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.

5 7

[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

1 0

[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"},

2 1

[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

2 3

[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

1 0

[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(-)

2 1

[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(-)

2 1

[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; }

2 1

[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

1 0

[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

1 0

[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

2 1

[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

2 1

[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(-)

2 1

[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