote: > On 17.07.2012 07:26, Dylan Reid wrote: >> >> On Sat, Jul 14, 2012 at 6:36 AM, Oleksij Rempel (fishor) >> <bug-track(a)fisher-privat.net> wrote: >>> >>> dmesg is in attachment >> >> Thanks Oleksij, >> >> I didn't get much from that though. What is your powersave timeout >> set to? > > > I do not know, it should be some default value. > > >> If you have time could you try adding the following and send >> me the output, I have two guesses about where it is spending it's time >> (no guesses about why yet though). I've got a couple of systems with >> a 269, I'll try to get some tests run on those tomorrow. > > > result is attached. > Great, thanks Oleksij, I tried this on an ACER AC700 which also has the ALC269VB and could not get anything like this to happen. So I'm sorry to keep bothering you for more information. I attached the alsa-info.sh output, if you see any interesting differences let me know. >From the output it looks like something is opening the driver about three times a second, do you know what is doing that? It appears to be doing what it should be in this case, although it is a little hard to tell. If you could strace the process doing all the opens that might provide some good info. Thanks again for the help. Dylan

1 0

[alsa-devel] ASoC updates for 3.6
by Mark Brown 19 Jul '12

19 Jul '12

The following changes since commit 6b16351acbd415e66ba16bf7d473ece1574cf0bc: Linux 3.5-rc4 (2012-06-24 12:53:04 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 409b78cc17a4a3d07a541037575da648ced99437: ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define (2012-07-18 22:09:40 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.6 This has been a pretty quiet release - very little activity in framework terms, mostly just a few new drivers and updates: - Added the ability to add and remove DAPM paths dynamically, mostly for reparenting on clock changes. - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500 reference platform and ttc-dkp. - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP, Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI Isabelle and Wolfson Microelectronics WM5102 and WM5110 ---------------------------------------------------------------- Adam Thomson (2): ASoC: core: Add single controls with specified range of values ASoC: codecs: Add DA732x codec driver Axel Lin (1): ASoC: ml26124: Convert to devm_regmap_init_i2c Benoît Thébaudeau (2): ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect ASoC: dapm: Remove incomplete stereo code Brian Austin (2): ASoC: cs42l73: Convert to devm_regmap_init_i2c() ASoC: cs42l52: Convert to devm_regmap_init_i2c() Dan Carpenter (3): ASoC: isabelle: using an uninitialized variable ASoC: Ux500: unlock on an error path ASoC: STA529: fix an error message Fabio Estevam (1): ASoC: imx-mc13783: Add audmux settings for mx27pdk Hui Wang (1): ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000 Jarkko Nikula (1): ASoC: tlv320aic3x: Change Class-D amplifier gain control name Jiri Prchal (5): ASoC: tlv320aic3x: add missing registers and bits ASoC: tlv320aic3x: extending registers cache ASoC: tlv320aic3x: add deemphasis switch ASoC: tlv320aic3x: add AGC settings ASoC: tlv320aic3x: add input clock selection Kuninori Morimoto (2): ASoC: fsi: use dmaengine helper functions ASoC: fsi: use PIO handler if DMA handler was invalid Lars-Peter Clausen (3): dmaengine: Add wrapper for device_tx_status callback ASoC: dmaengine-pcm: Rename and deprecate snd_dmaengine_pcm_pointer ASoC: dmaengine-pcm: Add support for querying stream position from DMA driver Laxman Dewangan (1): ASoC: tegra: use dmaengine based dma driver Liam Girdwood (2): ASoC: pcm: Clean up logging in soc_new_pcm() ASoC: dpcm: Allow FE to be opened without valid BE routes. M R Swami Reddy (2): ASoC: Support TI Isabelle Audio driver ASoC: MAINTAINERS: Add maintainer for TI Isabelle Audio driver MR Swami Reddy (1): ASoC: isabelle: Remove regmap_exit() Marek Belisko (1): ASoC: dapm: Fix compilation warning Mark Brown (61): ASoC: codecs: Refresh copyrights for Wolfson drivers ASoC: ac97: Remove empty remove() function ASoC: wm2000: Add register readability information ASoC: io: Use dev_get_regmap() if driver doesn't provide a regmap ASoC: wm8350: Convert to direct regmap API usage ASoC: dapm: Use devm_clk_get() ASoC: dapm: Bodge for lack of a widely available clk API ASoC: dapm: The clock API is even less consistent than thought Revert "ASoC: fsl_ssi: convert to use devm_clk_get" ASoC: lm59453: Unconstify dai_driver ASoC: max98095: Staticise non-exported functions and export jack detect ASoC: wm8903: Convert to devm_regmap_init_i2c() ASoC: wm8904: Convert to module_i2c_driver() ASoC: wm8904: Convert to devm_regmap_init_i2c() ASoC: wm8996: Remove write sequencer registers from the defaults table ASoC: wm8996: Convert to devm_regmap_init_i2c() ASoC: wm8903: Move pin configuration into I2C probe() function ASoC: wm8903: Make interrupt handler use regmap directly ASoC: wm8903: Move interrupt request to I2C probe ASoC: wm8903: Move register default changes to I2C probe Merge branch 'asoc-ab8500' into for-3.6 ASoC: wm5100: Remove stubs of ASoC-level register map code ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x ASoC: isabelle: Staticise non-exported isabelle_dai ASoC: core: Add DOUBLE_R variants of the _RANGE controls ASoC: dapm: Try to add all routes even if one fails ASoC: io: Don't dereference regmap if we failed to get one ASoC: Add shared code for Wolfson Arizona class devices ASoC: wm5102: Initial driver Merge tag 'v3.5-rc4' into for-3.6 ASoC: wm8904: Rely entirely on the core for bias level management ASoC: wm8904: Move regulator acquisition and device identification to I2C ASoC: wm8904: Move register default setup into I2C probe() ASoC: wm8996: Move regulator notifier callbacks into I2C level ASoC: wm8996: Inline wm8996_reset() and optimise cache-only usage ASoC: wm8996: Move register default configuration to I2C probe ASoC: dwc: Bodge around continuing absence of clock API stubs ASoC: wm5102: Remove unused platform data header ASoC: dwc: Add missing __iomem annotations ASoC: spdif: Build separate RX and TX objects ASoC: wm8962: Log AIF configuration requested by hw_params() ASoC: core: Try to use regmap if the driver doesn't set up any I/O ASoC: wm8994: Don't suspend accessory detection ASoC: dwc: Staticise non-exported i2s_start() ASoC: arizona: Rename current rates tables to bclk_rates ASoC: arizona: Implement AIF clock configuration ASoC: arizona: Be more forgiving in BCLK selection ASoC: arizona: Enable ASYNCCLK domain for audio interfaces ASoC: wm5102: Allow routing through the ASRCs ASoC: wm1250-ev1: Flag all supported rates in the DAI ASoC: dapm: Mark widgets as dirty when a route is added ASoC: dapm: Allow routes to be deleted at runtime ASoC: arizona: Change DAPM routes for AIF clocks when we change them ASoC: arizona: Export dai_ops ASoC: arizona: Add IN4 to the mixer tables ASoC: wm5102: Fix cut'n'paste for digital volume registers ASoC: arizona: Support variable FLL VCO multipliers ASoC: wm5110: Add audio CODEC driver ASoC: wm8962: Redo early init of the part on resume ASoC: wm8994: Update micdet for irqdomain conversion ASoC: littlemill: Add userspace control of the WM1250 I/O Matthias Kaehlcke (1): ASoC: Free memory in the error paths of soc_of_parse_audio_routing() Ola Lilja (7): ALSA: pcm: Add debug-print helper function ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY ASoC: Ux500: Add platform-driver ASoC: codecs: Add AB8500 codec-driver mfd/ab8500: Move platform-data for ab8500-codec into mfd-driver ASoC: Ux500: Correct license strings ASoC: Ux500: Add machine-driver Qiao Zhou (1): ASoC: add ttc-dkb machine support Rajeev Kumar (3): ASoC: Add support for synopsys i2s controller as per ASoC framework. ASoC: Add support for SPEAr ASoC pcm layer. ASoC: STA529: Add support for STA529 Audio Codec Richard Zhao (1): ASoC: fsl_ssi: convert to use devm_clk_get Sachin Kamat (3): ASoC: cs42l52: Remove version.h header file inclusion ASoC: lm49453: Remove version.h header file inclusion ASoC: smdk_wm8994: Convert to use snd_soc_register_card() Scott Jiang (3): ASoC: add sport driver for bf6xx soc ASoC: add i2s dai driver for bf6xx soc ASoC: bfin: use dev_err to print error log instead of dev_dbg Simon Wilson (1): ASoC: twl6040: fix spelling mistake Stephen Warren (18): ASoC: wm8903: init GPIOs during I2C probe not codec probe ASoC: tegra+wm8903: move all GPIO setup into probe ASoC: tegra+wm8903: Use devm_gpio_request_one ASoC: tegra+wm8903: unconditionally free jack GPIOs in remove ASoC: tegra+alc5632: move all GPIO setup into probe ASoC: tegra+alc5632: unconditionally free jack GPIOs in remove ASoC: tegra+wm8903: simplify gpio tests in widget callbacks ASoC: tegra+wm8903: remove non-DT support for Seaboard ASoC: make snd_soc_dai_link more symmetrical ASoC: tegra: add .stream_name to CPU DAIs ASoC: tegra: statically define DAI link format ASoC: tegra: remove usage of rtd->codec ASoC: tegra: use DAI's not card's dev for dev_err ASoC: tegra: use regmap more directly ASoC: when initializing CPU DAI, don't duplicate any CODEC init ASoC: when removing a CPU DAI, clean up its DAPM context ASoC: factor out soc_remove_platform() ASoC: probe CODECs and platforms before DAIs and links Sylwester Nawrocki (2): ASoC: Convert S3C24XX I2S driver to gpiolib API ASoC: Convert S3C2412 I2S driver to gpiolib API Torben Hohn (1): ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define Uwe Kleine-König (1): ASoC: dmaengine_pcm: fix typo in comment Vipin Kumar (3): ASoC: SPEAr spdif_in: Add spdif IN support ASoC: spdif_receiver: Add support for spdif in Audio Codec ASoC: SPEAr spdif_out: Add spdif out support Zhangfei Gao (3): ASoC: mmp: add audio dma support ASoC: mmp: add sspa support ASoC: add mmp brownstone support MAINTAINERS | 2 + arch/arm/mach-ux500/board-mop500.c | 14 + include/linux/dmaengine.h | 6 + include/linux/mfd/abx500/ab8500-codec.h | 52 + include/linux/mfd/abx500/ab8500.h | 2 + include/linux/platform_data/mmp_audio.h | 22 + include/sound/designware_i2s.h | 69 + include/sound/dmaengine_pcm.h | 1 + include/sound/pcm.h | 11 + include/sound/soc-dapm.h | 12 + include/sound/soc.h | 77 +- include/sound/spear_dma.h | 35 + include/sound/spear_spdif.h | 29 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/blackfin/Kconfig | 21 +- sound/soc/blackfin/Makefile | 4 + sound/soc/blackfin/bf6xx-i2s.c | 234 +++ sound/soc/blackfin/bf6xx-sport.c | 422 ++++++ sound/soc/blackfin/bf6xx-sport.h | 82 + sound/soc/codecs/Kconfig | 31 + sound/soc/codecs/Makefile | 19 +- sound/soc/codecs/ab8500-codec.c | 2522 +++++++++++++++++++++++++++++++ sound/soc/codecs/ab8500-codec.h | 590 ++++++++ sound/soc/codecs/ac97.c | 6 - sound/soc/codecs/arizona.c | 937 ++++++++++++ sound/soc/codecs/arizona.h | 159 ++ sound/soc/codecs/cs42l52.c | 19 +- sound/soc/codecs/cs42l73.c | 20 +- sound/soc/codecs/da732x.c | 1627 ++++++++++++++++++++ sound/soc/codecs/da732x.h | 133 ++ sound/soc/codecs/da732x_reg.h | 654 ++++++++ sound/soc/codecs/isabelle.c | 1176 ++++++++++++++ sound/soc/codecs/isabelle.h | 143 ++ sound/soc/codecs/lm49453.c | 3 +- sound/soc/codecs/max98095.c | 5 +- sound/soc/codecs/ml26124.c | 5 +- sound/soc/codecs/spdif_receiver.c | 67 + sound/soc/codecs/sta529.c | 442 ++++++ sound/soc/codecs/tlv320aic3x.c | 40 +- sound/soc/codecs/tlv320aic3x.h | 27 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/wm1250-ev1.c | 7 +- sound/soc/codecs/wm2000.c | 32 +- sound/soc/codecs/wm5100-tables.c | 2 +- sound/soc/codecs/wm5100.c | 11 +- sound/soc/codecs/wm5102.c | 903 +++++++++++ sound/soc/codecs/wm5102.h | 21 + sound/soc/codecs/wm5110.c | 950 ++++++++++++ sound/soc/codecs/wm5110.h | 21 + sound/soc/codecs/wm8350.c | 22 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8731.c | 1 + sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 316 ++-- sound/soc/codecs/wm8904.c | 272 ++-- sound/soc/codecs/wm8960.c | 2 + sound/soc/codecs/wm8961.c | 2 + sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 46 +- sound/soc/codecs/wm8996.c | 587 +++---- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 3 + sound/soc/dwc/designware_i2s.c | 455 ++++++ sound/soc/ep93xx/ep93xx-pcm.c | 2 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-audmux.h | 1 + sound/soc/fsl/imx-mc13783.c | 49 +- sound/soc/fsl/imx-pcm-dma.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 3 +- sound/soc/mxs/mxs-pcm.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/Kconfig | 42 + sound/soc/pxa/Makefile | 8 + sound/soc/pxa/brownstone.c | 174 +++ sound/soc/pxa/mmp-pcm.c | 297 ++++ sound/soc/pxa/mmp-sspa.c | 480 ++++++ sound/soc/pxa/mmp-sspa.h | 92 ++ sound/soc/pxa/ttc-dkb.c | 173 +++ sound/soc/samsung/littlemill.c | 7 + sound/soc/samsung/s3c2412-i2s.c | 10 +- sound/soc/samsung/s3c24xx-i2s.c | 10 +- sound/soc/samsung/smdk_wm8994.c | 36 +- sound/soc/sh/fsi.c | 48 +- sound/soc/soc-core.c | 328 +++- sound/soc/soc-dapm.c | 164 +- sound/soc/soc-dmaengine-pcm.c | 33 +- sound/soc/soc-io.c | 15 +- sound/soc/soc-pcm.c | 12 +- sound/soc/spear/spdif_in.c | 297 ++++ sound/soc/spear/spdif_in_regs.h | 60 + sound/soc/spear/spdif_out.c | 389 +++++ sound/soc/spear/spdif_out_regs.h | 79 + sound/soc/spear/spear_pcm.c | 214 +++ sound/soc/tegra/Kconfig | 3 +- sound/soc/tegra/tegra20_i2s.c | 94 +- sound/soc/tegra/tegra20_i2s.h | 1 - sound/soc/tegra/tegra20_spdif.c | 36 +- sound/soc/tegra/tegra20_spdif.h | 1 - sound/soc/tegra/tegra30_i2s.c | 85 +- sound/soc/tegra/tegra30_i2s.h | 1 - sound/soc/tegra/tegra_alc5632.c | 32 +- sound/soc/tegra/tegra_pcm.c | 115 ++ sound/soc/tegra/tegra_pcm.h | 2 + sound/soc/tegra/tegra_wm8753.c | 8 +- sound/soc/tegra/tegra_wm8903.c | 259 ++-- sound/soc/tegra/trimslice.c | 30 +- sound/soc/ux500/Kconfig | 18 + sound/soc/ux500/Makefile | 6 + sound/soc/ux500/mop500.c | 113 ++ sound/soc/ux500/mop500_ab8500.c | 431 ++++++ sound/soc/ux500/mop500_ab8500.h | 22 + sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_pcm.c | 318 ++++ sound/soc/ux500/ux500_pcm.h | 35 + 126 files changed, 16719 insertions(+), 1344 deletions(-) create mode 100644 include/linux/mfd/abx500/ab8500-codec.h create mode 100644 include/linux/platform_data/mmp_audio.h create mode 100644 include/sound/designware_i2s.h create mode 100644 include/sound/spear_dma.h create mode 100644 include/sound/spear_spdif.h create mode 100644 sound/soc/blackfin/bf6xx-i2s.c create mode 100644 sound/soc/blackfin/bf6xx-sport.c create mode 100644 sound/soc/blackfin/bf6xx-sport.h create mode 100644 sound/soc/codecs/ab8500-codec.c create mode 100644 sound/soc/codecs/ab8500-codec.h create mode 100644 sound/soc/codecs/arizona.c create mode 100644 sound/soc/codecs/arizona.h create mode 100644 sound/soc/codecs/da732x.c create mode 100644 sound/soc/codecs/da732x.h create mode 100644 sound/soc/codecs/da732x_reg.h create mode 100644 sound/soc/codecs/isabelle.c create mode 100644 sound/soc/codecs/isabelle.h create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/codecs/wm5102.c create mode 100644 sound/soc/codecs/wm5102.h create mode 100644 sound/soc/codecs/wm5110.c create mode 100644 sound/soc/codecs/wm5110.h create mode 100644 sound/soc/dwc/Kconfig create mode 100644 sound/soc/dwc/Makefile create mode 100644 sound/soc/dwc/designware_i2s.c create mode 100644 sound/soc/pxa/brownstone.c create mode 100644 sound/soc/pxa/mmp-pcm.c create mode 100644 sound/soc/pxa/mmp-sspa.c create mode 100644 sound/soc/pxa/mmp-sspa.h create mode 100644 sound/soc/pxa/ttc-dkb.c create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_pcm.c create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h

2 1

[alsa-devel] [PATCH] ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
by Torben Hohn 18 Jul '12

18 Jul '12

The MX31 Audmux also has 7 Ports. This patch adds the missing define, and makes the debugfs code iterate over that port too. Signed-off-by: Torben Hohn <torbenh(a)linutronix.de> --- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-audmux.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 0803274..e7c800e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab..b8ff44b 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 -- 1.7.10

3 2