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

[alsa-devel] How to support multiple audio codecs on a single board
by Aggarwal, Anuj 31 Aug '09

31 Aug '09

Hi, I have 3 instances of AIC23 on my board and all are connected to different I2C/McBSP and hence can be used independently. How should I use/modify the current AIC23 driver and write the asoc board-specific file to enable all the AIC23 instances? Is there any similar implementation in the audio subsystem which I can look as a reference? Regards, Anuj Aggarwal

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[alsa-devel] [PATCH] ALSA: allocation may fail in snd_pcm_oss_change_params()
by Roel Kluin 31 Aug '09

31 Aug '09

Allocation may fail, show if it did. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index dbe406b..ea91a0a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1047,6 +1047,10 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) runtime->oss.prepare = 1; vfree(runtime->oss.buffer); runtime->oss.buffer = vmalloc(runtime->oss.period_bytes); + if (runtime->oss.buffer == NULL) { + err = -ENOMEM; + goto failure; + } runtime->oss.buffer_used = 0; if (runtime->dma_area) snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes));

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[alsa-devel] [PATCH V2 2/3] add set_tdm_slot in tdm dai to define the relationship between audio channel and tdm slot
by Barry Song 31 Aug '09

31 Aug '09

For ad1938, the audio channel n just uses slot n, but for ad1836, it's different, channel 0 uses slot 0, channel 1 uses slot 4, channels 2 uses slot1, ... So add set_tdm_slot entry and use the mask field to define the relationship between audio channel and slot number. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad1938.c | 7 ++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +++++++++--- sound/soc/blackfin/bf5xx-tdm.c | 29 +++++++++++++++++++++-------- sound/soc/blackfin/bf5xx-tdm.h | 10 ++++++++++ 4 files changed, 46 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..05163ea 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -74,8 +74,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI slots, 8 channels, mask is defined as slot seq */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x76543210, 8, 32); + if (ret < 0) + 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, 8, 32); 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..044ef45 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; @@ -186,9 +189,11 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst = (unsigned int *)substream->runtime->dma_area; dst += pos * 8; + while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + (((tdm_port->slot_seq >> + (4 * i)) & 0xF))) = *src++; dst += 8; } } else { @@ -198,7 +203,8 @@ 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->slot_seq >> + (4 * i)) & 0xF))); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..0f4ee15 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 +173,24 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots, int width) +{ + /* + * For ad1938, the audio channel n just uses slot n, but for + * ad1836, it's different, channel 0 uses slot 0, channel 1 + * uses slot 4, channels 2 uses slot1, ... So add set_tdm_slot + * entry and use the mask field to define the relationship + * between audio channel and slot number. + * + * bit[0..3] means the slot channel 0 will use + * bit[4..7] means the slot channel 1 will use + * ... + */ + bf5xx_tdm.slot_seq = mask; + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +245,7 @@ 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_tdm_slot = bf5xx_tdm_set_tdm_slot, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +311,8 @@ 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..701c769 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,16 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + /* which slot used for the corresponding audio channel? */ + int slot_seq; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

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Re: [alsa-devel] [alsa-cvslog] alsa-tools: ALSA tools repository branch, master now at v1.0.20-5-ga539d8b
by Takashi Iwai 31 Aug '09

31 Aug '09

At Mon, 31 Aug 2009 12:32:40 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > commit a539d8b261699856095154a65e17b3b3b73155b6 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Mon Aug 31 12:16:48 2009 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Mon Aug 31 12:32:35 2009 +0200 > > qlo10k1: remove acinlude.m4 (QT configuration hell) - use PKG_CHECK_MODULES This breaks the build with Qt installed in the non-standard paths like /opt. Qt provides no pkgconfig files, so it can't be fixed in that way (and this program isn't for Gtk!) Please revert it. Takashi

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[alsa-devel] missing control in alsamixer capture for CA0110-IBG
by Susan Cragin 31 Aug '09

31 Aug '09

I have a Creative X-fi pci card (HD-Audio Generic) that uses Creative CA0110-IBG. Under alsamixer / Capture, there used to be two adjustments / sliders. The first was for "mic" and controlled the volume. I don't remember what the second was called, but it was very useful. It's default setting was "0." But recording worked ideally if it was to one notch above "mic." I used to set them at 45/46 respectively. Anyway, this handy adjustment / slider vanished a couple of weeks ago, and I was wondering where it went? Now I only see one slider, for "mic" and the performance of my microphone isn't as good, and I have to turn it up louder.

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[alsa-devel] Problems with snd-hda-intel ALC1200 on msi gt725-212us laptop Only one speaker works and headphones do not mute laptop sound
by Tony Rao 31 Aug '09

31 Aug '09

msi gt725-212us Codec: Realtek ALC1200 This laptop has 4.1 sound. Sound is only coming out of subwoofer. When I plug in headphones sounds still comes out of subwoofer, but headphones work perfectly. http://www.alsa-project.org/db/?f=e95a9401ebfa6cb0133b11fc1482ea115f766d5e Thanks, Tony R

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[alsa-devel] Error enabling dmix with ALSA
by ruchi sirauthiya 30 Aug '09

30 Aug '09

Hi, I am trying to enable a software mixer (using ALSA and dmix) on an OMAP3 based embedded target board using the twl4030 codec. When I try to play audio using the command, # aplay -D playback rec.wav I am getting the following error: ALSA lib pcm_direct.c:1098:(snd_pcm_direct_initialize_poll_fd) unable to open timer 'hw:CLASS=3,SCLASS=0,CARD=0,DEV=0,SUBDEV=0' ALSA lib pcm_dmix.c:943:(snd_pcm_dmix_open) unable to initialize poll_fd aplay: main:546: audio open error: No such file or directory The contents of my /etc/asound.conf is given below. pcm.playback { type dmix ipc_key 2048 ipc_key_add_uid true ipc_gid 1 slave { rate 44100 pcm hw:0 } } I have tested it on two kernel versions, 2.6.22 and 2.6.29, but it fails on both. Playback and recording works fine if -D option is not provided. I also tried configuring ALSA (dmix) with a different configuration file which is given in the following link: http://amarok.kde.org/wiki/Setting_up_Dmix_for_ALSA # aplay -D plug:ossmix /mnt/Audio1_1.wav ALSA lib pcm_direct.c:1098:(snd_pcm_direct_initialize_poll_fd) unable to open timer 'hw:CLASS=3,SCLASS=0,CARD=0,DEV=0,SUBDEV=0' ALSA lib pcm_dmix.c:943:(snd_pcm_dmix_open) unable to initialize poll_fd aplay: main:546: audio open error: No such file or directory But as you can see the same error turns up again. The topic discussed in the following link (a similar issue) suggests that it is unable to open a second timer. http://osdir.com/ml/linux.distributions.crux.general/2005-11/msg00134.html What could be the cause of this error? Is the above error related to my ALSA configuration (for dmix) or is it related to the sound hardware (or driver)? Do I need to patch the kernel or ALSA for using dmix? Please help me resolve this issue. Let me know if any more setup info is required. Thanks and Regards, Ruchi Sirauthiya. Some information I obtained from the proc filesystem is given below for your reference. ----------------------- For Linux kernel 2.6.29 ----------------------- [root@OMAP3 tests]# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.19 [root@OMAP3 tests]# cat /proc/asound/cards 0 [omap3evm ]: twl4030 - omap3evm omap3evm (twl4030) [root@OMAP3 tests]# omap-dss: Could not find exact pixel clock. Requested 25188 kHz, got 25187 kHz [root@OMAP3 tests]# cat /proc/asound/devices 2: : timer 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0] : control [root@OMAP3 tests]# cat /proc/asound/timers G0: system timer : 7812.500us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE [root@OMAP3 tests]# cat /proc/asound/pcm 00-00: TWL4030 twl4030-0 : : playback 1 : capture 1 [root@OMAP3 tests]# cat /proc/asound/devices 2: : timer 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0] : control [root@OMAP3 tests]# cat /proc/asound/pcm 00-00: TWL4030 twl4030-0 : : playback 1 : capture 1 [root@OMAP3 tests]# cat /proc/asound/oss/devices 3: [0- 0]: digital audio 4: [0- 0]: digital audio [root@OMAP3 tests]# cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.19 emulation code) Kernel: Linux OMAP3 2.6.29-rc3-omap1 #1 Fri Jul 24 14:45:37 IST 2009 armv7l Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: omap3evm (twl4030) Audio devices: 0: (DUPLEX) Synth devices: NOT ENABLED IN CONFIG Midi devices: NOT ENABLED IN CONFIG Timers: 31: system timer Mixers: NOT ENABLED IN CONFIG [root@OMAP3 tests]# aplay -D playback /mnt/Audio1_1.wav ALSA lib pcm_direct.c:1098:(snd_pcm_direct_initialize_poll_fd) unable to open timer 'hw:CLASS=3,SCLASS=0,CARD=0,DEV=0,SUBDEV=0' ALSA lib pcm_dmix.c:943:(snd_pcm_dmix_open) unable to initialize poll_fd aplay: main:546: audio open error: No such file or directory ----------------------- For Linux kernel 2.6.22 ----------------------- [root@OMAP3 tests]# cat /proc/asound/devices 2: : timer 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0] : control [root@OMAP3 tests]# cat /proc/asound/pcm 00-00: OMAP PCM : omap alsa pcm : playback 1 : capture 1 [root@OMAP3 tests]# cat /proc/asound/timers G0: system timer : 7812.500us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE [root@OMAP3 tests]# cat /proc/asound/ /proc/asound/TWL4030/ /proc/asound/devices /proc/asound/timers /proc/asound/card0/ /proc/asound/hwdep /proc/asound/version /proc/asound/cards /proc/asound/pcm [root@OMAP3 tests]# cat /proc/asound/cards 0 [TWL4030 ]: OMAP_ALSA - TWL4030 TWL4030 /proc/asound/oss was not found in the proc file system when booted with kernel 2.6.22.

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[alsa-devel] ASoC DAI on Atmel AT32AP7000
by Christos Nikolaou 30 Aug '09

30 Aug '09

Hi, Up to 2.6.28 the ASoC framework was working on AT32AP7000 Atmel platform. Now I try to recompile it on 2.6.31-rc8 and I get Advance Linux Sound Architecture Driver Version 1.0.20. No device for DAI atmel-ssc0 No device for DAI atmel-ssc1 No device for DAI atmel-ssc2 whereas the ssc devices are properly (as in 2.6.28) initialized ssc ssc.0: Atmel SSC device at 0xffe1c00 (irq 10) ssc ssc.1: Atmel SSC device at 0xffe2000 (irq 11) ssc ssc.2: Atmel SSC device at 0xffe2400 (irq 12) It looks that this message comes from snd_soc_register_dai in soc-core.c if (!dai->dev) printk(KERN_WARNING "No device for DAI %s\n", dai->name); it seems for some reason it thinks there is no device in dai->dev Unless I'm mistaken, that is being called from sound/soc/atmel/atmel_ssc_dai.c after some modifications done by broonie on these files according to http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=history;… . Is there anything that can be done to rectify this? Regards, Christos

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 30 Aug '09

30 Aug '09

The following changes since commit a0a499c5792b8656cd51e11d5e0db9fb21640f58: Eduardo Valentin (1): ASoC: OMAP: Use DMA operating mode of McBSP 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 Ben Dooks (4): ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec boards ASoC: tlv320aic3x: Change to use device model ASoC: tlv320aic3x: fixup board device changes ASoC: S3C24XX: Support for Simtec Hermes boards Kuninori Morimoto (2): ASoC: Add ak4642/ak4643 codec support ASoC: Add FSI-AK4642 sound support for SuperH Marek Vasut (5): power_supply: Make wm97xx-core pass platform_data to battery, ts power_supply: Convert WM97xx battery to use platform_data power_supply: Let wm97xx-battery use irq to detect charger state [ARM] pxa: Convert all platforms to pass platform_data to wm97xx battery ASoC: Pass correct platform data from pxa2xx-ac97 Mark Brown (1): ASoC: Add DAPM widget power decision debugfs files Roel Kluin (1): ASoC: free socdev if init_card() fails in wm9705_soc_probe() arch/arm/mach-pxa/mioa701.c | 10 +- arch/arm/mach-pxa/palmld.c | 19 +- arch/arm/mach-pxa/palmt5.c | 19 +- arch/arm/mach-pxa/palmte2.c | 20 +- arch/arm/mach-pxa/palmtx.c | 19 +- arch/arm/mach-pxa/palmz72.c | 20 +- arch/arm/mach-pxa/treo680.c | 1 - arch/arm/plat-s3c/include/plat/audio-simtec.h | 37 ++ drivers/input/touchscreen/wm97xx-core.c | 3 + drivers/power/wm97xx_battery.c | 56 ++- include/linux/wm97xx.h | 18 + include/linux/wm97xx_batt.h | 26 -- include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + sound/arm/pxa2xx-ac97.c | 4 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 502 ++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 20 + sound/soc/codecs/tlv320aic3x.c | 221 ++++++----- sound/soc/codecs/tlv320aic3x.h | 2 - sound/soc/codecs/wm9705.c | 2 +- sound/soc/davinci/davinci-evm.c | 12 +- sound/soc/omap/n810.c | 12 +- sound/soc/pxa/pxa2xx-ac97.c | 4 +- sound/soc/s3c24xx/Kconfig | 19 + sound/soc/s3c24xx/Makefile | 7 + sound/soc/s3c24xx/s3c24xx_simtec.c | 394 +++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 + sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 153 +++++++ sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 +++++++ sound/soc/s6000/s6105-ipcam.c | 12 +- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 107 +++++ sound/soc/soc-core.c | 8 + sound/soc/soc-dapm.c | 86 ++++ 37 files changed, 1784 insertions(+), 206 deletions(-) create mode 100644 arch/arm/plat-s3c/include/plat/audio-simtec.h delete mode 100644 include/linux/wm97xx_batt.h create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.c create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.h create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c create mode 100644 sound/soc/sh/fsi-ak4642.c

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[alsa-devel] [PATCH 4/4] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 29 Aug '09

29 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Enables tlv320aic3101 support on DM365 EVM. 2) Set tlv320aic3x i2c setup into DM365 EVM board file. This patch was tested on DM365 EVM rev c. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 3 +++ sound/soc/davinci/Kconfig | 9 +++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++++++-- 4 files changed, 40 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index fd2db78..cff68e1 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -187,6 +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 = { diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..b78b882 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -32,6 +32,15 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI +config SND_DM365_SOC_EVM + tristate "SoC Audio support for DaVinci DM365 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM365_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + + help + Say Y if you want to add support for SoC audio on TI + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..8238612 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,4 +13,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM365_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f3bb6f6..a859208 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -44,7 +44,8 @@ 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 @@ -179,6 +180,14 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &evm_dai, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +200,12 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +/* dm365 evm audio private data */ +static struct aic3x_setup_data dm365_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -198,13 +213,20 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* evm audio subsystem */ +/* dm6467 evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &dm6467_evm_aic3x_setup, }; +/* dm365 evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm365_evm_aic3x_setup, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +244,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; + }else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.6.0.4

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