September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa-lib 1.0.21a release
by Jaroslav Kysela 22 Sep '09

22 Sep '09

Hello, I made a quick release 1.0.21a of the alsa-lib package to fix regression for the configuration parsing (makes some device names unavailable). These patches were included: 19abc37 PCM - Change the hw_params determination order d1b70af Change dmix.conf to accept user configuration from defaults.dmix.<driver_id>.xxx eb9ae58 Revert "Fix driver conf parsing in snd_config_hook_load_for_all_cards()" f0a32a6 mixer: Add Speaker and Beep names to the weight list Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] (no subject)
by Delusion's Master 22 Sep '09

22 Sep '09

Hi, I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Windows Live™: Keep your life in sync. Check it out! http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009

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[alsa-devel] [PATCH] wm8753: fix mapping when MONOMIX is set to Stereo
by Phil Vandry 22 Sep '09

22 Sep '09

When MONOMIX is set to Stereo, Left PGA was not powered on but should be. Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue. Signed-off-by: Phil Vandry <vandry(a)TZoNE.ORG> --- sound/soc/codecs/wm8753.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414..5ad677c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Mono Capture mixer-mux */ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, + {"Capture Left Mixer", "Stereo", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},

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[alsa-devel] S/PDIF on NVidia CK804 not working
by Mitko Haralanov 21 Sep '09

21 Sep '09

Hi all, I am new to the list and was hoping that someone here could give me some help and advise as I am stuck and none of my research has turned anything up. I am having trouble getting any sound out of the S/PDIF port on my built-in sound card. The card is recognized and 'aplay -L' shows the iec958 device (as evident by the attached alsa-info.txt file) but when I try to play any sound through it (using aplay) I get one of two errors: 1. If I use the iec958 device directly: $ aplay -D iec958 /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:984: Channels count non available 2. If I try to use the plug:iec958: $ aplay -D plug:iec958 /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono (the sound file seems to play but there is no audio coming out of the receiver) Looking through the ALSA website, I see that my sound device is supported. Configuring /etc/asound.conf and /etc/modprobe.conf as per the ALSA website yields the same results as above. Can anyone please, help me get my S/PDIF working? Thank you very much! -- Mitko Haralanov

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[alsa-devel] Application and hw mixing for VoIP (modified)
by Fabien COMTE 21 Sep '09

21 Sep '09

Hi, I need some help please. I have an application that does VoIP and alsa interface. I want to : - send through VoIP application a mixed sound from mic and an another application (aplay should be ok for testing). mic ---------- soundcard 0 -------------> mix ---- alsa interface -----> voip application application 1 --- alsa interface ------> (aplay) - use sound received from VoIP with HP output and an another application (arecord should be ok for testing). ----- alsa interface ------> application 2 (arecord) voip application ----- alsa interface ----> share ------ soundcard 0 --------> HP output And a little question : How to define an application interface in asoundrc ? I tried a lot of asoundrc but they doesn't work. Could someone help me ? Thank you. Fabien

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[alsa-devel] [PATCH] ALSA: lx6464es - remove unused struct member
by Tim Blechmann 21 Sep '09

21 Sep '09

we cannot set the sampling rate of the device, but can only read it from the board, so we don't need the member for it. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.h | 1 - 1 files changed, 0 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] ALSA: lx6464es - cleanup of rmh message bus function
by Tim Blechmann 21 Sep '09

21 Sep '09

the rmh bus is not used asynchronously, so it is safe to remove the specific code pieces. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.h | 1 - sound/pci/lx6464es/lx_core.c | 98 +---------------------------------------- 2 files changed, 1 insertions(+), 98 deletions(-)

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[alsa-devel] [PATCH] sb16: request_region failure ignored in snd_sb16_isa_probe1()
by Roel Kluin 21 Sep '09

21 Sep '09

request_region() may fail. clean up if it does, also a release region later on was missing. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with sed: http://kernelnewbies.org/roelkluin build, sparse and checkpatch tested. diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 519c363..f50ed6f 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -502,6 +502,11 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) fm_port[dev] = port[dev]; /* block the 0x388 port to avoid PnP conflicts */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); + if (acard->fm_res == NULL) { + snd_card_free(card); + return -ENOMEM; + } + #ifdef SNDRV_SBAWE_EMU8000 /* non-PnP AWE port address is hardwired with base port address */ awe_port[dev] = port[dev] + 0x400; @@ -509,6 +514,7 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) snd_card_set_dev(card, pdev); if ((err = snd_sb16_probe(card, dev)) < 0) { + release_region(0x388, 4); snd_card_free(card); return err; }

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[alsa-devel] [PATCH] some minor changes for AD1836 and AD1938 codec drivers
by Barry Song 21 Sep '09

21 Sep '09

1. delete redundant assignment to bus field in spi_driver structure 2. fix lost assignment to set_bias_level entry in ad1938 codec dai 3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836" Signed-off-by: Barry Song <barry.song(a)analog.com> --- sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 01343dc..c48485f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-spi", - .bus = &spi_bus_type, + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 9a049a1..34b30ef 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { .name = "ad1938", - .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = ad1938_spi_probe, @@ -515,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; + codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.5.6.3

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[alsa-devel] Multiple, independent streams with HDA
by Stefan Bruens 20 Sep '09

20 Sep '09

Hi as HDA supports up to 16 independent streams (sample rate, resolution), is there a possibility to expose this to user space? Use cases: - Simultaneous playback of music to speaker and connecting HPs for VoIP - Multiseat - pair of I/O for each seat ... I think there is already limited support for this, as e.g. SPDIF and analog expose independent PCMS (a least on STAC9271), although it is just a single codec. Maybe even make this configurable, either at run time, or at module load time. E.g. "option hda-intel pcm_out_nids=0x2:0x3,0x4:0x1e pcm_in_nids=0x7:0x20" creates pcmC0D{0,1,2}p and pcmC0D{0,1}c, where 0p is analog stereo out, 1p is analog 4chan out, 2p digital out, 0c is analog stereo in, and finally 1c is digital stereo in. (This example is for STAC9271: cat /proc/asound/Intel/codec#2 | egrep "Node|Converter" | grep Conv -B1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 -- Node 0x1e [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 -- Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 ) Stefan -- Stefan Brüns / Bergstraße 21 / 52062 Aachen phone: +49 241 53809034 mobile: +49 151 50412019

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[alsa-devel] [[ALSA] PATCH - /pci/hda/patch_conexant.c 1/1] patch_conexant.c: cd pin added to mixer for benq r55 laptop,
by nowymarluk＠wp.pl 20 Sep '09

20 Sep '09

From: edoiks <nowymarluk(a)wp.pl> Signed-off-by: edoiks <nowymarluk(a)wp.pl> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 9d899ed..700a430 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -687,6 +687,8 @@ static struct hda_input_mux cxt5045_capture_source_benq = { { "IntMic", 0x1 }, { "ExtMic", 0x2 }, { "LineIn", 0x3 }, + { "CD Pin", 0x4 }, + { "Mixer ", 0x0 }, } }; @@ -816,6 +818,14 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), + {} }; -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 20 Sep '09

20 Sep '09

The following changes since commit cdc65fbe18aef15e92d2ebb410a189fbf956fb06: Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The only really big thing here is the addition of the AK4671 driver, other than that it's minor tweaks. Barry Song (4): ASoC: Provide API for reordering channels ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 ASoC: fix typos in Blackfin headers ASoC: Blackfin I2S: add lost platform_device parameter to resume function Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration Cliff Cai (2): ASoC: Blackfin AC97: add a few missing multichannel define handling ASoC: Blackfin I2S: fix resuming when device hasn't been used Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> Jassi (3): ASoC: S3C lrsync function made to work with IRQs disabled. ASoC: S3C I2S LRCLK polarity option. ASoC: Return correct codec clock in s3c64xx-i2s Joonyoung Shim (2): ASoC: Factor out I2C 8 bit address 8 bit data I/O ASoC: AK4671: add ak4671 codec driver Julia Lawall (1): ASoC: Clean up error handling in MPC5200 DMA setup Mark Brown (5): ASoC: Add source argument to PLL configuration ASoC: Allow per-route connectedness checks for supplies ASoC: Fix display of stream name in DAPM debugfs ASoC: Add S3C64xx IIS CDCLK source selection Merge branch 'for-2.6.32' into for-2.6.33 Miguel Aguilar (1): ASoC: Davinci: Add audio codec support for DM365 EVM Mike Frysinger (1): ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 2 + include/sound/soc-dai.h | 13 +- include/sound/soc-dapm.h | 7 + sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 8 + sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-ad1836.c | 7 + sound/soc/blackfin/bf5xx-ad1938.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 22 +- sound/soc/blackfin/bf5xx-i2s.h | 2 +- sound/soc/blackfin/bf5xx-sport.c | 2 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 45 +- sound/soc/blackfin/bf5xx-tdm.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/ak4671.c | 825 +++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 ++++ sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 4 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8900.c | 4 +- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8974.c | 5 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9713.c | 4 +- sound/soc/davinci/Kconfig | 4 +- sound/soc/davinci/davinci-evm.c | 7 +- sound/soc/davinci/davinci-mcasp.c | 24 +- sound/soc/fsl/mpc5200_dma.c | 33 +- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 46 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 19 +- sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/soc-cache.c | 30 + sound/soc/soc-core.c | 32 +- sound/soc/soc-dapm.c | 26 +- 48 files changed, 1312 insertions(+), 101 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 20 Sep '09

20 Sep '09

The following changes since commit d4e54e871f4d2ca29df081abf8e0d5209d252979: Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (2): ASoC: fix typos in Blackfin headers ASoC: Blackfin I2S: add lost platform_device parameter to resume function Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration Cliff Cai (2): ASoC: Blackfin AC97: add a few missing multichannel define handling ASoC: Blackfin I2S: fix resuming when device hasn't been used Mike Frysinger (1): ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 22 +++++++++------------- sound/soc/blackfin/bf5xx-i2s.h | 2 +- sound/soc/blackfin/bf5xx-sport.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 6 files changed, 42 insertions(+), 18 deletions(-)

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[alsa-devel] [PATCH] SMDK64XX I2S: Added machine driver for WM8580
by jassi brar 20 Sep '09

20 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 283 +++++++++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..e4b5e0f --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,283 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_enable_pin(codec, "Center/Sub"); + snd_soc_dapm_enable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

2 4

[alsa-devel] snd_device_name_hint problems
by David Henningsson 20 Sep '09

20 Sep '09

Hello, First; I'm not completely sure how much of this is your issue, and how much is my distribution's fault (Ubuntu). If you believe my problems is due to bugs in the distribution, please say so. I'm working with a small recording application and I want to enumerate possible recording sources/devices, just to put in a combobox. In my computer I have two soundcards, one on the motherboard and one PCI soundcard (ESI Juli@). I also have Pulseaudio installed, so the devices I want to show up are those three - and possibly alternatives under these, such as analog/digital inputs. According to Lennart's guide to sound API's, snd_device_name_hint is the way to go, as both snd_card_xxx and snd_config_xxx are deprecated. (Speak up if you don't agree.) So assuming I do about the same as "arecord -L" does, I get the following output (descriptions omitted) : front:CARD=Juli,DEV=0 surround40:CARD=Juli,DEV=0 surround41:CARD=Juli,DEV=0 surround50:CARD=Juli,DEV=0 surround51:CARD=Juli,DEV=0 surround71:CARD=Juli,DEV=0 iec958:CARD=Juli,DEV=0 null front:CARD=CK804,DEV=0 surround40:CARD=CK804,DEV=0 surround41:CARD=CK804,DEV=0 surround50:CARD=CK804,DEV=0 surround51:CARD=CK804,DEV=0 surround71:CARD=CK804,DEV=0 I have two problems with this: first thing is that the "pulse" device is not listed. I think that could be solved either by setting "defaults.namehint.showall" to on, or by setting the name hint of the pulse device to be included in the "basic" devices. Would you recommend doing either one of those, and if so, which one? Second problem is that I don't have surround input on either of my soundcards, so most of the device options above do not make any sense. Even if I try, I have a hard time recording from them (arecord fails to record it with various errors, such as "Sample format non available"). If I instead use "plughw:CARD=name", recording works. To sum it up, the output I would like to have, is at least these three: plughw:CARD=Juli plughw:CARD=CK804 pulse How do I/we get there? // David

2 3

Re: [alsa-devel] Application and hw mixing for VoIP
by Mark Brown 19 Sep '09

19 Sep '09

On Thu, Sep 17, 2009 at 11:51:33AM +0200, Fabien COMTE wrote: > I have an application that does VoIP and use OSS emulated interface. I suspect part of your problem here is the in-kernel OSS emulation - that bypasses all the ALSA user space mixing support. If you can't change the OSS using application to use ALSA then I believe there are some other options based on preloading a library to intercept the attempts to open the OSS device and emulate it but I'm not really familiar with them.

1 0

[alsa-devel] [PATCH] SMDK64XX I2S: Added machine driver for WM8580
by jassi brar 19 Sep '09

19 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 284 +++++++++++++++++++++++++++++++++++ 3 files changed, 294 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..186e1de --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,284 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* Currently, WM8580 driver doesn't support PLL-out rates + * other than those mentioned in Table-52 Page-58 of WM8580A + * manual. Some of these unavailable rates include + * {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a value that results in + * (params_rate(params) * rfs), and itself, acceptable to the both + * the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_enable_pin(codec, "Center/Sub"); + snd_soc_dapm_enable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

2 2

[alsa-devel] [PATCH] S3C64XX I2S: Return correct codec clock
by jassi brar 18 Sep '09

18 Sep '09

From: Jassi <jassi.brar(a)samsung.com> Instead of always returnig pointer to the 'audio-bus' clock, check which clock is used to generate internal clocks and then return it's pointer. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 62a7fce..66f4ded 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -133,8 +133,12 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) { struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - return i2s->iis_cclk; + if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; } EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); -- 1.6.2.5

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Fixes to McASP configuration
by Chaithrika U S 18 Sep '09

18 Sep '09

McASP register settings are not correct for DSP mode of operation. There is a channel swap initally. This patch provides fixes to the register values for proper working. Tested on DA830/OMAP-L137 EVM, DM6467 EVM. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 1 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index eca22d7..7a06c0a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -512,34 +512,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { u32 fmt = 0; + u32 mask, rotate; switch (channel_size) { case DAVINCI_AUDIO_WORD_8: fmt = 0x03; + rotate = 6; + mask = 0x000000ff; break; case DAVINCI_AUDIO_WORD_12: fmt = 0x05; + rotate = 5; + mask = 0x00000fff; break; case DAVINCI_AUDIO_WORD_16: fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; break; case DAVINCI_AUDIO_WORD_20: fmt = 0x09; + rotate = 3; + mask = 0x000fffff; break; case DAVINCI_AUDIO_WORD_24: fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; break; case DAVINCI_AUDIO_WORD_28: fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; break; case DAVINCI_AUDIO_WORD_32: fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; break; default: @@ -550,6 +565,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, RXSSZ(fmt), RXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + return 0; } @@ -638,7 +660,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ @@ -655,7 +676,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } -- 1.5.6

2 4

[alsa-devel] Questions regarding workaround for VMware
by Bankim Bhavsar 18 Sep '09

18 Sep '09

Takashi I came across a workaround for VMware in ALSA kernel code http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h… Thanks for the fix. In the comments you mention about inaccurate timer source. Could you elaborate on the problem? As of now we see that sound apps running in Fedora 11 guest OS with kernel 2.6.29.4-167 don't make any progress and can hang window manager like Metacity. VMware virtualizes Ensoniq ES1371 sound device. Is there something specific you would like us to fix in our virtual sound device or timer source? Thanks, Bankim.

4 6

Re: [alsa-devel] Fwd: Dell Studio 14, internal mic
by Alistair Boyle 18 Sep '09

18 Sep '09

Takashi, As mentioned below I have a Dell Studio 14, with a two channel mic built into the screen, speakers, and two headphone jacks and a mic jack on the side. I've attached an updated alsa-info.sh output for 1.0.21. Things are working significantly better now (vs. 1.0.20) but a little messed up in the mixer controls. (When the module loads it says its making a guess for the chip.) For the playback settings, "headphone" controls both headphones (should only set one of the headphone jacks), "headphone as lineout" sets the front-most headphone jack as lineout and disables the speaker off feature when something is plugged into the front-most jack (good, unless this is supposed to be available/work for both headphone jacks?), "headphone 1" does nothing (bad, should be other headphone control?), "front" sets the built-in speaker volume as expected, "pc beep" makes a lot of crackling on the speaker when a "ding" is hit on a terminal and so is muted (maybe not alsa's fault?) Plugging in the headphones (with "headphone as lineout" disabled) now mutes the "front" speakers as expected. For the capture settings, "digital" should be the first setting since its the master capture control, "front mic" controls the stereo built-in mic (good), "line" doesn't do anything(?), "mic" sets the mic jack (good), there are two capture volumes (capture and capture 1) and two "input source"s (line, mic, or front mic). I think there should only be mic or front mic unless there's someway to set the mic jack as "line in" somewhere else? I can now record from the built-in mic ("front mic") vs 1.0.20. So to summarize: - independent headphones don't have separate headphone volumes - "digital" should be the master capture control - there seems to be duplicate capture controls (capture & capture 1, wit corresponding "input sources") - "pc beep" seems to be a bit busted Alistair On Mon, Sep 14, 2009 at 6:29 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sat, 12 Sep 2009 22:48:28 -0400, > Alistair Boyle wrote: >> >> Hi, >> >> I sent this email to the alsa-user mailing list a while back but >> didn't get much of a response. It looks like you're the guy to talk to >> about this from looking at the alsa repository. Hopefully you can help >> me get this working. > > Please post to alsa-devel ML and add me to Cc. > > What I can advise now is to try the very latest alsa-driver, e.g. > 1.0.21 release. > > > HTH, > > Takashi > >> >> Alistair Boyle >> >> >> ---------- Forwarded message ---------- >> From: Alistair Boyle <alistair.js.boyle(a)gmail.com> >> Date: Tue, Aug 4, 2009 at 2:26 AM >> Subject: Dell Studio 14, internal mic >> To: alsa-users(a)alsa-project.org >> >> >> Hello, >> >> I've got a new Dell Studio 14. The laptop has a built-in mic and >> speakers, two head-phone jacks and a microphone jack. Using the >> snd-hda-intel driver (kernel 2.6.30), sound plays fine through the >> built-in speakers and I can record through the mic jack. It doesn't >> look like there's a way to set the 2 headphone jack volumes, nor is >> there a way to set the capture source to the built-in mic. Finally, >> and not surprisingly, when headphones are plugged in the built-in >> speakers don't mute. >> >> (The built-in mic is the current priority.) >> >> It looks like there were tweaks added to handle the (nominally >> similar) Dell Studio 15 and 17 which can be forced on with the >> "model=dell-m6" driver option but adding this option to the module >> removed all the capture devices from the amixer options. A similar >> result occurs for dell-m6-amic and dmic. >> >> I'm at a loss as to where to look next. I'm "code competent" and with >> a pointer in the right direction I can debug. >> >> I've attached the suggested debug info. >> >> So a couple questions, in no particular order: >> >> 1) Has anyone dealt with the Dell Studio 14 laptops so far? >> >> 2) Is there any other debug output that would be helpful? >> >> 3) Where should I start looking in the code? >> >> Alistair >

2 9

Re: [alsa-devel] ASoC updates for 2.6.32
by Mark Brown 17 Sep '09

17 Sep '09

The following changes since commit cdc65fbe18aef15e92d2ebb410a189fbf956fb06: Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> Jassi (1): ASoC: S3C lrsync function made to work with IRQs disabled. Julia Lawall (1): ASoC: Clean up error handling in MPC5200 DMA setup Mark Brown (1): ASoC: Fix display of stream name in DAPM debugfs sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/fsl/mpc5200_dma.c | 33 ++++++++++++++++++++------------- sound/soc/s3c24xx/s3c-i2s-v2.c | 16 ++++++++++------ sound/soc/soc-dapm.c | 7 ++++--- 6 files changed, 34 insertions(+), 25 deletions(-)

2 1

[alsa-devel] [PATCH 1/7] ak4114 - fix errors in output selector bits
by pavel.hofman＠ivitera.com 17 Sep '09

17 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

3 17

[alsa-devel] [PATCH] S3C64XX I2S: Added machine driver for WM8580
by jassi brar 17 Sep '09

17 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 284 +++++++++++++++++++++++++++++++++++ 3 files changed, 294 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..186e1de --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,284 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* Currently, WM8580 driver doesn't support PLL-out rates + * other than those mentioned in Table-52 Page-58 of WM8580A + * manual. Some of these unavailable rates include + * {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a value that results in + * (params_rate(params) * rfs), and itself, acceptable to the both + * the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + /* Exceptions for this CPU-CODEC combination */ + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* LineIn enabled by default */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* Stereo enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_disable_pin(codec, "Center/Sub"); + snd_soc_dapm_disable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

2 7

[alsa-devel] [PATCH 1/5] ASoC: Blackfin AC97: add a few missing multichannel define handling
by Mike Frysinger 17 Sep '09

17 Sep '09

From: Cliff Cai <cliff.cai(a)analog.com> Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was merged, but two places were missed (the probe/resume functions). Restore handling of this option so it gets initialized properly. Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 2758b90..e693229 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- 1.6.5.rc1

3 7

[alsa-devel] Application and hw mixing for VoIP
by Fabien COMTE 17 Sep '09

17 Sep '09

Hi, I need some help please. I have an application that does VoIP and use OSS emulated interface. I want to : - send through VoIP application a mixed sound from mic and an another application (aplay should be ok for testing). mic -----------------> mix ---- oss emulation -----> voip application application 1 -------> (aplay) - use sound received from VoIP with HP output and an another application (arecord should be ok for testing). ---------> application 2 (arecord) voip application ----- oss emulation ----> copy ---------> HP output I tried a lot of asoundrc but they doesn't work. Every try I did, Dsnoop cannot mix sound for application 1 and copy is already in use. Could someone help me ? # wm8731 is the name of my hardware codec pcm_slave.wm8731 { type hw card 0 device 0 } pcm.from_voip { ipc_key 1025 type dmix slave.pcm wm8731 } #one called "to_voip" for capturing pcm.to_voip { ipc_key 1027 type dsnoop slave.pcm wm8731 } #and this is the real magic pcm.temp_voip { type asym playback.pcm "from_voip" capture.pcm "to_voip" } #a quick plug plugin for above device to do the converting magic pcm.voip { type plug slave.pcm "temp_voip" } #a ctl device to keep xmms happy ctl.voip { type hw card 0 } #for aoss: pcm.dsp0 { type plug slave.pcm "voip" } ctl.mixer0 { type hw card 0 } pcm.from_voip_copy { type plug slave from_voip route_policy copy } Thank you. Fabien

1 0

[alsa-devel] [PATCH] S3C64XX I2S: Added machine driver for WM8580
by Jassi 17 Sep '09

17 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. I2S controller-0 is set as default, set S3C64XX_I2S_CNTRLR to 2 inorder to use I2S_v4 controller of S3C6410. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 288 +++++++++++++++++++++++++++++++++++ 3 files changed, 298 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..6489e52 --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,288 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_CNTRLR 0 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ (12000000) + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_rate(params)) { + case 8000: + pll_out = 8192000 / 4; break; + case 11025: + pll_out = 11289600 / 4; break; + case 16000: + pll_out = 8192000 / 2; break; + case 22050: + pll_out = 11289600 / 2; break; + case 32000: + pll_out = 8192000; break; + case 44100: + pll_out = 11289600; break; + case 48000: + pll_out = 12288000; break; + case 64000: + pll_out = 24576000; break; + case 88200: + pll_out = 11289600 * 2; break; + case 96000: + pll_out = 12288000 * 2; break; + default: + return -EINVAL; + } + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* We block CODCLK from MCLK of WM8580 */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CODCLKSRC_EXT, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-ADC/DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_ADC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Don't output to the disconnected pin */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_CLKOUTSRC, + WM8580_CLKSRC_NONE); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + rfs = 256; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + rfs = 256; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* LineIn enabled by default */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* Stereo enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_disable_pin(codec, "Center/Sub"); + snd_soc_dapm_disable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_CNTRLR], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_CNTRLR], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +static void __exit smdk64xx_audio_exit(void) +{ + platform_device_unregister(smdk64xx_snd_device); +} +module_exit(smdk64xx_audio_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

3 2

[alsa-devel] Can't resume from suspend because firmware is not found ?!?
by Giuliano Pochini 17 Sep '09

17 Sep '09

I tested suspend/resume on a laptop because I failed to make it work on my G4 (grr!!). Suspend work, resume does not. From dmesg (below) it looks like the soundcard driver is asked to resume before the ATA driver, thus is cannot load the firmware. Am I overlooking something ? Suggestions ? ---- Echoaudio Indigo IOx 0000:07:00.0: resume Echoaudio Indigo IOx 0000:07:00.0: PCI INT A -> GSI 18 (level, low) -> IRQ 18 Echoaudio Indigo IOx 0000:07:00.0: enabling bus mastering ALSA echoaudio.c:2177: res res1 Echoaudio Indigo IOx 0000:07:00.0: firmware: requesting ea/indigo_iox_dsp.fw PM: Adding info for No Bus:0000:07:00.0 ata4.00: ACPI cmd ef/03:0c:00:00:00:a0 filtered out ata4.00: ACPI cmd ef/03:42:00:00:00:a0 filtered out ata4.00: configured for UDMA/33 ata2: SATA link down (SStatus 0 SControl 300) ata3: SATA link down (SStatus 0 SControl 300) ata1: SATA link up 1.5 Gbps (SStatus 113 SControl 300) ata1.00: _GTF unexpected object type 0x1 ata1.00: _GTF unexpected object type 0x1 ata1.00: configured for UDMA/100 tg3: eth0: Link is up at 10 Mbps, half duplex. tg3: eth0: Flow control is off for TX and off for RX. PM: Removing info for No Bus:0000:07:00.0 ALSA echoaudio.c:47: get_firmware(): Firmware not available (-2) pm_op(): pci_pm_resume+0x0/0x79 returns -2 PM: Device 0000:07:00.0 failed to resume: error -2 ---- -- Giuliano.

2 4

[alsa-devel] Support for Infrasonic Quartet
by Pavel Hofman 16 Sep '09

16 Sep '09

Hi Takashi, I am sending patches for Infrasonic Quartet. The card has three external clock modes therefore I had to modify ice1724.c to support multiple spdif_master types. While working on the driver, I did some minor fixes to ak4xxx-adda.c too. For proc files I added a conversion function from int to string representing its binary format (quartet.c:get_binary). If you know of an existing utility, please let me know and I will use it instead. The binary output helps a lot with debugging. The quartet patch does not comply with checkpatch.pl in 2.6.27 since the script throws a false positive for [] in #define. Thanks a lot for comments. Best regards, Pavel.

1 1

[alsa-devel] [PATCH 1/9] ASoC: remove unused #include <linux/version.h>
by Huang Weiyi 16 Sep '09

16 Sep '09

Remove unused #include <linux/version.h>('s) in sound/soc/codecs/ad1836.c sound/soc/codecs/ad1938.c sound/soc/codecs/wm8974.c Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> --- sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/wm8974.c | 1 - 3 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 3612bb9..01343dc 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -18,7 +18,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index e62b277..9a049a1 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -28,7 +28,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d8a013a..98d663a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -12,7 +12,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.6.1.3

2 1

Re: [alsa-devel] ALSA: hda - add speaker and fix headphone switch for ALC883
by Takashi Iwai 16 Sep '09

16 Sep '09

At Wed, 16 Sep 2009 14:45:12 +0200, okias wrote: > > log is with auto > > so: no channels (even 6ch), only stereo. Speaker switch work > correctly. Headphone doesn't work (nothing happening, probably > incorrect value). Disconnecting and connecting headphone jack > disabling speaker, so this is okey. int. mic works. The auto model doesn't support multi-channel mode for shared jacks, so it's expected. Could you try the patch below with the very latest alsa-driver snapshot tarball? ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Apply it on alsa-driver/alsa-kernel directory with patch -p2 option. thanks, Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7702f15..1296058 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7927,8 +7927,9 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), @@ -7947,8 +7948,9 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = { static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -7960,6 +7962,15 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9167,7 +9178,8 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc882_targa_automute, }, [ALC883_TARGA_8ch_DIG] = { - .mixers = { alc883_base_mixer, alc883_chmode_mixer }, + .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer, + alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, alc883_targa_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids),

1 0

[alsa-devel] Ethernet AVB - XMOS reference desing
by errordeveloper＠gmail.com 16 Sep '09

16 Sep '09

Hello. I m just getting my head around the whole lot of ethernet audio protocols avaliable, it's a big deal of brands really .. but now it looks like there will be a new rather standard protocol - ieee802.1 ethernet AV bridge. here is a reference design which includes driver source-code. i haven't quite figured is tha tlicene is comptible with gpl really.. but i suppose you don't need to use that code, just look at it as a piece of documentation .. it could be so great to have ethernet audio in linux! i have seen the audio science interface and draver exists, but they are using cobra-net ..which is probably good, but it's not as standard as ieee AVB. here you can find an article about the current status of the protocol mess: http://prosoundnewseurope.com/pdf/PSNLive/PSNLive_2009.pdf (page 28) best regards, -- ilya .d

1 0

Re: [alsa-devel] ALSA: hda - add speaker and fix headphone switch for ALC883
by Takashi Iwai 16 Sep '09

16 Sep '09

At Wed, 16 Sep 2009 14:16:11 +0200, okias wrote: > > > At Tue, 15 Sep 2009 21:13:00 +0200, > > okias wrote: > >> > >> ALSA: hda - add speaker and fix headphone switch for ALC883 > >> > >> This patch changing "Headphone Playback Switch" from 0x1b to 0x14 > >> (documentation says "Port-D", 3,5 jack) and adding "Speaker Playback > >> Switch" 0x1b (Port-E, LINE2). > >> > >> Signed-off-by: David Heidelberger <d.okias(a)gmail.com> > > > > Thanks for the patch. > > For which machine is this change necessary? Could you give > > alsa-info.sh output (run with --no-upload option)? > > > > yes, attaching to message Thanks. Just wondering whether model=auto works properly for your device. If it does, we can get rid of the quirk entry but let driver parsing the BIOS pin-configs instead. So far, the quirk for 8ch mode is used although BIOS tells that there is no 8-channel outputs. Takashi > > > Apart from that your patch is for older codes, it's not good to > > change alc88*_base_mixer[]. This definition is for the fixed pin > > design. If it doesn't match with your machine, it means that the > > quirk is wrong (or the auto-parser is buggy). Better to define the > > fixed mixer array, and don't change the common definition just for a > > single model. > > > > > > Takashi > > > >> > >> --- a/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:37.067310114 +0200 > >> +++ b/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:43.444437970 +0200 > >> @@ -7979,6 +7979,8 @@ > >> static struct snd_kcontrol_new alc883_base_mixer[] = { > >> HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), > >> HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), > >> + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), > >> + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), > >> HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), > >> HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), > >> HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, > >> HDA_OUTPUT), > >> @@ -7987,7 +7989,6 @@ > >> HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), > >> HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), > >> HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), > >> - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), > >> HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), > >> HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), > >> HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > >> [2 alc883_speaker-headphone_fix.patch <application/octet-stream (base64)>] > >> > >> [3 <text/plain; us-ascii (7bit)>] > >> _______________________________________________ > >> Alsa-devel mailing list > >> Alsa-devel(a)alsa-project.org > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > [2 alsa-info.txt.Y4QfxkuHwO <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH 1/3] ASoC: TWL6030: Add twl6030 codec driver
by Lopez Cruz, Misael 16 Sep '09

16 Sep '09

Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary digital audio interface called McPDM. TWL6030 codec has two power modes: low-power and high-performance, this initial version only supports high-performance mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc-dai.h | 3 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 670 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 95 +++++++ 5 files changed, 774 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..8b0d04b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -35,6 +35,9 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J +/* propietary formats */ +#define SND_SOC_DAIFMT_MCPDM 0x10 /* Texas Instruments McPDM */ + /* * DAI Clock gating. * diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..0effb52 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL6030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -114,6 +115,9 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TWL4030 tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..b70c8a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -13,6 +13,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -50,6 +51,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..c5e76fa --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,670 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if ((reg < TWL6030_REG_INTID) || (reg > TWL6030_REG_ACCCTL)) + /* read-only registers */ + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL6030_MODULE_AUDIO, value, reg); +} + +static void twl6030_init_chip(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + /* set all audio section registers to reasonable defaults */ + /* avoid read-only (ASICID, ASICREV, STATUS) and non-used registers */ + for (i = TWL6030_REG_INTMR; i <= TWL6030_REG_DLB; i++) + twl6030_write(codec, i, cache[i]); + for (i = TWL6030_REG_TRIM1; i < TWL6030_REG_ACCCTL; i++) + twl6030_write(codec, i, cache[i]); +} + +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + struct snd_soc_device *socdev = codec->socdev; + struct twl6030_setup_data *setup = socdev->codec_data; + + setup->codec_enable(1); + + /* sync registers updated during power-up sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); +} + +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + struct snd_soc_device *socdev = codec->socdev; + struct twl6030_setup_data *setup = socdev->codec_data; + + setup->codec_enable(0); + + /* sync registers updated during power-down sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier (Attenuator) Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Amplifier Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxers for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + twl6030_power_up(codec); + break; + case SND_SOC_BIAS_PREPARE: + twl6030_power_up(codec); + break; + case SND_SOC_BIAS_STANDBY: + twl6030_power_up(codec); + break; + case SND_SOC_BIAS_OFF: + twl6030_power_down(codec); + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl6030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int rate, format; + + /* hardware dai (McPDM) requires bit stream of twice + * the actual rate, application processor (audio backend) + * should upsample accordingly + */ + rate = params_rate(params); + switch (rate) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unknown rate %d\n", rate); + return -EINVAL; + } + + /* the sample lenght handled by codec at A-D and D-A + * conversion is 16-bits, but the dai requires 32-bits + */ + format = params_format(params); + if (format != SNDRV_PCM_FORMAT_S32_LE) { + dev_err(codec->dev, "unknown format %d\n", format); + return -EINVAL; + } + + return 0; +} + +static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 hppll, lppll; + + hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppll &= TWL6030_HPLLRST; + + switch (freq) { + case 12000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_12000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 19200000: + /* MCLK input, PLL disabled */ + hppll = TWL6030_MCLK_19200KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLBP; + break; + case 26000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_26000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer input, PLL disabled */ + hppll = TWL6030_MCLK_38400KHZ + | TWL6030_HPLLSQRENA + | TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + + /* Disable LPPLL and select HPPLL */ + lppll = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + return 0; +} + +static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + int format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* propietary dai format */ + if (format != SND_SOC_DAIFMT_MCPDM) + return -EINVAL; + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .hw_params = twl6030_hw_params, + .set_sysclk = twl6030_set_dai_sysclk, + .set_fmt = twl6030_set_dai_fmt, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialize the driver + */ +static int twl6030_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + int ret = 0; + + dev_info(codec->dev, "TWL6030 Audio Codec init\n"); + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + twl6030_init_chip(codec); + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *twl6030_socdev; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + codec->dev = &pdev->dev; + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + twl6030_socdev = socdev; + twl6030_init(socdev); + + return 0; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dev_info(codec->dev, "TWL6030 Audio Codec remove\n"); + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __init twl6030_modinit(void) +{ + return snd_soc_register_dai(&twl6030_dai); +} +module_init(twl6030_modinit); + +static void __exit twl6030_exit(void) +{ + snd_soc_unregister_dai(&twl6030_dai); +} +module_exit(twl6030_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..a83f9e7 --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,95 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_FIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +struct twl6030_setup_data { + void (*codec_enable)(int enable); +}; + +#endif /* End of __TWL6030_H__ */ -- 1.5.4.3

2 5

[alsa-devel] alsamixer is not able to set pulseaudio master volume
by Ozan Çağlayan 16 Sep '09

16 Sep '09

Hi, I've just noticed that using alsamixer to set pulseaudio's master volume is not working anymore. I switched back to an earlier alsa-utils snapshot (20090624) still the same. I didn't do any changes to pulseaudio for a long time, it's 0.9.15 with the proposed patchset of Lennart. alsamixer is launched, the master channel is displayed but pressing Up and Down doesn't change anything on the UI. Pressing M changes 'OO' to 'MM'. Here's what happens on PA's side when I click UP/DOWN: I: alsa-source.c: Device suspended... I: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0 idle for too long, suspending ... I: alsa-sink.c: Device suspended... I: client.c: Created 1 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 15, local 15 I: protocol-native.c: Got credentials: uid=1000 gid=100 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: module-augment-properties.c: Looking for .desktop file for alsamixer I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. .. .. Synced.

2 2

[alsa-devel] ALSA: hda - add speaker and fix headphone switch for ALC883
by okias 16 Sep '09

16 Sep '09

ALSA: hda - add speaker and fix headphone switch for ALC883 This patch changing "Headphone Playback Switch" from 0x1b to 0x14 (documentation says "Port-D", 3,5 jack) and adding "Speaker Playback Switch" 0x1b (Port-E, LINE2). Signed-off-by: David Heidelberger <d.okias(a)gmail.com> --- --- a/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:37.067310114 +0200 +++ b/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:43.444437970 +0200 @@ -7979,6 +7979,8 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), @@ -7987,7 +7989,6 @@ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),

2 2

[alsa-devel] Radeon RV730 (Radeon HD 4670) not playing back
by Anders Kirchenbauer 16 Sep '09

16 Sep '09

I am unable to have sound playback through HDMI, presumably because there is no PCM codec support with the current state of my driver. An alsa-info.sh dump is below, and below that a pastebin.ca of speaker-test http://www.alsa-project.org/db/?f=2e2c6ba9f4052ef0a2defc25688067d597737af9 http://pastebin.ca/1567621 Also, changing the PCI ID to 1002aa38 in the two places in patch_atihdmi.c resulted in no sound or mixer device being available when the drivers were loaded. Went back to distro-default drivers afterwards. Only IEC958 Mute/Unmute control works, rest are permission denied. All permission denied are denied under user and root accounts. alsamixer defines card as "ATI R6xx HDMI"

1 0

[alsa-devel] [PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 15 Sep '09

15 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Note: this patch was created based on temp/asoc branch. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 4 ++-- sound/soc/davinci/davinci-evm.c | 7 ++++--- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad..047ee39 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..7ccbe66 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci-evm audio machine driver */ +/* davinci dm6446, dm355 or dm365 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -243,7 +244,7 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm()) { + if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { -- 1.6.0.4

3 3

[alsa-devel] [PATCH 1/6] AK4114 - fix output selector bits, fix spelling
by pavel.hofman＠ivitera.com 15 Sep '09

15 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

3 17

[alsa-devel] [PATCH v2] use set_channel_map api to reorder channels for AD1938 and AD1836
by Barry Song 15 Sep '09

15 Sep '09

From: Barry Song <barry.song(a)analog.com> Signed-off-by: Barry Song <barry.song(a)analog.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 7 +++++ sound/soc/blackfin/bf5xx-ad1938.c | 9 ++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 ++++-- sound/soc/blackfin/bf5xx-tdm.c | 45 +++++++++++++++++++++++++++++------ sound/soc/blackfin/bf5xx-tdm.h | 11 ++++++++ 5 files changed, 69 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e3..0f45a3f 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..2ef1e50 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..a8c73cb 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..600987d 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_tdm.tx_map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_tdm.rx_map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..04189a1 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define BFIN_TDM_DAI_MAX_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; + unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

2 1

[alsa-devel] [PATCH] alsa-lib: mixer/simple_none: fix softfloat compilation
by Manuel Lauss 15 Sep '09

15 Sep '09

mixer/simple_none.c uses HAVE_SOFT_FLOAT tests to work around sqrt() uses but the definition in config.h is never picked up, leading to link errors in libasound.so when built with --with-softfloat. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Tested on MIPS. The pcm_ladspa plugin still calls 'sqrt()' and should be excluded from the build if --with-softfloat is specified. src/mixer/mixer_simple.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/src/mixer/mixer_simple.h b/src/mixer/mixer_simple.h index e88b007..bc9d20c 100644 --- a/src/mixer/mixer_simple.h +++ b/src/mixer/mixer_simple.h @@ -19,6 +19,7 @@ * */ +#include <config.h> #include "mixer_abst.h" /* make local functions really local */ -- 1.6.5.rc1

1 0

[alsa-devel] [PATCH 1/2] ak4113 build support
by pavel.hofman＠ivitera.com 15 Sep '09

15 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/i2c/other/Makefile b/i2c/other/Makefile index 37f1603..a86990e 100644 --- a/i2c/other/Makefile +++ b/i2c/other/Makefile @@ -5,7 +5,7 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf -export-objs += ak4xxx-adda.o ak4114.o ak4117.o pt2258.o tea575x-tuner.o +export-objs += ak4xxx-adda.o ak4113.o ak4114.o ak4117.o pt2258.o tea575x-tuner.o include $(SND_TOPDIR)/alsa-kernel/i2c/other/Makefile diff --git a/i2c/other/ak4113.c b/i2c/other/ak4113.c new file mode 100644 index 0000000..41cb4ce --- /dev/null +++ b/i2c/other/ak4113.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/i2c/other/ak4113.c" diff --git a/utils/insert b/utils/insert index af911d1..bfbbb83 100755 --- a/utils/insert +++ b/utils/insert @@ -395,6 +395,7 @@ case "$1" in #insert snd-i2c$SUFF #insert snd-cs8427$SUFF insert snd-ak4114$SUFF + insert snd-ak4113$SUFF insert snd-ak4xxx-adda$SUFF insert snd-ice17xx-ak4xxx$SUFF insert snd-ice1724$SUFF enable=1,1 -- 1.5.6.3

1 1

[alsa-devel] Support for Infrasonic Quartet
by Pavel Hofman 15 Sep '09

15 Sep '09

Hi Takashi, I am sending patches for Infrasonic Quartet. The card has three external clock modes therefore I had to modify ice1724.c to support multiple spdif_master types. While working on the driver, I did some minor fixes to ak4xxx-adda.c too. For proc files I added a conversion function from int to string representing its binary format (quartet.c:get_binary). If you know of an existing utility, please let me know and I will use it instead. The binary output helps a lot with debugging. The quartet patch does not comply with checkpatch.pl in 2.6.27 since the script throws a false positive for [] in #define. Thanks a lot for comments. Best regards, Pavel.

1 0

[alsa-devel] [PATCH] RFC: ASoC: Davinci: Fix data_type divide by zero
by Troy Kisky 15 Sep '09

15 Sep '09

This is a proposed fix for the data_type divide by 0 error. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- This applies on top of my last patchset to Kevin's temp/asoc branch sound/soc/davinci/davinci-i2s.c | 18 +++++++----------- sound/soc/davinci/davinci-mcasp.h | 7 ++++++- sound/soc/davinci/davinci-pcm.c | 14 ++++++-------- 3 files changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 5f61016..252c3ea 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -123,13 +123,18 @@ static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { }; struct davinci_mcbsp_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to an array of struct davinci_pcm_dma_params ** in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params *dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 int mode; u32 pcr; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; /* * Combining both channels into 1 element will at least double the * amount of time between servicing the dma channel, increase @@ -255,14 +260,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) toggle_clock(dev, playback); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - #define DEFAULT_BITPERSAMPLE 16 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -393,7 +390,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_soc_dai *codec_dai = dev->codec_dai; struct snd_interval *i = NULL; @@ -407,6 +403,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int right_first = 0; int convert_mono_stereo = 0; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); bits_per_sample = snd_interval_value(i); @@ -529,7 +526,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 554354c..c6ea04d 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,10 +39,15 @@ enum { }; struct davinci_audio_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to an array of struct davinci_pcm_dma_params ** in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params *dma_params[2]; void __iomem *base; int sample_rate; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; unsigned int codec_fmt; /* McASP specific data */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2729a80..0bd3b2e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -265,8 +265,7 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = runtime->private_data; struct snd_dma_buffer *iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *dma_data = prtd->params; unsigned int data_type = dma_data->data_type; unsigned int acnt = dma_data->acnt; unsigned int convert_mono_stereo = dma_data->convert_mono_stereo; @@ -482,15 +481,12 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *dma_data = prtd->params; int link; if (!dma_data) return -ENODEV; - prtd->params = dma_data; - /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(dma_data->channel, davinci_pcm_dma_irq, substream, EVENTQ_0); @@ -660,7 +656,8 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) static int davinci_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params **pa = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = pa[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; struct snd_pcm_hardware *ppcm; @@ -668,7 +665,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, dma_data->sram_size, ppcm); + allocate_sram(substream, dma_params->sram_size, ppcm); snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -681,6 +678,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = dma_params; prtd->asp_channel = -1; prtd->asp_link[0] = prtd->asp_link[1] = -1; prtd->ram_channel = -1; -- 1.5.6.3

3 4

[alsa-devel] SPDIF with X-Fi
by Daniel L. Miller 14 Sep '09

14 Sep '09

Perhaps I'm going about this the wrong way (if so, please tell me), but... My X-Fi card is now working with the new driver. I am connected to my amplifier via a fiber optic cable. Sending standard two-channel audio to the "front" alsa plug gets directed to the spdif output and sounds great. But I want to have my full surround active - particularly for movies already encoded with dts and/or ac-3. Thus far, I have not been able to get mplayer to send digital audio - I either get mplayer errors or silence. Once, and only once, I got xbmc to work with digital settings - never since. So I know it SHOULD be possible - I'm just missing something. Is anyone getting this to work? -- Daniel

1 0

[alsa-devel] iecset with X-Fi
by Daniel L. Miller 14 Sep '09

14 Sep '09

Is the iecset utility supposed to do anything with X-Fi cards and the new driver? For me, it reports status - but seems to ignore any parameters to change the settings. -- Daniel

1 0

[alsa-devel] [PATCH] WM8580 DAI: Debugged
by jassisinghbrar＠gmail.com 14 Sep '09

14 Sep '09

From: Jassi <jassi.brar(a)samsung.com> Debugged improper definition of 'struct snd_soc_dai wm8580_dai' Also, implemented various clock sourcing options for WM8580 blocks. Signed-Off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 116 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/wm8580.h | 17 ++++--- 2 files changed, 93 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..105d452 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -106,9 +106,16 @@ /* CLKSEL (register 8h) */ #define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_MCLK 0x00 #define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 #define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK 0x00 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLA 0x04 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLB 0x08 +#define WM8580_CLKSEL_ADC_CLKSEL_MCLK 0x0c + /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 #define WM8580_AIF_RATE_128 0x0 @@ -121,15 +128,16 @@ #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 -#define WM8580_AIF_BCLKSEL_128 0x08 -#define WM8580_AIF_BCLKSEL_256 0x10 +#define WM8580_AIF_BCLKSEL_32 0x08 +#define WM8580_AIF_BCLKSEL_16 0x10 #define WM8580_AIF_BCLKSEL_SYSCLK 0x18 #define WM8580_AIF_MS 0x20 #define WM8580_AIF_CLKSRC_MASK 0xc0 +#define WM8580_AIF_CLKSRC_ADCMCLK 0x00 #define WM8580_AIF_CLKSRC_PLLA 0x40 -#define WM8580_AIF_CLKSRC_PLLB 0x40 +#define WM8580_AIF_CLKSRC_PLLB 0x80 #define WM8580_AIF_CLKSRC_MCLK 0xc0 /* AIF control 2 (registers ch-eh) */ @@ -630,6 +638,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_DAC_CLKSEL_MCLK; break; case WM8580_CLKSRC_PLLA: @@ -646,6 +655,33 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_CLKSEL, reg); break; + case WM8580_ADC_CLKSEL: + reg = snd_soc_read(codec, WM8580_CLKSEL); + reg &= ~WM8580_CLKSEL_ADC_CLKSEL_MASK; + + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_MCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLB; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_CLKSEL, reg); + break; + case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -672,6 +708,32 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; + case WM8580_PAIF_CLKSEL: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_CLKSRC_MASK; + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_AIF_CLKSRC_ADCMCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_AIF_CLKSRC_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_AIF_CLKSRC_PLLB; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_AIF_CLKSRC_MCLK; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + default: return -EINVAL; } @@ -731,7 +793,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8580_dai_ops_playback = { +static struct snd_soc_dai_ops wm8580_dai_ops = { .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, @@ -739,38 +801,26 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = { .digital_mute = wm8580_digital_mute, }; -static struct snd_soc_dai_ops wm8580_dai_ops_capture = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, -}; - struct snd_soc_dai wm8580_dai[] = { - { - .name = "WM8580 PAIFRX", - .id = 0, - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 6, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = WM8580_FORMATS, - }, - .ops = &wm8580_dai_ops_playback, + { + .name = "WM8580 PAIF", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8580_FORMATS, }, - { - .name = "WM8580 PAIFTX", - .id = 1, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = WM8580_FORMATS, - }, - .ops = &wm8580_dai_ops_capture, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8580_FORMATS, }, + .ops = &wm8580_dai_ops, + }, }; EXPORT_SYMBOL_GPL(wm8580_dai); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..65f9225 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -20,13 +20,16 @@ #define WM8580_MCLK 1 #define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 - -#define WM8580_CLKSRC_MCLK 1 -#define WM8580_CLKSRC_PLLA 2 -#define WM8580_CLKSRC_PLLB 3 -#define WM8580_CLKSRC_OSC 4 -#define WM8580_CLKSRC_NONE 5 +#define WM8580_ADC_CLKSEL 3 +#define WM8580_CLKOUTSRC 4 +#define WM8580_PAIF_CLKSEL 5 + +#define WM8580_CLKSRC_ADCMCLK 0 +#define WM8580_CLKSRC_MCLK 1 +#define WM8580_CLKSRC_PLLA 2 +#define WM8580_CLKSRC_PLLB 3 +#define WM8580_CLKSRC_OSC 4 +#define WM8580_CLKSRC_NONE 5 #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 -- 1.6.2.5

3 3

[alsa-devel] [PATCH v4 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 14 Sep '09

14 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Registers the platform device for McBSP on dm365. 2) Add platform data to DM365 EVM board file. 3) Set i2c address for audio codec at DM365 EVM board file. Note: This patch was created based on temp/asoc branch Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 7 ++++ arch/arm/mach-davinci/dm365.c | 45 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 56 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index f6adf79..77f815c 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -177,6 +177,8 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static struct snd_platform_data dm365_evm_snd_data; + static struct i2c_board_info i2c_info[] = { { I2C_BOARD_INFO("dm365evm_keys", 0x25), @@ -185,6 +187,9 @@ static struct i2c_board_info i2c_info[] = { I2C_BOARD_INFO("24c256", 0x50), .platform_data = &eeprom_info, }, + { + I2C_BOARD_INFO("tlv320aic3x", 0x18), + }, }; static struct davinci_i2c_platform_data i2c_pdata = { @@ -476,6 +481,8 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); + + dm365_init_asp(&dm365_evm_snd_data); } static __init void dm365_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index e815174..c8bff14 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -32,6 +32,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -456,7 +457,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("voice_codec", NULL, &voicecodec_clk), - CLK("soc-audio.0", NULL, &asp0_clk), + CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -603,6 +604,9 @@ INT_CFG(DM365, INT_IMX1_ENABLE, 24, 1, 1, false) INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) + +EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) +EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif }; @@ -806,6 +810,31 @@ static struct platform_device dm365_edma_device = { .resource = edma_resources, }; +static struct resource dm365_asp_resources[] = { + { + .start = DAVINCI_DM365_ASP0_BASE, + .end = DAVINCI_DM365_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_asp_device = { + .name = "davinci-asp", + .id = 0, + .num_resources = ARRAY_SIZE(dm365_asp_resources), + .resource = dm365_asp_resources, +}; + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -907,6 +936,20 @@ static struct davinci_soc_info davinci_soc_info_dm365 = { .sram_len = SZ_32K, }; +void __init dm365_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_MCBSP0_BDX); + davinci_cfg_reg(DM365_MCBSP0_X); + davinci_cfg_reg(DM365_MCBSP0_BFSX); + davinci_cfg_reg(DM365_MCBSP0_BDR); + davinci_cfg_reg(DM365_MCBSP0_R); + davinci_cfg_reg(DM365_MCBSP0_BFSR); + davinci_cfg_reg(DM365_EVT2_ASP_TX); + davinci_cfg_reg(DM365_EVT3_ASP_RX); + dm365_asp_device.dev.platform_data = pdata; + platform_device_register(&dm365_asp_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..fef12b9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -11,6 +11,9 @@ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 +/* Bases of dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of dm646x register banks */ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 09db434..2291c0d 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM365_EMAC_BASE (0x01D07000) #define DM365_EMAC_CNTRL_OFFSET (0x0000) @@ -25,5 +26,6 @@ #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm365_init(void); +void __init dm365_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ -- 1.6.0.4

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[PATCH] ASoC: Make WM8580 DAI clocking more standard
by Mark Brown 14 Sep '09

14 Sep '09

Move the WM8580 driver over to more of a standard clocking setup method, specifying the master clock rate for the DAI using the set_sysclk() API and configuring the LRCLK automatically using that. This is an initial update only to present external APIs, further work that needs doing includes enforcing the constraints flowing from the SYSCLK selection, automatic tie in with PLL configuration and BCLK configuration (the current BCLK configuration is compatible with all sample rates). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 121 +++++++++++++++++++++++++++++++-------------- sound/soc/codecs/wm8580.h | 4 +- 2 files changed, 85 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..dffc093 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -105,19 +105,18 @@ #define WM8580_PLLB4_CLKOUTSRC_OSCCLK 0x180 /* CLKSEL (register 8h) */ -#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 -#define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 -#define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_SHIFT 0 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_SHIFT 2 + +#define WM8580_CLKSEL_PLLA 0x01 +#define WM8580_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_MCLK 0x03 /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 -#define WM8580_AIF_RATE_128 0x0 -#define WM8580_AIF_RATE_192 0x1 -#define WM8580_AIF_RATE_256 0x2 -#define WM8580_AIF_RATE_384 0x3 -#define WM8580_AIF_RATE_512 0x4 -#define WM8580_AIF_RATE_768 0x5 -#define WM8580_AIF_RATE_1152 0x6 +#define WM8580_AIF_RATE_SHIFT 0x2 #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 @@ -203,6 +202,7 @@ struct wm8580_priv { u16 reg_cache[WM8580_MAX_REGISTER + 1]; struct pll_state a; struct pll_state b; + int sysclk[WM8580_NUM_DAI]; }; static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); @@ -476,6 +476,10 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } +static int lrclk_ratios[] = { + 128, 192, 256, 384, 512, 768, 1152 +}; + /* * Set PCM DAI bit size and sample rate. */ @@ -483,12 +487,31 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = dai->codec; + struct wm8580_priv *priv = codec->private_data; u16 paifb = snd_soc_read(codec, WM8580_PAIF3 + dai->id); + int sysclk, i; + + /* SYSCLK needs to be configured prior to this being called. */ + if (!priv->sysclk[dai->id]) { + dev_err(dai->dev, "SYSCLK not configured\n"); + return -EINVAL; + } + sysclk = priv->sysclk[dai->id]; + dev_dbg(dai->dev, "SYSCLK is %dHz\n", sysclk); + + paifb &= ~(WM8580_AIF_LENGTH_MASK | WM8580_AIF_RATE_MASK); + + /* Look for an exact match; the steps are too big otherwise */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (params_rate(params) == sysclk / lrclk_ratios[i]) + break; + } + if (i == ARRAY_SIZE(lrclk_ratios)) + return -EINVAL; + dev_dbg(dai->dev, "LRCLK ratio %d\n", lrclk_ratios[i]); + paifb |= i << WM8580_AIF_RATE_SHIFT; - paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -624,28 +647,6 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; - case WM8580_DAC_CLKSEL: - reg = snd_soc_read(codec, WM8580_CLKSEL); - reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK; - - switch (div) { - case WM8580_CLKSRC_MCLK: - break; - - case WM8580_CLKSRC_PLLA: - reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLA; - break; - - case WM8580_CLKSRC_PLLB: - reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLB; - break; - - default: - return -EINVAL; - } - snd_soc_write(codec, WM8580_CLKSEL, reg); - break; - case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -679,6 +680,48 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, return 0; } +static int wm8580_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8580_priv *priv = codec->private_data; + int mask, val, ret; + + switch (clk_id) { + case WM8580_CLKSRC_MCLK: + val = WM8580_CLKSEL_MCLK; + break; + case WM8580_CLKSRC_PLLA: + val = WM8580_CLKSEL_PLLA; + break; + case WM8580_CLKSRC_PLLB: + val = WM8580_CLKSEL_PLLB; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case WM8580_DAI_PAIFRX: + mask = WM8580_CLKSEL_DAC_CLKSEL_MASK; + break; + + case WM8580_DAI_PAIFTX: + mask = WM8580_CLKSEL_ADC_CLKSEL_MASK; + val <<= WM8580_CLKSEL_ADC_CLKSEL_SHIFT; + break; + + default: + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, WM8580_CLKSEL, mask, val); + if (ret == 0) + priv->sysclk[dai->id] = freq; + + return ret; +} + static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -736,6 +779,7 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, + .set_sysclk = wm8580_set_dai_sysclk, .digital_mute = wm8580_digital_mute, }; @@ -743,13 +787,14 @@ static struct snd_soc_dai_ops wm8580_dai_ops_capture = { .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, + .set_sysclk = wm8580_set_dai_sysclk, .set_pll = wm8580_set_dai_pll, }; struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", - .id = 0, + .id = WM8580_DAI_PAIFRX, .playback = { .stream_name = "Playback", .channels_min = 1, @@ -761,7 +806,7 @@ struct snd_soc_dai wm8580_dai[] = { }, { .name = "WM8580 PAIFTX", - .id = 1, + .id = WM8580_DAI_PAIFTX, .capture = { .stream_name = "Capture", .channels_min = 2, diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..f03ae9e 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -19,8 +19,7 @@ #define WM8580_PLLB 2 #define WM8580_MCLK 1 -#define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 +#define WM8580_CLKOUTSRC 2 #define WM8580_CLKSRC_MCLK 1 #define WM8580_CLKSRC_PLLA 2 @@ -30,6 +29,7 @@ #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 +#define WM8580_NUM_DAI 2 extern struct snd_soc_dai wm8580_dai[]; extern struct snd_soc_codec_device soc_codec_dev_wm8580; -- 1.6.3.3

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[alsa-devel] [PATCH 3/3] ASoC: TWL6030: Handle power-up seq completion thru audio interrupt
by Lopez Cruz, Misael 14 Sep '09

14 Sep '09

NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. When the codec is powered-up through external AUDPWRON line it will start its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. CODEC driver starts a wait_for_completion just after calling external power-up callback. It's signaled as complete when servicing READYINT interrupt. MACHINE drivers should request IRQ line used in corresponding hardware platform and initialize workqueue and completion structs of twl6030_setup_data as well. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 103 +++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl6030.h | 17 +++++++ 2 files changed, 113 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 0f2269e..dc85399 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -43,6 +43,7 @@ /* codec private data */ struct twl6030_priv_data { + int codec_powered; unsigned int sysclk; }; @@ -54,7 +55,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -127,6 +128,23 @@ static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, } /* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg > TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* * write to the twl6030 register space */ static int twl6030_write(struct snd_soc_codec *codec, @@ -159,27 +177,89 @@ static void twl6030_init_chip(struct snd_soc_codec *codec) static void twl6030_power_up(struct snd_soc_codec *codec) { struct snd_soc_device *socdev = codec->socdev; + struct twl6030_priv_data *priv = codec->private_data; struct twl6030_setup_data *setup = socdev->codec_data; + if (priv->codec_powered) + return; + setup->codec_enable(1); + /* wait for ready interrupt */ + wait_for_completion(&setup->ready_completion); + /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); } static void twl6030_power_down(struct snd_soc_codec *codec) { struct snd_soc_device *socdev = codec->socdev; + struct twl6030_priv_data *priv = codec->private_data; struct twl6030_setup_data *setup = socdev->codec_data; setup->codec_enable(0); + udelay(500); /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + + priv->codec_powered = 0; +} + +/* audio interrupt handler */ +irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct twl6030_setup_data *setup = data; + + schedule_work(&setup->audint_work); + + /* disable audint irq to let workqueue to execute */ + disable_irq_nosync(irq); + + return IRQ_HANDLED; +} + +void twl6030_naudint_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct twl6030_setup_data *setup; + struct twl6030_priv_data *priv; + u8 intid; + + setup = container_of(work, struct twl6030_setup_data, audint_work); + codec = setup->codec; + priv = codec->private_data; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + priv->codec_powered = 1; + complete(&setup->ready_completion); + break; + default: + dev_err(codec->dev, "unknown twl6030 audio interrupt\n"); + } + + enable_irq(setup->irq); } /* set headset dac and driver power mode */ @@ -679,6 +759,7 @@ static int twl6030_resume(struct platform_device *pdev) static int twl6030_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_setup_data *setup = socdev->codec_data; int ret = 0; dev_info(codec->dev, "TWL6030 Audio Codec init\n"); @@ -703,6 +784,14 @@ static int twl6030_init(struct snd_soc_device *socdev) goto pcm_err; } + /* platform setup data is required for power-off/off the device */ + if (setup == NULL) { + printk(KERN_ERR "twl6030: platform setup data missing\n"); + goto pcm_err; + } + + setup->codec = codec; + /* power on device */ twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 9e49c85..1e1fb3f 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -64,6 +64,16 @@ #define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -105,8 +115,15 @@ extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; +irqreturn_t twl6030_naudint_handler(int irq, void *data); +void twl6030_naudint_work(struct work_struct *work); + struct twl6030_setup_data { void (*codec_enable)(int enable); + int irq; + struct work_struct audint_work; + struct completion ready_completion; + struct snd_soc_codec *codec; }; #endif /* End of __TWL6030_H__ */ -- 1.5.4.3

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