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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.

2 1

[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

2 3

[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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[alsa-devel] [PATCH 3/4] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 29 Aug '09

29 Aug '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. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++ arch/arm/mach-davinci/dm365.c | 42 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 50 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..fd2db78 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), @@ -476,6 +478,8 @@ 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..cba09b0 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), @@ -806,6 +807,31 @@ 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 +933,20 @@ 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..fbcbed0 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of da850/da830 McASP0 register banks */ #define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 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

2 1

[alsa-devel] [PATCH v2 2/2] 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> This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 9 +++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++++++-- 3 files changed, 37 insertions(+), 2 deletions(-) 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

1 0

[alsa-devel] ASoC: at91sam9260-PCM1808/TAS5709-based board driver
by Pedro Sanchez 29 Aug '09

29 Aug '09

Hello, I have to develop a driver for a custom board based on the at91sam9260 with a PCM1808/TAS5709 chipset combo. I am addressing the PCM1808 first because it is a simpler device. Since I'm new to ASoC I'm looking for some help in this list on how to go about writing this driver. So far I have identified two files that I need to work on (other than the needed kconfig modifications). First is the codec driver at sound/soc/codecs/PCM1808.[c|h] which is based on the existent pcm3008 driver; I believe I'm fine with this. The second file is the machine driver at sound/soc/atmel/myBoard.c. I'm using sam9g20_wm8731.c as reference, on which I believe I found a typo which I reported in a separated e-mail. So my first question is, is this two-files step the right approach? am I in the right track? I'm using the 2.6.30 kernel. Other questions will come and I hope I will get some cycles from you to help me direct my work. Of course I'll be glad to submit my driver work for inclusion in the kernel if it proves useful. Regards, -- Pedro

5 16

[alsa-devel] [PATCH 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 28 Aug '09

28 Aug '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. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++ arch/arm/mach-davinci/dm365.c | 42 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 50 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..fd2db78 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), @@ -476,6 +478,8 @@ 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..cba09b0 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), @@ -806,6 +807,31 @@ 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 +933,20 @@ 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..fbcbed0 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of da850/da830 McASP0 register banks */ #define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 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

2 1

[alsa-devel] [PATCH 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 28 Aug '09

28 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

2 1

[alsa-devel] [PATCH] Davinci: Enable ASoC support for DM365 EVM in defconfig
by miguel.aguilar＠ridgerun.com 28 Aug '09

28 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/configs/davinci_all_defconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/arch/arm/configs/davinci_all_defconfig b/arch/arm/configs/davinci_all_defconfig index ec63c15..cbe18b4 100644 --- a/arch/arm/configs/davinci_all_defconfig +++ b/arch/arm/configs/davinci_all_defconfig @@ -1099,6 +1099,7 @@ CONFIG_SND_DAVINCI_SOC_I2S=m CONFIG_SND_DAVINCI_SOC_MCASP=m CONFIG_SND_DAVINCI_SOC_EVM=m CONFIG_SND_DM6467_SOC_EVM=m +CONFIG_SND_DM365_SOC_EVM=m # CONFIG_SND_DAVINCI_SOC_SFFSDR is not set CONFIG_SND_SOC_I2C_AND_SPI=m # CONFIG_SND_SOC_ALL_CODECS is not set -- 1.6.0.4

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 28 Aug '09

28 Aug '09

The following changes since commit d09a2afc9359407114b7062519101f1ee2d05388: Jarkko Nikula (1): ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop 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 Candelaria Villareal, Jorge (1): ASoC: SDP3430: Fix TWL GPIO6 pin mux request Mark Brown (4): ASoC: Add S3C24xx dependencies for Simtec machines ASoC: Make platform data optional for TLV320AIC3x ASoC: Fix s3c-i2s-v2 build ASoC: Don't reconfigure WM8350 FLL if not needed Shine Liu (1): ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time sound/soc/codecs/tlv320aic3x.c | 13 ++++++------- sound/soc/codecs/wm8350.c | 9 +++++++++ sound/soc/omap/sdp3430.c | 15 +++++++++++---- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 11 +++++++++++ sound/soc/s3c24xx/s3c2443-ac97.c | 10 ++++++++++ sound/soc/s3c24xx/s3c24xx-i2s.c | 5 +++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 1 - 8 files changed, 54 insertions(+), 14 deletions(-)

2 5

[alsa-devel] Audio capture not working with AIC23/McBSP1 on OMAP3517
by Aggarwal, Anuj 28 Aug '09

28 Aug '09

Hi, I am trying to capture audio through LineIn on my OMAP3517 EVM, using AIC23 codec and McBSP1, and getting the following error: arecord: pcm_read:1529: read error: Input/output error Playback is working fine using the same combination. I have tried using another instance of AIC23 codec which uses McBSP2 and both playback & capture works fine there. Has anyone faced this issue before? Any suggestions? Regards, Anuj Aggarwal

4 11

[alsa-devel] Skype beta better audio, but crashing after launch...
by jack craig 28 Aug '09

28 Aug '09

I thought i'd start a new thread given the progress the beta provides. now my test sound makes a noise and my test call lanuches and stops when i tell it to, BUT! it crashes with the below. pulseaudio[3400]: resampler.c: Using float32le as working format. pulseaudio[3400]: resampler.c: Choosing speex quality setting 3. pulseaudio[3400]: source-output.c: Created output 0 "Input" on alsa_input.pci_8086_24c5_sound_card_0 with sample spec s16le 1ch 16000Hz and channel map mono pulseaudio[3400]: protocol-native.c: Final latency 199.89 ms = 99.94 ms + 99.95 ms pulseaudio[3400]: source.c: Assertion 'pa_cvolume_compatible(volume, &s->sample_spec)' failed at pulsecore/source.c:666, function pa_source_set_volume(). Aborting. any suggestions on fixing this ? tia, jackc... -- jack craig jackc(a)LinuxLightHouse.com 831-684-1375 (Office) 831-596-6924 (cell) IM: jackcraigaptos (AIM) _________________________________ This email has been ClamScanned ! www.LinuxLightHouse.com

1 0

[alsa-devel] Any patches missing for 2.6.32?
by Takashi Iwai 28 Aug '09

28 Aug '09

Hi, now 2.6.31-rc8 came up, and we are pretty close to the next merge window. If you have any patches that should go into 2.6.32, please speak up, post them now. Otherwise it'll be postponed after 2.6.32. thanks, Takashi

1 0

[alsa-devel] SNDRV_PCM_TRIGGER_STOP and audio still queued in the driver
by Jon Smirl 28 Aug '09

28 Aug '09

In the mpc5200 driver I'm receiving SNDRV_PCM_TRIGGER_STOP with audio still queued. I noticed this in an app that uses a 2MB ALSA buffer. It is filling the 2MB buffer which starts playing. After the buffer is filled I receive the STOP. This STOP comes before the hardware has a chance to play all of the music. There is still about twenty seconds of music in the driver buffer waiting to be played. How is this supposed to work? -- Jon Smirl jonsmirl(a)gmail.com

2 11

[alsa-devel] [PATCH 1/1] sound: usb-audio: support for Digidesign Mbox 2
by Damien Zammit 27 Aug '09

27 Aug '09

Provides magic boot sequence which activates the device, must loop until device is ready, (returns status bytes). Provides extra magic to set the device into 24 bit mode. -Playback works @ 24-bit -MIDI I/O works -Capture is untested, but PCM stream is active -S/PDIF (RCA) I/O is untested, but lights activate on the device S/PDIF mode works by setting the required flag in usbaudio.c. I think the device can't do both simultaneously, since different altsettings are required to set it. Remaining issues: 1) Double check endianess of 24 bit capture. 2) Check duplex stability 3) Tell ALSA to correctly detect S24_3BE on the device. Currently, speaker-test assumes S16_LE and won't allow S24_3BE mode, but jackd 0.116-2 detects and plays S24_3BE ? 4) Create a kernel module switch to select spdif/analogue I/O Playback sample format : S24_3BE MIDI format : MIDIMAN protocol Signed-off-by: Damien Zammit <damien.zammit(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 43d7d88..3060941 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -943,7 +943,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream { struct snd_usb_substream *subs; snd_pcm_uframes_t hwptr_done; - + subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; @@ -2441,6 +2441,12 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor fp->altsetting, sample_width, sample_bytes); break; } + + /* Digidesign Mbox 2 workaround: + * supports S24_3BE BIG ENDIAN */ + if (chip->usb_id == USB_ID(0x0dba, 0x3000)) + pcm_format = SNDRV_PCM_FORMAT_S24_3BE; + break; case USB_AUDIO_FORMAT_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; @@ -2550,13 +2556,20 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * switch (chip->usb_id) { case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - if (device_setup[chip->index] == 0x00 && + if (device_setup[chip->index] == 0x00 && fp->altsetting == 6) pcm_format = SNDRV_PCM_FORMAT_S16_BE; else pcm_format = SNDRV_PCM_FORMAT_S16_LE; break; - default: + + /* Digidesign Mbox 2 supports SPDIF 24 bit (havent tested endianess) */ + case USB_ID(0x0dba, 0x3000): + pcm_format = SNDRV_PCM_FORMAT_S24_3BE; + break; + + default: + snd_printdd("Format III detected, setting default S16_LE\n"); pcm_format = SNDRV_PCM_FORMAT_S16_LE; } } else { @@ -2655,6 +2668,8 @@ static unsigned char parse_datainterval(struct snd_usb_audio *chip, static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -2700,13 +2715,19 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x0763, 0x2003) && + if (chip->usb_id == USB_ID(0x0763, 0x2003) && audiophile_skip_setting_quirk(chip, iface_no, altno)) continue; + /* Digidesign Mbox 2: skip altsets incompatible with device_setup + */ + if (chip->usb_id == USB_ID(0x0dba, 0x3000) && + mbox2_skip_setting_quirk(chip, iface_no, altno)) + continue; + /* get audio formats */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); if (!fmt) { @@ -2995,7 +3016,7 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. */ static int create_uaxx_quirk(struct snd_usb_audio *chip, @@ -3264,6 +3285,97 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac return 0; } + +#define MBOX2_FIRMWARE_SIZE 646 +#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ +#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ + +static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + u8 enablemagic[3]; + u8 bootresponse; + int fwsize; + + if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { + snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); + + /* From USB Snoop, + * + * SetupPacket = RT RQ VHVL INDX SIZE + * RT = 0xRT Request Type + * RQ = 0xRQ Request + * VHVL = 0xVLVH Value in reverse byte order + * INDX = 0xDXIN Index in reverse byte order + * SIZE = 0xZESI Size in reverse byte order + * + * Magic boot code setup packet: c0 85 01 00 00 00 12 00 + * RQ RT VLVH DXIN ZESI + * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT + * for snd_usb_ctl_msg() + */ + +mbox2_reboot: + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); + + if (bootresponse == MBOX2_BOOT_LOADING) { + snd_printdd("device not ready, resending boot sequence...\n"); + goto mbox2_reboot; + } + + if (bootresponse == MBOX2_BOOT_READY) { + snd_printdd("device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); + err = usb_reset_configuration(dev); + if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); + snd_printdd("mbox2_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) + * but now we will try sending capture/playback enable magic + * + * 80 bb 00 = 24bit mode - S24_3BE + * 44 ac 00 = 16bit mode? + * + */ + //enablemagic[0]=0x44; + //enablemagic[1]=0xac; + //enablemagic[2]=0x00; + + /* We want 24 bit mode for now */ + enablemagic[0]=0x80; + enablemagic[1]=0xbb; + enablemagic[2]=0x00; + + /* Send the magic! */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); + + return 0; /* Succesful boot */ + } + + snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); + return -ENODEV; + } + snd_printdd("Invalid firmware size: %d\n",fwsize); + return -ENODEV; +} + static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) { u8 buf = 1; @@ -3366,10 +3478,79 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5) return 1; /* skip this altsetting */ - } + } return 0; /* keep this altsetting */ } +/* Digidesign Mbox 2 I/O selector + * Uncomment this line to enable SPDIF */ +//#define MBOX2_SPDIF_IO + +#define MBOX2_FIRMWARE_INTERFACE 0 +#define MBOX2_CONTROL_INTERFACE 1 +#define MBOX2_MIDI_INTERFACE 6 + +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* Reset ifaces 2-5 to 0 altsetting. */ + usb_set_interface(chip->dev, iface, 0); + +#ifdef MBOX2_SPDIF_IO + /* Enable SPDIF altsettings */ + switch (iface) { + case MBOX2_FIRMWARE_INTERFACE: + case MBOX2_CONTROL_INTERFACE: + return 0; /*keep*/ + break; + case 2: + if (altno == 2) return 0; else return 1; + break; + case 3: + return 1; + break; + case 4: + if (altno == 2) return 0; else return 1; + break; + case 5: + return 1; + break; + case MBOX2_MIDI_INTERFACE: + return 0; /*keep*/ + break; + default: + snd_printdd("Invalid interface for Mbox 2: %d\n",iface); + return 1; /* skip altsetting */ + } +#else + /* Use analogue I/O */ + switch (iface) { + case MBOX2_FIRMWARE_INTERFACE: + case MBOX2_CONTROL_INTERFACE: + return 0; + break; + case 2: + if (altno == 3) return 0; else return 1; + break; + case 3: + if (altno == 6) return 0; else return 1; + break; + case 4: + if (altno == 3) return 0; else return 1; + break; + case 5: + if (altno == 3) return 0; else return 1; + break; + case MBOX2_MIDI_INTERFACE: + return 0; + break; + default: + snd_printdd("Invalid interface for Mbox 2: %d\n",iface); + return 1; /* skip altsetting */ + } +#endif +} + /* * audio-interface quirks * @@ -3392,6 +3573,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, + [QUIRK_MIDI_MBOX2] = snd_usb_create_midi_interface, [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, @@ -3425,7 +3607,7 @@ static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_ { struct snd_usb_audio *chip = entry->private_data; if (!chip->shutdown) - snd_iprintf(buffer, "%04x:%04x\n", + snd_iprintf(buffer, "%04x:%04x\n", USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); } @@ -3616,6 +3798,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* Digidesign Mbox 2 needs magic boot sequence */ + if (id == USB_ID(0x0dba, 0x3000)) { + if (snd_usb_mbox2_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f789..21aab48 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -157,6 +157,7 @@ enum quirk_type { QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, QUIRK_MIDI_US122L, + QUIRK_MIDI_MBOX2, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19c..bc11fe4 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -423,7 +423,7 @@ static void snd_usbmidi_maudio_broken_running_status_input( u8 cin = buffer[i] & 0x0f; struct usbmidi_in_port *port = &ep->ports[cable]; int length; - + length = snd_usbmidi_cin_length[cin]; if (cin == 0xf && buffer[i + 1] >= 0xf8) ; /* realtime msg: no running status change */ @@ -617,13 +617,13 @@ static struct usb_protocol_ops snd_usbmidi_standard_ops = { static struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; @@ -1577,7 +1577,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) - return 0; + return 0; intf = umidi->iface; if (!intf || intf->num_altsetting < 1) @@ -1615,7 +1615,7 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint_info* endpoints) { int err, i; - + err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { if (endpoints[i].out_ep) @@ -1876,6 +1876,18 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_MBOX2: + /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ + umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; + /* + * We have to make sure that the USB core looks + * again at interface 6 by calling usb_set_interface() on it. + */ + usb_set_interface(umidi->chip->dev, 6, 0); + memcpy(&endpoints[0], quirk->data, + sizeof(struct snd_usb_midi_endpoint_info)); + err = 0; + break; case QUIRK_MIDI_FASTLANE: umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; /* diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201e..e961f5e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2082,6 +2082,60 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* DIGIDESIGN MBOX 2 */ +{ + /* Damien Zammit <damien.zammit(a)gmail.com> */ + + USB_DEVICE(0x0dba, 0x3000), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 2", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 5, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 6, + .type = QUIRK_MIDI_MBOX2, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_ep = 0x02, + .out_cables = 0x0001, + .in_ep = 0x81, + .in_interval = 0x01, + .in_cables = 0x0001 + } + + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- 1.5.4.3

2 3

[alsa-devel] ALSA default buffer sizes (recent now being subscribed with correct address... hopefully)
by Manuel Jander 26 Aug '09

26 Aug '09

Hi there, I'm somewhat upset because audacious does not work with my AU8830. The problem is that audacious never calls any function that would give any clue about the desired buffer size. So my AU8830 ends up with 32 periods, 1 byte each. Great ! I'm not going to blame the author of that driver :P ... but: The question is, who is the entity deciding those default values ? Other soundcards seem to get reasonable defaults, but the AU8830 does not. So I would like to fix that, but I want to do that at the right place. Best Regards, and many thanks for any hint.

3 4

[alsa-devel] [PATCH] vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
by Roel Kluin 26 Aug '09

26 Aug '09

Since !LI_CCFG_* evaluates to 0, this did not change anything to cfgval and ctlval. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- // vi sound/oss/vwsnd.c +392 # define LI_CCTL_RESET 0x80000000 ... # define LI_CCTL_DMA_ENABLE 0x08000000 ... # define LI_CCFG_LOCK 0x00008000 I am not sure this is what was intended, does anyone have the specs? diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 187f727..6713110 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -628,7 +628,7 @@ static void li_setup_dma(dma_chan_t *chan, ASSERT(!(buffer_paddr & 0xFF)); chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8); - chan->cfgval = (!LI_CCFG_LOCK | + chan->cfgval = ((chan->cfgval & ~LI_CCFG_LOCK) | SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) | desc->direction | mode | @@ -638,9 +638,9 @@ static void li_setup_dma(dma_chan_t *chan, tmask = 13 - fragshift; /* See Lithium DMA Notes above. */ ASSERT(size >= 2 && size <= 7); ASSERT(tmask >= 1 && tmask <= 7); - chan->ctlval = (!LI_CCTL_RESET | + chan->ctlval = ((chan->ctlval & ~LI_CCTL_RESET) | SHIFT_FIELD(size, LI_CCTL_SIZE) | - !LI_CCTL_DMA_ENABLE | + (chan->ctlval & ~LI_CCTL_DMA_ENABLE) | SHIFT_FIELD(tmask, LI_CCTL_TMASK) | SHIFT_FIELD(0, LI_CCTL_TPTR));

2 1

[alsa-devel] [PATCH 1/4] SOUND: atiixp, use uninitialized_var(chip)
by Jiri Slaby 26 Aug '09

26 Aug '09

To avoid wrong compiler warnings, use unitialized_var(chip) in snd_atiixp_probe's from atiixp and atiixp_modem drivers. 'chip' is unused when unset due to retval being nonzero. Signed-off-by: Jiri Slaby <jirislaby(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752df..436ffea 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1648,7 +1648,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp *chip; + struct atiixp *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147b..b6191f2 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1285,7 +1285,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp_modem *chip; + struct atiixp_modem *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); -- 1.6.3.3

2 4

[alsa-devel] alsa random crash
by some guy 26 Aug '09

26 Aug '09

Problem: Alsa randomly crashes after ~10 hours of use; this varies from 2 to 20 hours. Symptoms: Sound stops while media is playing (in web browser, or mplayer), or convulses sporatically. The issue is not always reproducible (i.e I can reproduce the result when I want), but it is perisistant. Info: http://www.alsa-project.org/db/?f=fe434884ea7a16b1ae7c06cfd282832971f41fd6 lspci erhttp://pastebin.ca/1542518 kernel version is gentoo-2.6.28-r5 I really appreciate it. -anonymous alsa'er. _________________________________________________________________ More storage. Better anti-spam and antivirus protection. Hotmail makes it simple. http://go.microsoft.com/?linkid=9671357

1 0

[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..4a3f62d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

2 1

Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 25 Aug '09

25 Aug '09

2009/8/25 Takashi Iwai <tiwai(a)suse.de> > BTW, did you check, as I suggested, whether acer-aspire-6530g with > changing the channel mode from 2ch to 6ch works for 7730g? > I guess we can merge acer-aspire-* variants to one or two models. > Can you describe in details what I need to do, please?

1 0

Re: [alsa-devel] Acer Aspire 7730G support
by Takashi Iwai 25 Aug '09

25 Aug '09

At Mon, 24 Aug 2009 19:57:58 +0400, Denis Kup wrote: > > 2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > > At Mon, 24 Aug 2009 17:10:10 +0400, > > Denis Kup wrote: > > > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > > What about my patch? > > > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > > whether it still work and repost the patch. There have been lots of > > changes in HD-audio driver code, so an old patch might not work any > > more. > > > > > > thanks, > > > > Takashi > > > I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it > should. Try rather with the snapshot tarball in ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… The version you tried isn't the latest one. > It's diff between patched version( patch_realtek_new.c ) and unpatched: Make a patch with diff -u option (and better with -p, too). thanks, Takashi

3 5

[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..7ad99d2 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x3); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x2); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

3 4

[alsa-devel] [PATCH] ASoC: s3c24xx platform: Fix s3c2410_dma_started called at wrong time
by Shine Liu 25 Aug '09

25 Aug '09

s3c24xx dma has the auto reload feature, when the the trnasfer is done, CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So the transmission is repeated. IRQ is issued while auto reload occurs. We change the DISRC and DCON[19:0] in the ISR, but at this time, the auto reload has been performed already. The first block is being re-transmitted by the DMA. So we need rewrite the DISRC and DCON[19:0] for the next block immediatly after the this block has been started to be transported. The function s3c2410_dma_started() is for this perpose, which is called in the form of "s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger(). But it is not correct. DMA transmission won't start until DMA REQ signal arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in s3c24xx_i2s_trigger(). In the current framework, s3c24xx_pcm_trigger() is always called before s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in this function. However, s3c2410_dma_started() is dma related, to call this function we should provide the channel number, which is given by substream->runtime->private_data->params->channel. The private_data points to a struct s3c24xx_runtime_data object, which is define in s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c To move the struct s3c24xx_runtime_data defination to s3c24xx-pcm.h is one solution. If not moved, the patch shold be like: Signed-off-by: Shine Liu <shinel(a)foxmail.com> ---------------------------------------------------------------- --- a/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-20 16:06:14.000000000 +0800 @@ -28,4 +28,6 @@ extern struct snd_soc_platform s3c24xx_soc_platform; extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream); + #endif --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-20 15:56:46.000000000 +0800 @@ -66,6 +66,28 @@ struct s3c24xx_pcm_dma_params *params; }; +/* s3c24xx_pcm_dma_started + * + * To load the second dma buffer immediately after the + * first dma buffer has been started transferring, so that + * in the dma autoreload situation, the first data block + * won't be re-transferred after the first irq issued. + * + * This function is called in the s3c24xx-i2s-trigger() of + * s3c24xx-i2s.c, just after the dma transfer is really started. + * + * It's ugly here. Need a elegant way to fix this bug. + * FIXME + * + */ + +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); + +} + /* s3c24xx_pcm_enqueue * * place a dma buffer onto the queue for the dma system @@ -255,7 +277,6 @@ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-20 16:08:06.000000000 +0800 @@ -296,6 +296,9 @@ s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + /* DMA transfer is really started, equeue the next buffer */ + s3c24xx_pcm_dma_started(substream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

2 24

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Aug '09

25 Aug '09

The following changes since commit c6ea2af76ab79b82a0bfed313e4c2b68b10c2bc1: Takashi Iwai (1): ASoC: Remove unneeded inclusion of linux/regulator/consumer.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 Janusz Krzysztofik (1): ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Jarkko Nikula (3): OMAP: McBSP: Use textual values in DMA operating mode sysfs files ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop Mark Brown (4): ASoC: Add input and output AIF widgets ASoC: WM8993 digital mixing support ASoC: Select core DMA when building for S3C64xx Merge branch 'topic/digital-mixing' into for-2.6.32 arch/arm/plat-omap/dma.c | 10 +++ arch/arm/plat-omap/include/mach/mcbsp.h | 2 - arch/arm/plat-omap/mcbsp.c | 132 ++++++++++++------------------ include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/wm8993.c | 88 ++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 13 +--- sound/soc/s3c24xx/Kconfig | 1 + sound/soc/soc-dapm.c | 58 ++++++++++---- 8 files changed, 199 insertions(+), 113 deletions(-)

2 1

[alsa-devel] [PATCH 0/3] Few patches to recent OMAP McBSP and ASoC changes
by Jarkko Nikula 25 Aug '09

25 Aug '09

Hi I would like to add these tree patches to recent set: [PATCHv5 00/20] OMAP ASoC changes in DMA utilization http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020619.html These are actually independent patches but probably easier to handle as with the set above since some of the patches are touching both ASoC and OMAP subsystems. First patch is the most important since it changes the user space API for McBSP DMA operating mode selection. It would be more difficult to change it later if software starts to use current numeric value based operating mode selection in sysfs files. -- Jarkko

3 12

[alsa-devel] multiple devices or one device w/ multiple subdevices?
by pl bossart 25 Aug '09

25 Aug '09

Hi, I have a piece of custom hardware that provides 7 independent PCM outputs. This is not intended for multichannel playback, the content on each output can be completely different; playback will not be synchronized and will start separately. The only restriction is that all outputs rely on the same baserate and share the same bitclock, i.e. all the outputs work at 48kHz. _From the mailling list archives, subdevices seem to be relevant when doing hw mixing or when lots of streams are handled. None of this applies to my case. Also it looks like PulseAudio knows nothing about subdevices in its detection loop, meaning the user would need to manually add some load-module commands in the PulseAudio configuration files. And last but not least, I didn't see very many lines of code (and no documentation) that showed how subdevices are registered/handled, etc. My conclusion is that multiple devices would seem like a much better option. But I wanted to check with the community that I am on the right track. Thanks for your feedback! - P

3 2

[alsa-devel] [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
by Janusz Krzysztofik 25 Aug '09

25 Aug '09

Implement DMA channel self linking on OMAP1510 using AUTO_INIT and REPEAT flags of the DMA CCR register. Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Note: In its current form, enable_lnk() seems not very consistent (ie. result of a conditional assignment is then overwritten unconditionally). As rather different from OMAP1510 that I am familiar with, I am not quite sure how support for other OMAP models should really look like. So I decided to leave it as is and put my bit in a separate block, adding an additional FIXME comment. --- linux-2.6.31-rc5/arch/arm/plat-omap/dma.c.orig 2009-08-07 13:38:02.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/dma.c 2009-08-10 09:01:41.000000000 +0200 @@ -653,8 +653,21 @@ static inline void enable_lnk(int lch) { u32 l; + if (omap_dma_in_1510_mode()) { + /* only self linking is supported */ + if (dma_chan[lch].next_lch == lch) { + /* set AUTO_INIT | REPEAT bits in CCR */ + l = dma_read(CCR(lch)); + l |= 3 << 8; + dma_write(l, CCR(lch)); + } + return; + } + l = dma_read(CLNK_CTRL(lch)); + /* FIXME: + * this conditional assignment is then overwritten unconditionally */ if (cpu_class_is_omap1()) l &= ~(1 << 14); @@ -675,13 +688,21 @@ static inline void disable_lnk(int lch) { u32 l; - l = dma_read(CLNK_CTRL(lch)); + if (omap_dma_in_1510_mode()) + l = dma_read(CCR(lch)); + else + l = dma_read(CLNK_CTRL(lch)); /* Disable interrupts */ if (cpu_class_is_omap1()) { dma_write(0, CICR(lch)); - /* Set the STOP_LNK bit */ - l |= 1 << 14; + if (omap_dma_in_1510_mode()) { + /* Unset AUTO_INIT & REPEAT bits */ + l &= ~(3 << 8); + } else { + /* Set the STOP_LNK bit */ + l |= 1 << 14; + } } if (cpu_class_is_omap2()) { @@ -690,7 +711,10 @@ static inline void disable_lnk(int lch) l &= ~(1 << 15); } - dma_write(l, CLNK_CTRL(lch)); + if (omap_dma_in_1510_mode()) + dma_write(l, CCR(lch)); + else + dma_write(l, CLNK_CTRL(lch)); dma_chan[lch].flags &= ~OMAP_DMA_ACTIVE; } @@ -928,7 +952,7 @@ void omap_start_dma(int lch) { u32 l; - if (!omap_dma_in_1510_mode() && dma_chan[lch].next_lch != -1) { + if (dma_chan[lch].next_lch != -1) { int next_lch, cur_lch; char dma_chan_link_map[OMAP_DMA4_LOGICAL_DMA_CH_COUNT]; @@ -979,7 +1003,7 @@ void omap_stop_dma(int lch) { u32 l; - if (!omap_dma_in_1510_mode() && dma_chan[lch].next_lch != -1) { + if (dma_chan[lch].next_lch != -1) { int next_lch, cur_lch = lch; char dma_chan_link_map[OMAP_DMA4_LOGICAL_DMA_CH_COUNT]; @@ -1130,7 +1154,7 @@ int omap_dma_running(void) */ void omap_dma_link_lch(int lch_head, int lch_queue) { - if (omap_dma_in_1510_mode()) { + if (omap_dma_in_1510_mode() && (lch_head != lch_queue)) { printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1152,7 +1176,7 @@ EXPORT_SYMBOL(omap_dma_link_lch); */ void omap_dma_unlink_lch(int lch_head, int lch_queue) { - if (omap_dma_in_1510_mode()) { + if (omap_dma_in_1510_mode() && (lch_head != lch_queue)) { printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1822,7 +1846,7 @@ static int omap1_dma_handle_ch(int ch) if (unlikely(csr & OMAP_DMA_DROP_IRQ)) printk(KERN_WARNING "DMA synchronization event drop occurred " "with device %d\n", dma_chan[ch].dev_id); - if (likely(csr & OMAP_DMA_BLOCK_IRQ)) + if (likely(csr & OMAP_DMA_BLOCK_IRQ) && (dma_chan[ch].next_lch != ch)) dma_chan[ch].flags &= ~OMAP_DMA_ACTIVE; if (likely(dma_chan[ch].callback != NULL)) dma_chan[ch].callback(ch, csr, dma_chan[ch].data);

4 10

Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > At Mon, 24 Aug 2009 17:10:10 +0400, > Denis Kup wrote: > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > What about my patch? > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > whether it still work and repost the patch. There have been lots of > changes in HD-audio driver code, so an old patch might not work any > more. > > > thanks, > > Takashi I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it should. It's diff between patched version( patch_realtek_new.c ) and unpatched: dener@linux-wpg9:~/Documents/1> diff ./patch_realtek_new.c ./patch_realtek_old.c 223d222 < ALC888_ACER_ASPIRE_7730G, 8341,8366d8339 < static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { < {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, < {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, < {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, < { } /* end */ < }; < < static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) < { < unsigned int present; < < present = snd_hda_codec_read(codec, 0x15, 0, < AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; < snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, < HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); < snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, < HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); < } < < static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, < unsigned int res) < { < if ((res >> 26) == ALC880_HP_EVENT) < alc888_acer_aspire_7730G_front_automute(codec); < } < 8558d8530 < [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", 8605d8576 < SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), 9094,9107d9064 < }, < [ALC888_ACER_ASPIRE_7730G] = { < .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, < .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, < .num_dacs = ARRAY_SIZE(alc883_dac_nids), < .dac_nids = alc883_dac_nids, < .dig_out_nid = ALC883_DIGOUT_NID, < .dig_in_nid = ALC883_DIGIN_NID, < .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), < .channel_mode = alc883_3ST_6ch_modes, < .need_dac_fix = 1, < .input_mux = &alc883_capture_source, < .unsol_event = alc888_acer_aspire_7730G_unsol_event, < .init_hook = alc888_acer_aspire_7730G_front_automute,

1 0

[alsa-devel] [PATCHv4 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 24 Aug '09

24 Aug '09

Hello again guys, Here is version 4 of these changes. No real changes from previous versions. The only thing is that now I've rebased on top of Takashi's tree: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=summary as there are pending things there which Jarkko has pointed on previous series. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 78 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 37

Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/6/24 Denis Kup <dener.kup(a)gmail.com>: > 2009/6/24 Takashi Iwai <tiwai(a)suse.de>: >> At Wed, 24 Jun 2009 16:09:58 +0400, >> Denis Kup wrote: >>> >>> Hey, when i first installed alsa to my laptop, there no sound from >>> subwoofer, jacks were map very strange (2 surrounds for ex.). I try >>> all models from the list, but all don't support my subwoofer. After a >>> weak of hard work i made a patch that enables ACER_ASPIRE_7730G model >>> for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal >>> frontal speakers and internal subwoofer.You can see acer.com for more >>> info about hardware and jacks. >>> >>> It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: >> >> Can you try Emilio's patch recently posted (and pass >> model=acer-aspire-6530g) with the very latest alsa-driver-snapshot? >> This essentially includes your changes. It's in: >> From: Emilio López <buhitoescolar(a)gmail.com> >> Subject: Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G >> Date: Tue, 23 Jun 2009 22:32:45 -0300 >> Message-ID: <d54969c50906231832w48ce7315uc5cf8b3f81534c76(a)mail.gmail.com> >> >>> After changing file i install driver, and it automatically detect its >>> model (maybe there are old settings from previous install, so i dont >>> 100% sure that it was detected). With this settings subwoofer works >>> perfect, internal stereo and subwoofer mute when headphone are plugged >>> all works, except 1 thing: subwoofer uses center channel instead LFE. >>> To fix it i use such asound.conf file: >> ... >>> Is there any chance to swap Center and LFE in patch_realtek.c? >> >> Unlikely as ALC888* doesn't support the channel-swap feature (although >> some other ALC codecs do). >> >> >> thanks, >> >> Takashi >> > > You are talking about that > http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html > ?? Its right link to drivers http://www.alsa-project.org/snapshot/ ?? > And there is easier way to apply patches than manually adding strings > to files? > I manage with installing. I use alsa-driver-snapshot.tar.gz http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ and this Amilo changes http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html . But this patch has smaller functionality against i have. There is advantages and disadvantages of both patches: My patch: + 1) when black jack plugged, internal speakers and subwoofer mute + 2) there is 3 maps for jacks: (black/pink/blue) 2ch: front/ext mic/ (something i dont know, it should be line in), 4ch: front/ ext mic/ side 6ch: front/ CLFE / side; (its changes in channel mode option in mixer) + 3) internal mic works + 4) there are separate boost mic channels for internal and external mics - 5) internal subwoofer controls with Center channel in mixer ? 6) i dont know how line in works (i will check it when reinstall to my patch) ? 7) i dont know how spdif works, because i have not any spdif speakers in house Amilo patch: - 1) when black jack pluged, only internal speaker mutes, subwoofer continue to work (it has no meaning when i listen to music in headphones) - 2) there is only 1 map for jacks: front/ext mic/ (something i dont know, it should be line in) - 3) there no side and center channels - 3) internal mic dont work - 4) there no line in in input source menu in mixer + 5) Subwoofer controls with LFE channel in mixer ? 7) i dont know how spdif works, because i have not any spdif speakers in house

3 4

[alsa-devel] [RFC] [PATCH 3/3] ASoC: OMAP: Enhance OMAP1510 DMA progress software counter
by Janusz Krzysztofik 24 Aug '09

24 Aug '09

Enhance period index accuracy, particularly just before buffer rewind, by making use of DMA interrupt status flags in addition to simply counting interrupts This patch applies on top of patch 2 from this series: [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c.orig 2009-08-10 08:31:08.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c 2009-08-10 08:55:09.000000000 +0200 @@ -68,13 +68,28 @@ static void omap_pcm_dma_irq(int ch, u16 * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); - if (prtd->period_index >= 0) { - if (++prtd->period_index == runtime->periods) { + if (stat & OMAP_DMA_LAST_IRQ) { + /* last period of a buffer has been started */ + if (prtd->period_index == (runtime->periods - 1)) { + /* we are in sync, do nothing */ + spin_unlock_irqrestore(&prtd->lock, flags); + return; + } else if (prtd->period_index >= 0) { + /* possible IRQ loss, update the pointer */ + prtd->period_index = runtime->periods - 1; + } + } else if (prtd->period_index >= 0) { + /* playback in progress - increment the counter, + * check for end of buffer */ + if ((++prtd->period_index >= runtime->periods) || + (stat & OMAP_DMA_BLOCK_IRQ)) { + /* end of buffer reached, loop back */ prtd->period_index = 0; } } spin_unlock_irqrestore(&prtd->lock, flags); - } + } else if (stat & OMAP_DMA_LAST_IRQ) + return; snd_pcm_period_elapsed(substream); } @@ -175,7 +190,8 @@ static int omap_pcm_prepare(struct snd_p dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | + (cpu_is_omap1510() ? OMAP_DMA_LAST_IRQ : 0)); return 0; }

4 12

[alsa-devel] [PATCH 1/2] ASoC: Add input and output AIF widgets
by Mark Brown 24 Aug '09

24 Aug '09

Currently DAPM interfaces with the audio streams to and from the processor at the DAC and ADC widgets. As the digital capabilities of parts increases this is becoming a less and less able to meet the needs of parts. To meet the needs of these devices create new widgets interfacing with the TDM bus but not integrated into any other functionality. Audio can then be routed to and from these widgets using existing routing widgets. A slot number is provided in the definition but this is currently not used yet. This is intended to support devices which can use more than one TDM slot on a single interface. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 8 ++++++ sound/soc/soc-dapm.c | 58 ++++++++++++++++++++++++++++++++------------- 2 files changed, 49 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 35814ce..3388405 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -137,6 +137,12 @@ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} @@ -312,6 +318,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8846131..a225e5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,17 +55,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_micbias] = 2, - [snd_soc_dapm_mic] = 3, - [snd_soc_dapm_mux] = 4, - [snd_soc_dapm_value_mux] = 4, - [snd_soc_dapm_dac] = 5, - [snd_soc_dapm_mixer] = 6, - [snd_soc_dapm_mixer_named_ctl] = 6, - [snd_soc_dapm_pga] = 7, - [snd_soc_dapm_adc] = 8, - [snd_soc_dapm_hp] = 9, - [snd_soc_dapm_spk] = 9, - [snd_soc_dapm_post] = 10, + [snd_soc_dapm_aif_in] = 3, + [snd_soc_dapm_aif_out] = 3, + [snd_soc_dapm_mic] = 4, + [snd_soc_dapm_mux] = 5, + [snd_soc_dapm_value_mux] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mixer] = 7, + [snd_soc_dapm_mixer_named_ctl] = 7, + [snd_soc_dapm_pga] = 8, + [snd_soc_dapm_adc] = 9, + [snd_soc_dapm_hp] = 10, + [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_post] = 11, }; static int dapm_down_seq[] = { @@ -81,8 +83,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_micbias] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_value_mux] = 9, - [snd_soc_dapm_supply] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_aif_in] = 10, + [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_supply] = 11, + [snd_soc_dapm_post] = 12, }; static void pop_wait(u32 pop_time) @@ -224,6 +228,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: p->connect = 1; break; /* does effect routing - dynamically connected */ @@ -497,8 +503,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (widget->id == snd_soc_dapm_supply) return 0; - if (widget->id == snd_soc_dapm_adc && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -537,8 +549,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 0; /* active stream ? */ - if (widget->id == snd_soc_dapm_dac && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -1052,6 +1070,8 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -1342,6 +1362,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_pre: case snd_soc_dapm_post: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: list_add(&path->list, &codec->dapm_paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -1444,9 +1466,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) dapm_new_mux(codec, w); break; case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: -- 1.6.3.3

1 1

[alsa-devel] No Sound - X-Fi CA0110 IBG - Ubuntu 910 with Kernel 2.6.31-6
by David Favor 24 Aug '09

24 Aug '09

root@home1:~# uname -r 2.6.31-6-generic root@home1:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 0: CA0110 Analog [CA0110 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 1: CA0110 Digital [CA0110 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 root@home1:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.20. root@home1:~# cat /proc/asound/card0/codec#1 Codec: Creative CA0110-IBG Address: 1 Function Id: 0x1 Vendor Id: 0x1102000a Subsystem Id: 0x11021007 Revision Id: 0x100000 No Modem Function Group found Default PCM: N/A Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=255, o=255, i=255, unsolicited=1, wake=1 root@home1:~# lsmod | grep snd snd_hda_codec_ca0110 7868 1 snd_hda_intel 26600 6 snd_hda_codec 69276 2 snd_hda_codec_ca0110,snd_hda_intel snd_pcm 75360 4 snd_hda_intel,snd_hda_codec snd_timer 22244 1 snd_pcm snd 59204 15 snd_hda_codec_ca0110,snd_hda_intel,snd_hda_codec,snd_pcm,snd_timer soundcore 7264 1 snd snd_page_alloc 9156 2 snd_hda_intel,snd_pcm root@home1:~# lspci -vvnn 0d:00.0 Audio device [0403]: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG [1102:0009] Subsystem: Creative Labs Device [1102:0018] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 26 Region 0: Memory at fbefc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel I find no mention anywhere of anyone getting this card to work. Someone let me know if you have a working card or if I should toss this one. Thanks.

3 4

[alsa-devel] [PATCH] hda: move open coded tricks into get_wcaps_channels()
by Wu Fengguang 24 Aug '09

24 Aug '09

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_local.h | 10 ++++++++++ sound/pci/hda/hda_proc.c | 4 +--- sound/pci/hda/patch_atihdmi.c | 3 +-- 3 files changed, 12 insertions(+), 5 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_local.h 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_local.h 2009-08-24 09:49:44.000000000 +0800 @@ -410,6 +410,16 @@ static inline u32 get_wcaps(struct hda_c /* get the widget type from widget capability bits */ #define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT) +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = ((chans << 1) | 1) + 1; + + return chans; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); --- sound-2.6.orig/sound/pci/hda/hda_proc.c 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_proc.c 2009-08-24 09:49:44.000000000 +0800 @@ -515,9 +515,7 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); if (wid_caps & AC_WCAP_STEREO) { - unsigned int chans; - chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + unsigned int chans = get_wcaps_channels(wid_caps); if (chans == 2) snd_iprintf(buffer, " Stereo"); else --- sound-2.6.orig/sound/pci/hda/patch_atihdmi.c 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_atihdmi.c 2009-08-24 09:49:44.000000000 +0800 @@ -141,8 +141,7 @@ static int atihdmi_build_pcms(struct hda /* FIXME: we must check ELD and change the PCM parameters dynamically */ chans = get_wcaps(codec, CVT_NID); - chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + chans = get_wcaps_channels(chans); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; return 0;

2 1

[PATCH] ASoC: add missing inclusion of debugfs.h
by Takashi Iwai 24 Aug '09

24 Aug '09

To fix compile errors. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2f5295d..7b74547 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -37,6 +37,7 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.6.3.3

1 0

[alsa-devel] [PATCH] ali5451: remove dead code
by Bartlomiej Zolnierkiewicz 23 Aug '09

23 Aug '09

From: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> Subject: [PATCH] ali5451: remove dead code Remove code covered by #if/endif 0 and #ifdef/endif CODEC_RESET (CODEC_RESET is never defined). Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 65 -------------------------------------------- 1 file changed, 65 deletions(-) Index: b/sound/pci/ali5451/ali5451.c =================================================================== --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -477,45 +477,6 @@ static int snd_ali_reset_5451(struct snd return 0; } -#ifdef CODEC_RESET - -static int snd_ali_reset_codec(struct snd_ali *codec) -{ - struct pci_dev *pci_dev; - unsigned char bVal; - unsigned int dwVal; - unsigned short wCount, wReg; - - pci_dev = codec->pci_m1533; - - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); - udelay(5000); - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); - udelay(5000); - - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal |= 0x02; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(5000); - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal &= 0xfd; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(15000); - - wCount = 200; - while (wCount--) { - wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN); - if ((wReg & 0x000f) == 0x000f) - return 0; - udelay(5000); - } - return -1; -} - -#endif - /* * ALI 5451 Controller */ @@ -560,22 +521,6 @@ static void snd_ali_disable_address_inte outl(gc, ALI_REG(codec, ALI_GC_CIR)); } -#if 0 /* not used */ -static void snd_ali_enable_voice_irq(struct snd_ali *codec, - unsigned int channel) -{ - unsigned int mask; - struct snd_ali_channel_control *pchregs = &(codec->chregs); - - snd_ali_printk("enable_voice_irq channel=%d\n",channel); - - mask = 1 << (channel & 0x1f); - pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten)); - pchregs->data.ainten |= mask; - outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten)); -} -#endif - static void snd_ali_disable_voice_irq(struct snd_ali *codec, unsigned int channel) { @@ -676,16 +621,6 @@ static void snd_ali_free_channel_pcm(str } } -#if 0 /* not used */ -static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel) -{ - unsigned int mask = 1 << (channel & 0x1f); - - snd_ali_printk("start_voice: channel=%d\n",channel); - outl(mask, ALI_REG(codec,codec->chregs.regs.start)); -} -#endif - static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel) { unsigned int mask = 1 << (channel & 0x1f);

2 1

[alsa-devel] [PATCH] ali5451: fix timeout handling in snd_ali_{codecs, timer}_ready()
by Bartlomiej Zolnierkiewicz 23 Aug '09

23 Aug '09

From: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> Subject: [PATCH] ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready() Modify loops in such way that the register value is checked also after the timeout condition, just in case the heavy interrupt load etc. caused the thread to sleep for the time period exceeding the timeout value. While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready(). Reported-by: Jack Byer <ojbyer(a)usa.net> Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) Index: b/sound/pci/ali5451/ali5451.c =================================================================== --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -314,8 +314,11 @@ static int snd_ali_codec_ready(struct sn res = snd_ali_5451_peek(codec,port); if (!(res & 0x8000)) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } while (1); + snd_ali_5451_poke(codec, port, res & ~0x8000); snd_printdd("ali_codec_ready: codec is not ready.\n "); return -EIO; @@ -327,15 +330,17 @@ static int snd_ali_stimer_ready(struct s unsigned long dwChk1,dwChk2; dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER); - dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); end_time = jiffies + msecs_to_jiffies(250); do { dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); if (dwChk2 != dwChk1) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } while (1); + snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n"); return -EIO; }

2 3

[alsa-devel] [PATCH] ASoC: free socdev if init_card() fails in wm9705_soc_probe()
by Roel Kluin 23 Aug '09

23 Aug '09

Free socdev if snd_soc_init_card() fails. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- It doesn't seem to happen in snd_soc_init_card(), but please review. diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index fa88b46..e7d2840 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,7 +406,7 @@ static int wm9705_soc_probe(struct platform_device *pdev) ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register card\n"); - goto pcm_err; + goto reset_err; } return 0;

2 1

Re: [alsa-devel] ALSA synchronization
by arif setiawan 23 Aug '09

23 Aug '09

Hi, We are working on audio processing (encryption-decryption) and audio will be sent using radio. Our configuration is : Audio source - ALSA audio in - encryption - ALSA audio out - transmission over radio (now we still use audio cable) - ALSA audio in - decryption - ALSA audio out - speaker. So data will be sent through radio channel and its converted to analog form by ALSA PCM during playback. We want to solve synchronization issue by using data header so that we can decrypt correct data block. So far we still unable to verify that data we received in the decryption side is similar to data that we sent from encryption box. Supposed that we filled buffer in encryption box with some values, when we check buffer in decryption box, we can't get same data. Our guess that data somehow changed in digital-analog/analog-digital conversion or we might make mistakes in filling the buffer so data is not compatible with ALSA PCM. Any suggestion to solve this problem? Thank you, Best Regards, Arif

3 3

[alsa-devel] [PATCH] Convert WM97xx-battery to dev_pm_ops
by Marek Vasut 22 Aug '09

22 Aug '09

Hi, Anton, are you fine with this one? Mark, in case Anton is ok, would you mind applying? Thanks! Cheers! Marek Vasut

1 0

[alsa-devel] ALSA synchronization
by arif setiawan 21 Aug '09

21 Aug '09

Hi all, We are developing a AES based speech encryption module using beagleboard. We have two beagleboard: one for encryption and other for decryption. We use audio-in/audio-out for sending speech. We have problem with synchronization of audio stream from encryption board to decryption board. Since AES is block cipher, the buffer content received by decryption board must be the same as sent from encryption board. Right now, we want to use some bytes as data header, but we don't now how to make this work. Any idea how to solve synchronization in ALSA? Best Regards, Arif

2 1

[alsa-devel] [PATCH] various acer-mic-internal bugs solution
by epic 21 Aug '09

21 Aug '09

Several mic related bugs maybe solved with this patch or a derived one 1 -My internal-mic was not working and front jack mic sound was terrible (useless for headset mic). 2 -Detected offset level from inputs {line, mic}, this offset level increases with specific [boost] and/or [capture(NID 0x23)] gain. * I just played around several days compiling and testing alsa with my acer-alc268, checked all data communication with alc268, pin config etc. and tested different values for offset in pins. * Found internal-mic comes alive with a 50% offset v_reference and mic sound is much better. *This patch applies/solves only for alc268 with option model=acer in alsa-base.conf, but i think the others acer-alcxxx bugs maybe solved in the same way. changing PIN_VREF80 to PIN_VREF50 at NID 0x18 mic NID 0x19 internal-mic NID 0x1a line-in (not tested) *Don't know what can happen to other acer-alcxxx, so i only patched acer-alc268. But this patch solved my mic-internal issue and most important for me now i can use a headset mic. *It remains a level offset, lower than before, but mic is usable, the level is showed in the pictures. *Pictures: input with two channels and a positive offset in left channel (red), negative offset in rigth channel (blue) http://imagebin.ca/view/Y8TyujP.html working internal microphone ok http://imagebin.ca/view/nzTzeX.html Realtek datasheet 1.3 is available for alc268 2.6.26-2-686 #1 SMP i686 GNU/Linux debian/squeeze description: Computer product: Aspire 7720 vendor: Acer version: V1.17 TAGS: alc268 offset level internal mic microphone alsa hda intel realtek Signed-off-by: EliasTG <epicadocan a.t gmail.com> upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Tue Aug 18 21:27:39 UTC 2009 !!Linux Distribution !!------------------ Debian GNU/Linux squeeze/sid \n \l !!Kernel Information !!------------------ Kernel release: 2.6.26-2-686 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 11 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1025:011e !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-hda-intel: index=0 model=test !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : 0,-1,-1,-1,-1,-1,-1,-1 model : test,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC268 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19840: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30941: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0281304e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000010: OUT Pin Default 0x02451130: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Processing Coefficient: 0x03 Coefficient Index: 0x01 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 Codec: Conexant ID 2c06 Address: 1 Function Id: 0x2 Vendor Id: 0x14f12c06 Subsystem Id: 0x1025011e Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Aug 18 22:02 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Aug 18 22:02 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Aug 18 22:02 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Aug 18 22:02 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Aug 18 22:02 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Aug 18 05:24 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 18 05:24 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0x9b300000 irq 11' Mixer name : 'Realtek ALC268' Components : 'HDA:10ec0268,1025011e,00100003 HDA:14f12c06,1025011e,00100000' Controls : 12 Simple ctrls : 9 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 44 [69%] [-20.00dB] [on] Front Right: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 185 [73%] [-14.00dB] Front Right: Playback 185 [73%] [-14.00dB] Simple mixer control 'Line In Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 2 [100%] Front Right: 2 [100%] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 21 [68%] [15.00dB] [on] Front Right: Capture 21 [68%] [15.00dB] [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 12 Mono: Front Left: Playback 0 [0%] [-24.00dB] [off] Front Right: Playback 0 [0%] [-24.00dB] [off] Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Internal Mic' 'Line' Item0: 'Mic' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 2 [100%] Front Right: 2 [100%] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 44 value.1 44 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Master Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Mic Boost' value.0 2 value.1 2 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Internal Mic Boost' value.0 2 value.1 2 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Line In Boost' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 21 value.1 21 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.8 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Internal Mic' comment.item.2 Line iface MIXER name 'Input Source' value Mic } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 12' comment.dbmin -2400 comment.dbmax 0 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 false value.1 false } control.11 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 185 value.1 185 } control.12 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_hwdep usb_storage nvidia ppdev parport_pc lp parport ipv6 sbp2 loop snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy snd_seq_oss snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device mxl5005s zl10353 snd arc4 ecb crypto_blkcipher soundcore joydev dvb_usb_ce6230 dvb_usb iwl3945 snd_page_alloc dvb_core uvcvideo serio_raw mac80211 i2c_i801 acer_wmi firmware_class videodev v4l1_compat cfg80211 i2c_core psmouse pcspkr led_class evdev video output battery button ac wmi ext2 mbcache usbhid hid ff_memless ide_cd_mod cdrom ide_pci_generic sd_mod ata_generic uhci_hcd ahci sdhci libata ricoh_mmc mmc_core ohci1394 scsi_mod dock ieee1394 piix ide_core tg3 ehci_hcd usbcore intel_agp agpgart thermal processor fan thermal_sys dm_mirror dm_log dm_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x411111f0 0x13 0x411111f0 0x14 0x02211020 0x15 0x99130110 0x16 0x411111f0 0x18 0x02a19840 0x19 0x99a30941 0x1a 0x0281304e 0x1c 0x411111f0 0x1d 0x4017952d 0x1e 0x02451130 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D1/init_pin_configs: 0x73 0x016a10f0 /sys/class/sound/hwC0D1/driver_pin_configs: /sys/class/sound/hwC0D1/user_pin_configs: /sys/class/sound/hwC0D1/init_verbs: Codec: Realtek ALC268 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19840: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30941: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0281304e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000010: OUT Pin Default 0x02451130: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Processing Coefficient: 0xa948 Coefficient Index: 0x02 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 [617100.073588] hda_codec:patch_alc268 here we go! v2 [617100.079773] hda_codec:snd_hda_codec_write: 0x170e01 [617100.079856] hda_codec:snd_hda_codec_read: 0x1f000d --> 0x0 [617100.079954] hda_codec:snd_hda_codec_read: 0x100f0000 --> 0x14f12c06 [617100.080028] hda_codec:snd_hda_codec_read: 0x100f0001 --> 0x0 [617100.080105] hda_codec:snd_hda_codec_read: 0x100f0002 --> 0x100000 [617100.080170] hda_codec:snd_hda_codec_read: 0x100f0004 --> 0x20001 [617100.080257] hda_codec:snd_hda_codec_read: 0x102f0005 --> 0x102 [617100.080333] hda_codec:snd_hda_codec_read: 0x102f0004 --> 0x700004 [617100.080398] hda_codec:snd_hda_codec_read: 0x170f0009 --> 0xf00000 [617100.080464] hda_codec:snd_hda_codec_read: 0x171f0009 --> 0x100100 [617100.080548] hda_codec:snd_hda_codec_read: 0x172f0009 --> 0x0 [617100.080625] hda_codec:snd_hda_codec_read: 0x173f0009 --> 0x400100 [617100.080689] hda_codec:snd_hda_codec_read: 0x173f1c00 --> 0x16a10f0 [617100.080756] hda_codec:snd_hda_codec_read: 0x102f2000 --> 0x1025011e [617100.080762] hda_codec:snd_hda_codec_write: 0x10270500 [617100.110915] hda_codec:snd_hda_codec_read: 0x102f0500 --> 0x0 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f000a --> 0xe0560 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f000b --> 0x1 [617100.114318] hda_codec:snd_hda_codec_read: 0x8f000a --> 0x60160 [617100.114318] hda_codec:snd_hda_codec_read: 0x8f000b --> 0x1 [617100.114318] hda_codec:snd_hda_codec_write: 0x17ff00 [617100.114318] hda_codec:snd_hda_codec_write: 0x23b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x33b000 [617100.114318] hda_codec:snd_hda_codec_write: 0xe37000 [617100.114318] hda_codec:snd_hda_codec_write: 0xe70100 [617100.114318] hda_codec:snd_hda_codec_write: 0xf37000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1037000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470740 [617100.114318] hda_codec:snd_hda_codec_write: 0x15707c0 [617100.114318] hda_codec:snd_hda_codec_write: 0x1670740 [617100.114318] hda_codec:snd_hda_codec_write: 0x1870722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1970722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1c70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1d70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x143b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x153b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x163b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x183b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a3b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x1d37000 [617100.114318] hda_codec:snd_hda_codec_write: 0xf37180 [617100.114318] hda_codec:snd_hda_codec_write: 0x1037180 [617100.114318] hda_codec:snd_hda_codec_write: 0x2370100 [617100.114318] hda_codec:snd_hda_codec_write: 0x233b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x2470100 [617100.114318] hda_codec:snd_hda_codec_write: 0x243b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470c02 [617100.114318] hda_codec:snd_hda_codec_write: 0x1570c02 [617100.114318] hda_codec:snd_hda_codec_write: 0x1270720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1370720 [617100.114318] hda_codec:snd_hda_codec_write: 0x14707c0 [617100.114318] hda_codec:snd_hda_codec_write: 0x1570740 [617100.114318] hda_codec:snd_hda_codec_write: 0x1870722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a70722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470884 [617100.114318] hda_codec:snd_hda_codec_read: 0x14f0900 --> 0x0 [617100.114318] hda_codec:snd_hda_codec_read: 0x14ba000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x15ba000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x15b8000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f0012 --> 0x34040 [617100.114318] hda_codec:snd_hda_codec_read: 0x23f0012 --> 0x80051f0b [617100.135224] hda_codec:snd_hda_codec_read: 0x14f0900 --> 0x80000000 [617100.162363] hda_codec:snd_hda_codec_read: 0x18f000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x19f000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x1af000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x2ba000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x23a036 [617100.162363] hda_codec:snd_hda_codec_read: 0x2b8000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x239036 [617100.162363] hda_codec:snd_hda_codec_read: 0x3ba000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x33a036 [617100.162363] hda_codec:snd_hda_codec_read: 0x3b8000 --> 0x0 [617100.162377] hda_codec:snd_hda_codec_write: 0x339036 [617100.162387] hda_codec:snd_hda_codec_write: 0x143a000 [617100.162491] hda_codec:snd_hda_codec_read: 0x14b8000 --> 0x80 [617100.162498] hda_codec:snd_hda_codec_write: 0x1439000 [617100.162557] hda_codec:snd_hda_codec_read: 0x18b2000 --> 0x0 [617100.162621] hda_codec:snd_hda_codec_read: 0x18b0000 --> 0x0 [617100.162702] hda_codec:snd_hda_codec_read: 0x19b2000 --> 0x0 [617100.162789] hda_codec:snd_hda_codec_read: 0x19b0000 --> 0x0 [617100.162869] hda_codec:snd_hda_codec_read: 0x1ab2000 --> 0x0 [617100.162955] hda_codec:snd_hda_codec_read: 0x1ab0000 --> 0x0 [617100.163035] hda_codec:snd_hda_codec_read: 0x23ba000 --> 0x0 [617100.163122] hda_codec:snd_hda_codec_read: 0x23b8000 --> 0x0 [617100.163128] hda_codec:snd_hda_codec_write: 0x2339006 [617100.163136] hda_codec:snd_hda_codec_write: 0x233a080 [617100.163239] hda_codec:snd_hda_codec_read: 0x1db2000 --> 0x0 [617100.163306] hda_codec:snd_hda_codec_read: 0x1db0000 --> 0x0 [617100.163385] hda_codec:snd_hda_codec_read: 0xfb2001 --> 0x80 [617100.163452] hda_codec:snd_hda_codec_read: 0xfb0001 --> 0x80 [617100.163538] hda_codec:snd_hda_codec_read: 0x10b2001 --> 0x80 [617100.163615] hda_codec:snd_hda_codec_read: 0x10b0001 --> 0x80 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000a --> 0xe0560 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000b --> 0x1 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000d --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0012 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0004 --> 0x20023 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0011 --> 0x40000004 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1600 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1700 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1800 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1900 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1a00 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1500 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0009 --> 0x1d [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0012 --> 0x34040 [617165.253398] hda_codec:snd_hda_codec_read: 0x2ba000 --> 0x36 [617165.253398] hda_codec:snd_hda_codec_read: 0x2b8000 --> 0x36 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0600 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f000a --> 0xe0560 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f000b --> 0x1 [617165.253398] hda_codec:snd_hda_codec_read: 0x3f0009 --> 0x1d [617165.253398] hda_codec:snd_hda_codec_read: 0x3f0012 --> 0x34040 [617165.253398] hda_codec:snd_hda_codec_read: 0x3ba000 --> 0x36 [617165.255878] hda_codec:snd_hda_codec_read: 0x3b8000 --> 0x36 [617165.255942] hda_codec:snd_hda_codec_read: 0x3f0600 --> 0x0 [617165.256009] hda_codec:snd_hda_codec_read: 0x3f000a --> 0xe0560 [617165.256095] hda_codec:snd_hda_codec_read: 0x3f000b --> 0x1 [617165.256174] hda_codec:snd_hda_codec_read: 0x4f0009 --> 0xf00000 [617165.256239] hda_codec:snd_hda_codec_read: 0x5f0009 --> 0xf00000 [617165.256316] hda_codec:snd_hda_codec_read: 0x6f0009 --> 0x211 [617165.256383] hda_codec:snd_hda_codec_read: 0x6f0600 --> 0x0 [617165.256447] hda_codec:snd_hda_codec_read: 0x6f0d00 --> 0x0 [617165.256524] hda_codec:snd_hda_codec_read: 0x6f000a --> 0x1e05e0 [617165.256590] hda_codec:snd_hda_codec_read: 0x6f000b --> 0x1 [617165.256676] hda_codec:snd_hda_codec_read: 0x7f0009 --> 0x100111 [617165.256754] hda_codec:snd_hda_codec_read: 0x7f000e --> 0x1 [617165.256818] hda_codec:snd_hda_codec_read: 0x7f0200 --> 0x24 [617165.256884] hda_codec:snd_hda_codec_read: 0x7f0600 --> 0x0 [617165.256961] hda_codec:snd_hda_codec_read: 0x7f0400 --> 0x0 [617165.257025] hda_codec:snd_hda_codec_read: 0x7f000a --> 0x60160 [617165.257092] hda_codec:snd_hda_codec_read: 0x7f000b --> 0x1 [617165.257171] hda_codec:snd_hda_codec_read: 0x8f0009 --> 0x100111 [617165.257235] hda_codec:snd_hda_codec_read: 0x8f000e --> 0x1 [617165.257321] hda_codec:snd_hda_codec_read: 0x8f0200 --> 0x23 [617165.257398] hda_codec:snd_hda_codec_read: 0x8f0600 --> 0x0 [617165.257464] hda_codec:snd_hda_codec_read: 0x8f0400 --> 0x0 [617165.257529] hda_codec:snd_hda_codec_read: 0x8f000a --> 0x60160 [617165.257605] hda_codec:snd_hda_codec_read: 0x8f000b --> 0x1 [617165.257674] hda_codec:snd_hda_codec_read: 0x9f0009 --> 0xf00000 [617165.257759] hda_codec:snd_hda_codec_read: 0xaf0009 --> 0xf00000 [617165.257836] hda_codec:snd_hda_codec_read: 0xbf0009 --> 0xf00000 [617165.257903] hda_codec:snd_hda_codec_read: 0xcf0009 --> 0xf00000 [617165.257980] hda_codec:snd_hda_codec_read: 0xdf0009 --> 0xf00000 [617165.258047] hda_codec:snd_hda_codec_read: 0xef0009 --> 0x20010a [617165.258112] hda_codec:snd_hda_codec_read: 0xef000e --> 0x1 [617165.258188] hda_codec:snd_hda_codec_read: 0xef0200 --> 0x2 [617165.258255] hda_codec:snd_hda_codec_read: 0xef000d --> 0x80000000 [617165.258320] hda_codec:snd_hda_codec_read: 0xeb0000 --> 0x0 [617165.258397] hda_codec:snd_hda_codec_read: 0xff0009 --> 0x20010b [617165.258464] hda_codec:snd_hda_codec_read: 0xff000e --> 0x2 [617165.258528] hda_codec:snd_hda_codec_read: 0xff0200 --> 0x1d02 [617165.258605] hda_codec:snd_hda_codec_read: 0xff000d --> 0x80000000 [617165.258672] hda_codec:snd_hda_codec_read: 0xfb2000 --> 0x0 [617165.258756] hda_codec:snd_hda_codec_read: 0xfb0000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0xfb2001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0xfb0001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f0009 --> 0x20010b [617165.259833] hda_codec:snd_hda_codec_read: 0x10f000e --> 0x3 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f0200 --> 0x21d03 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f000d --> 0x80000000 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2002 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0002 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x11f0009 --> 0xf00000 [617165.259833] hda_codec:snd_hda_codec_read: 0x12f0009 --> 0x400001 [617165.259868] hda_codec:snd_hda_codec_read: 0x12f000c --> 0x20 [617165.259945] hda_codec:snd_hda_codec_read: 0x12f1c00 --> 0x411111f0 [617165.260012] hda_codec:snd_hda_codec_read: 0x12f0700 --> 0x20 [617165.260089] hda_codec:snd_hda_codec_read: 0x13f0009 --> 0x400001 [617165.260155] hda_codec:snd_hda_codec_read: 0x13f000c --> 0x20 [617165.260220] hda_codec:snd_hda_codec_read: 0x13f1c00 --> 0x411111f0 [617165.260298] hda_codec:snd_hda_codec_read: 0x13f0700 --> 0x20 [617165.260364] hda_codec:snd_hda_codec_read: 0x14f0009 --> 0x40018d [617165.260449] hda_codec:snd_hda_codec_read: 0x14f000e --> 0x1 [617165.260526] hda_codec:snd_hda_codec_read: 0x14f0200 --> 0xf [617165.260590] hda_codec:snd_hda_codec_read: 0x14f0012 --> 0x80000000 [617165.260657] hda_codec:snd_hda_codec_read: 0x14ba000 --> 0x0 [617165.260734] hda_codec:snd_hda_codec_read: 0x14b8000 --> 0x0 [617165.260798] hda_codec:snd_hda_codec_read: 0x14f000c --> 0x1003c [617165.260865] hda_codec:snd_hda_codec_read: 0x14f0c00 --> 0x2 [617165.260949] hda_codec:snd_hda_codec_read: 0x14f1c00 --> 0x2211020 [617165.261027] hda_codec:snd_hda_codec_read: 0x14f0700 --> 0xc0 [617165.261094] hda_codec:snd_hda_codec_read: 0x14f0800 --> 0x84 [617165.261179] hda_codec:snd_hda_codec_read: 0x15f0009 --> 0x40018d [617165.261256] hda_codec:snd_hda_codec_read: 0x15f000e --> 0x1 [617165.261322] hda_codec:snd_hda_codec_read: 0x15f0200 --> 0x10 [617165.261399] hda_codec:snd_hda_codec_read: 0x15f0012 --> 0x80000000 [617165.261464] hda_codec:snd_hda_codec_read: 0x15ba000 --> 0x80 [617165.261530] hda_codec:snd_hda_codec_read: 0x15b8000 --> 0x80 [617165.261615] hda_codec:snd_hda_codec_read: 0x15f000c --> 0x1003c [617165.261692] hda_codec:snd_hda_codec_read: 0x15f0c00 --> 0x2 [617165.261756] hda_codec:snd_hda_codec_read: 0x15f1c00 --> 0x99130110 [617165.261845] hda_codec:snd_hda_codec_read: 0x15f0700 --> 0x40 [617165.261921] hda_codec:snd_hda_codec_read: 0x15f0800 --> 0x0 [617165.262004] hda_codec:snd_hda_codec_read: 0x16f0009 --> 0x40010c [617165.262071] hda_codec:snd_hda_codec_read: 0x16f000e --> 0x1 [617165.262156] hda_codec:snd_hda_codec_read: 0x16f0200 --> 0xe [617165.262232] hda_codec:snd_hda_codec_read: 0x16f0012 --> 0x80000000 [617165.262299] hda_codec:snd_hda_codec_read: 0x16b8000 --> 0x80 [617165.262386] hda_codec:snd_hda_codec_read: 0x16f000c --> 0x10 [617165.262462] hda_codec:snd_hda_codec_read: 0x16f1c00 --> 0x411111f0 [617165.262551] hda_codec:snd_hda_codec_read: 0x16f0700 --> 0x40 [617165.262628] hda_codec:snd_hda_codec_read: 0x17f0009 --> 0xf00000 [617165.262693] hda_codec:snd_hda_codec_read: 0x18f0009 --> 0x40018f [617165.262780] hda_codec:snd_hda_codec_read: 0x18f000e --> 0x1 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0200 --> 0x2 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f000d --> 0x4f0200 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b2000 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b0000 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0012 --> 0x80000000 [617165.263834] hda_codec:snd_hda_codec_read: 0x18ba000 --> 0x80 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b8000 --> 0x80 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f000c --> 0x3734 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f1c00 --> 0x2a19840 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0700 --> 0x22 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0800 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x19f0009 --> 0x40008b [617165.263834] hda_codec:snd_hda_codec_read: 0x19f000d --> 0x4f0200 [617165.263838] hda_codec:snd_hda_codec_read: 0x19b2000 --> 0x0 [617165.263904] hda_codec:snd_hda_codec_read: 0x19b0000 --> 0x0 [617165.263969] hda_codec:snd_hda_codec_read: 0x19f000c --> 0x3724 [617165.264046] hda_codec:snd_hda_codec_read: 0x19f1c00 --> 0x99a30941 [617165.264114] hda_codec:snd_hda_codec_read: 0x19f0700 --> 0x22 [617165.264199] hda_codec:snd_hda_codec_read: 0x19f0800 --> 0x0 [617165.264276] hda_codec:snd_hda_codec_read: 0x1af0009 --> 0x40018f [617165.264343] hda_codec:snd_hda_codec_read: 0x1af000e --> 0x1 [617165.264429] hda_codec:snd_hda_codec_read: 0x1af0200 --> 0x2 [617165.264505] hda_codec:snd_hda_codec_read: 0x1af000d --> 0x4f0200 [617165.264570] hda_codec:snd_hda_codec_read: 0x1ab2000 --> 0x0 [617165.264657] hda_codec:snd_hda_codec_read: 0x1ab0000 --> 0x0 [617165.264733] hda_codec:snd_hda_codec_read: 0x1af0012 --> 0x80000000 [617165.264800] hda_codec:snd_hda_codec_read: 0x1aba000 --> 0x80 [617165.264884] hda_codec:snd_hda_codec_read: 0x1ab8000 --> 0x80 [617165.264961] hda_codec:snd_hda_codec_read: 0x1af000c --> 0x3734 [617165.265026] hda_codec:snd_hda_codec_read: 0x1af1c00 --> 0x281304e [617165.265114] hda_codec:snd_hda_codec_read: 0x1af0700 --> 0x22 [617165.265191] hda_codec:snd_hda_codec_read: 0x1af0800 --> 0x0 [617165.265258] hda_codec:snd_hda_codec_read: 0x1bf0009 --> 0xf00000 [617165.265335] hda_codec:snd_hda_codec_read: 0x1cf0009 --> 0x400001 [617165.265400] hda_codec:snd_hda_codec_read: 0x1cf000c --> 0x20 [617165.265467] hda_codec:snd_hda_codec_read: 0x1cf1c00 --> 0x411111f0 [617165.265545] hda_codec:snd_hda_codec_read: 0x1cf0700 --> 0x20 [617165.265610] hda_codec:snd_hda_codec_read: 0x1df0009 --> 0x400000 [617165.265697] hda_codec:snd_hda_codec_read: 0x1df000c --> 0x20 [617165.265773] hda_codec:snd_hda_codec_read: 0x1df1c00 --> 0x4017952d [617165.265842] hda_codec:snd_hda_codec_read: 0x1df0700 --> 0x20 [617165.265927] hda_codec:snd_hda_codec_read: 0x1ef0009 --> 0x400380 [617165.266003] hda_codec:snd_hda_codec_read: 0x1ef000e --> 0x1 [617165.266070] hda_codec:snd_hda_codec_read: 0x1ef0200 --> 0x6 [617165.266146] hda_codec:snd_hda_codec_read: 0x1ef000c --> 0x10 [617165.266210] hda_codec:snd_hda_codec_read: 0x1ef1c00 --> 0x2451130 [617165.266278] hda_codec:snd_hda_codec_read: 0x1ef0700 --> 0x40 [617165.266355] hda_codec:snd_hda_codec_read: 0x1ef0800 --> 0x0 [617165.266420] hda_codec:snd_hda_codec_read: 0x1ff0009 --> 0xf00000 [617165.266507] hda_codec:snd_hda_codec_read: 0x20f0009 --> 0xf00040 [617165.266584] hda_codec:snd_hda_codec_read: 0x20f0010 --> 0xa00 [617165.266651] hda_codec:snd_hda_codec_read: 0x20c0000 --> 0x3 [617165.266738] hda_codec:snd_hda_codec_read: 0x20d0000 --> 0x1 [617165.266815] hda_codec:snd_hda_codec_read: 0x21f0009 --> 0xf00000 [617165.267835] hda_codec:snd_hda_codec_read: 0x22f0009 --> 0xf00000 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0009 --> 0x30010d [617165.267835] hda_codec:snd_hda_codec_read: 0x23f000e --> 0x7 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0200 --> 0x1c1a1918 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0204 --> 0x121514 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0012 --> 0x80051f0b [617165.267835] hda_codec:snd_hda_codec_read: 0x23ba000 --> 0x80 [617165.267835] hda_codec:snd_hda_codec_read: 0x23b8000 --> 0x6 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0100 --> 0x0 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0009 --> 0x30010d [617165.267835] hda_codec:snd_hda_codec_read: 0x24f000e --> 0x7 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0200 --> 0x1c1a1918 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0204 --> 0x131514 [617165.267870] hda_codec:snd_hda_codec_read: 0x24f0012 --> 0x80051f0b [617165.267947] hda_codec:snd_hda_codec_read: 0x24ba000 --> 0x0 [617165.268012] hda_codec:snd_hda_codec_read: 0x24b8000 --> 0x0 [617165.268098] hda_codec:snd_hda_codec_read: 0x24f0100 --> 0x0 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Acer Incorporated [ALI] Device [1025:011e] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 11 Region 0: Memory at 9b300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

2 1

[alsa-devel] [PATCH v3 4/4] sh: add FSI driver support for ms7724se
by Kuninori Morimoto 21 Aug '09

21 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove un-used dma-sh.h I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 arch/sh/boards/mach-se/7724/setup.c | 89 +++++++++++++++++++++++++++++++++++ 1 files changed, 89 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..eb144f6 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,6 +22,7 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/sh_fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> @@ -246,6 +247,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +411,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +479,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +621,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

2 1

[alsa-devel] [PATCH v3 3/4] ASoC: Add FSI-AK4642 sound support for SuperH
by Kuninori Morimoto 21 Aug '09

21 Aug '09

This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove un-used ak4642_setup_data I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 sound/soc/sh/Kconfig | 8 +++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 107 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 117 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-ak4642.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 01943a1..9154b43 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,4 +40,12 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. +config SND_FSI_AK4642 + bool "FSI-AK4642 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_AK4642 + help + This option enables generic sound support for the + FSI - AK4642 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 9fbcc4a..a699787 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,5 +12,7 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o +snd-soc-fsi-ak4642-objs := fsi-ak4642.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o +obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c new file mode 100644 index 0000000..c7af097 --- /dev/null +++ b/sound/soc/sh/fsi-ak4642.c @@ -0,0 +1,107 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include <../sound/soc/codecs/ak4642.h> + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK4642", + .stream_name = "AK4642", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak4642_dai, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak4642, +}; + +#define AK4642_BUS 0 +#define AK4642_ADR 0x12 +static int ak4642_add_i2c_device(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = AK4642_ADR; + strlcpy(info.type, "ak4642", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(AK4642_BUS); + if (!adapter) { + printk(KERN_DEBUG "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_DEBUG "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak4642_init(void) +{ + int ret = -ENOMEM; + + ak4642_add_i2c_device(); + + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak4642_init); +module_exit(fsi_ak4642_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

2 1

[alsa-devel] [PATCH v3 2/4] ASoC: Add ak4642/ak4643 codec support
by Kuninori Morimoto 21 Aug '09

21 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o add ak4643 in i2c table o remove un-used hw_read o remove un-used ak4642_setup_data I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 502 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 20 ++ 4 files changed, 528 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 167a5ce..0edca93 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -92,6 +93,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fbab43b..fb4af28 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -54,6 +55,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..e057c7b --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,502 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#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 <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 "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { "ak4643", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..e476833 --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,20 @@ +/* + * ak4642.h -- AK4642 Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ak4535.c + * + * 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. + */ + +#ifndef _AK4642_H +#define _AK4642_H + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

2 1

[alsa-devel] [PATCH] pyalsa: fix integer overflow in alsaseq.c
by Chris Coleman 21 Aug '09

21 Aug '09

Hello alsa-devel, I've been using the alsaseq python module and I found a bug. Sometimes the SEQ_* constants have extremely large and incorrect values. For example, 25769803811 instead of 35. The lower 32-bits are always correct. Obviously, I'm running a 64-bit operating system. The problem is that the `value` member of the `ConstantObject` structure is an `unsigned int` whereas it should be a `long`. I've attached a patch. It's against the latest released version, 1.0.20. Regards, Chris Coleman Signed-off-by: Chris Coleman <chris.coleman83 at gmail.com>

2 3

[alsa-devel] [patch 0/4] ASoC S3C24XX/Simtec support
by Ben Dooks 21 Aug '09

21 Aug '09

re-worked against ASoC git branch Will be away from until 26th, any comments will be sorted when back. --

2 6

[alsa-devel] Need help compiling alsa-kernel from git
by Damien Zammit 21 Aug '09

21 Aug '09

Hi, I downloaded a 2.6.30.4 vanilla kernel source tree, merged it with alsa 1.0.20 by copying the files into my kernel source tree, and used that as a base for modifying a sound driver module. I want to submit a patch because I got a new device working. I cloned alsa-kernel git and merged my changes into the tree and committed it. Now, how do I compile it to test it? Do I need to clone linux-2.6 git? I would prefer not to clone that whole tree if possible... I tried copying the sound directory from my working alsa-kernel git tree into a 2.6.30.4 vanilla kernel tree, but it doesnt work. Please help.

1 0

[alsa-devel] [PATCH v2 4/4] sh: add FSI driver support for ms7724se
by Kuninori Morimoto 21 Aug '09

21 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o become independent patch arch/sh/boards/mach-se/7724/setup.c | 90 +++++++++++++++++++++++++++++++++++ 1 files changed, 90 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..bffc90e 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,11 +22,13 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/sh_fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> #include <asm/clock.h> #include <asm/sh_keysc.h> +#include <asm/dma-sh.h> #include <cpu/sh7724.h> #include <mach-se/mach/se7724.h> @@ -246,6 +248,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +412,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +480,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +622,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

3 3

[alsa-devel] Making only some targets on cross-compilation
by Philip A. Prindeville 21 Aug '09

21 Aug '09

Hi. I'm working on the astlinux project (www.astlinux.org) and had a question about doing a build for only a subset of the $SUBDIRS targets. I tried doing something like: make -C $(ALSA-LIB_DIR) \ SUBDIRS='src include modules aserver' but this fails. The problem is that if we look into the environment in the top-level $(RECURSIVE_TARGETS): we get: SUBDIRS=include src modules aserver ... MAKEFLAGS=w --jobserver-fds=3,4 -j -- SUBDIRS=include\ src\ modules\ aserver CC=/home/philipp/asterisk/build_i586/staging_dir/bin/i586-linux-uclibc-gcc so those targets eventually get picked up by the make running in include/, which then overrides his own Makefile value: SUBDIRS = sound and tries to build the same top-level targets now in include/. Clearly this is wrong. Anyone have a work-around to this? We're on an embedded system, and don't want to build (nor install, obviously) "doc", "test", "alsalisp", "utils", etc. Having something like: SUBDIRS=$(filter-out $(SUBDIRS_EXCLUDE),$(SUBDIRS)) would be really sweet. Then I could just build as: make -C $(ALSA-LIB_DIR) \ all SUBDIRS_EXCLUDE='doc test utils' and only get the targets I want... Thanks, -Philip

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 21 Aug '09

21 Aug '09

The following changes since commit 474e09ca017bd0f6e9bf41af643364a21bc27f43: Mark Brown (1): ASoC: Provide default set_bias_level() implementation 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 Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Kuninori Morimoto (1): ASoC: Add SuperH FSI driver support for ALSA Mark Brown (1): ALSA: Restore support for DMAless DAIs on PXA Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1, 3, 4, 5 OMAP: McBSP: Add link DMA mode selection Shine Liu (1): ASoC: S3C24XX : Align the peroid size to the buffer size arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++- include/sound/sh_fsi.h | 83 +++ sound/arm/pxa2xx-pcm-lib.c | 3 + sound/soc/omap/omap-mcbsp.c | 77 +++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/s3c24xx/s3c24xx-pcm.c | 1 + sound/soc/sh/Kconfig | 7 +- sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi.c | 1004 +++++++++++++++++++++++++++++++ 12 files changed, 1610 insertions(+), 14 deletions(-) create mode 100644 include/sound/sh_fsi.h create mode 100644 sound/soc/sh/fsi.c

2 1

[alsa-devel] [PATCH v2 3/4] Add FSI-AK4642 sound support for SuperH
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o move to sound/soc/sh sound/soc/sh/Kconfig | 8 +++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 111 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 121 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-ak4642.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 01943a1..9154b43 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,4 +40,12 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. +config SND_FSI_AK4642 + bool "FSI-AK4642 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_AK4642 + help + This option enables generic sound support for the + FSI - AK4642 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 9fbcc4a..a699787 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,5 +12,7 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o +snd-soc-fsi-ak4642-objs := fsi-ak4642.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o +obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c new file mode 100644 index 0000000..3e5498e --- /dev/null +++ b/sound/soc/sh/fsi-ak4642.c @@ -0,0 +1,111 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include <../sound/soc/codecs/ak4642.h> + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK4642", + .stream_name = "AK4642", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak4642_dai, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +struct ak4642_setup_data ak4642_setup = { + .i2c_bus = 0, + .i2c_address = 0x12, /* 0x13 */ +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak4642, + .codec_data = &ak4642_setup, +}; + +static int ak4642_add_i2c_device(const struct ak4642_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ak4642", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + printk(KERN_DEBUG "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_DEBUG "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak4642_init(void) +{ + int ret = -ENOMEM; + + ak4642_add_i2c_device(&ak4642_setup); + + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak4642_init); +module_exit(fsi_ak4642_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

2 1

[alsa-devel] [PATCH v2 2/4] Add ak464x codec support
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o based on wm8731.c o add explain for register setting sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 516 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 25 +++ 4 files changed, 547 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..ee794f1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -74,6 +75,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..c8c567e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..1b439b4 --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,516 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#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 <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 "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4642_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..aa58c14 --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,25 @@ +/* + * ak464x.h -- AK464x Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * 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. + */ + +#ifndef _AK4642_H +#define _AK4642_H + +struct ak4642_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

2 1

[alsa-devel] [PATCH] pcm_lib.c: Fixed inaccurate calculation of hw_ptr_interrupt in snd_pcm_update_hw_ptr_interrupt function
by Shine Liu 20 Aug '09

20 Aug '09

There's no restriction on that the runtime->period_size should be the multiplicator of the runtime->buffer_size in current kernel code. So "hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;" is not always the trueth. I've encounter the problem of this bug, the kernel told me "PCM: Unexpected hw_pointer value (stream=0, pos=8, intr_ptr=16384)". The period_size and buffer_size I used was 0x800 and 0x3e80 when the bug occured. Following patch will fix the bug. Signed-off-by: Shine Liu <shinel(a)foxmail.com> ----------------------------------------------------------------------- --- a/sound/core/pcm_lib.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/core/pcm_lib.c 2009-08-19 17:59:00.000000000 +0800 @@ -248,6 +248,12 @@ hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; + /* if the period_size is not the multiplicator of the + * buffer_size, the hw_ptr_interrupt calculated above + * may not be correct, should be fixed. + */ + if(hw_ptr_interrupt > hw_base + runtime->buffer_size) + hw_ptr_interrupt = hw_base + runtime->buffer_size; delta = new_hw_ptr - hw_ptr_interrupt; if (hw_ptr_interrupt >= runtime->boundary) { hw_ptr_interrupt -= runtime->boundary;

4 10

[alsa-devel] [PATCHv5 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 20 Aug '09

20 Aug '09

Hello guys, I've updated this series after some previous confusion from my side. Now that all dependencies are merged (Janusz's patches and Jarkko's patch), I've based this series on Takashi's tree (master branch). I've also talked with Tony and he agrees to have this changes merged by alsa tree. And as pointed by Jarkko, the mcbsp code is currently used only by Audio inside OMAP code. So, that should be safe. It is the same series. It now includes patch description for every patch. And also, as I already said, it is based on master branch of sound-2.6. Tested with beagle and rx51. Compiled tested for n770. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 77 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 13 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

5 4

[PATCH] ALSA: pcm - Fix drain behavior in non-blocking mode
by Takashi Iwai 20 Aug '09

20 Aug '09

The current PCM core has the following problems regarding PCM draining in non-blocking mode: - the current f_flags isn't checked in snd_pcm_drain(), thus changing the mode dynamically via snd_pcm_nonblock() after open doesn't work. - calling drain in non-blocking mode just return -EAGAIN error, but doesn't provide any way to sync with draining. This patch fixes these issues. - check file->f_flags in snd_pcm_drain() properly - when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state but quits ioctl immediately without waiting the whole drain; the caller can sync the drain manually via poll() Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 12 +++++++--- sound/core/pcm_native.c | 52 ++++++++++++++++++++++++++-------------------- 2 files changed, 37 insertions(+), 27 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 72cfd47..e3e78c7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -197,12 +197,16 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, avail = snd_pcm_capture_avail(runtime); if (avail > runtime->avail_max) runtime->avail_max = avail; - if (avail >= runtime->stop_threshold) { - if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + if (avail >= runtime->buffer_size) { snd_pcm_drain_done(substream); - else + return -EPIPE; + } + } else { + if (avail >= runtime->stop_threshold) { xrun(substream); - return -EPIPE; + return -EPIPE; + } } if (avail >= runtime->control->avail_min) wake_up(&runtime->sleep); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ac2150e..b08898c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1343,8 +1343,6 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { - if (substream->f_flags & O_NONBLOCK) - return -EAGAIN; substream->runtime->trigger_master = substream; return 0; } @@ -1392,7 +1390,6 @@ static struct action_ops snd_pcm_action_drain_init = { struct drain_rec { struct snd_pcm_substream *substream; wait_queue_t wait; - snd_pcm_uframes_t stop_threshold; }; static int snd_pcm_drop(struct snd_pcm_substream *substream); @@ -1404,13 +1401,15 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream); * After this call, all streams are supposed to be either SETUP or DRAINING * (capture only) state. */ -static int snd_pcm_drain(struct snd_pcm_substream *substream) +static int snd_pcm_drain(struct snd_pcm_substream *substream, + struct file *file) { struct snd_card *card; struct snd_pcm_runtime *runtime; struct snd_pcm_substream *s; int result = 0; int i, num_drecs; + int nonblock = 0; struct drain_rec *drec, drec_tmp, *d; card = substream->pcm->card; @@ -1428,6 +1427,15 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } + if (file) { + if (file->f_flags & O_NONBLOCK) + nonblock = 1; + } else if (substream->f_flags & O_NONBLOCK) + nonblock = 1; + + if (nonblock) + goto lock; /* no need to allocate waitqueues */ + /* allocate temporary record for drain sync */ down_read(&snd_pcm_link_rwsem); if (snd_pcm_stream_linked(substream)) { @@ -1449,16 +1457,11 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) d->substream = s; init_waitqueue_entry(&d->wait, current); add_wait_queue(&runtime->sleep, &d->wait); - /* stop_threshold fixup to avoid endless loop when - * stop_threshold > buffer_size - */ - d->stop_threshold = runtime->stop_threshold; - if (runtime->stop_threshold > runtime->buffer_size) - runtime->stop_threshold = runtime->buffer_size; } } up_read(&snd_pcm_link_rwsem); + lock: snd_pcm_stream_lock_irq(substream); /* resume pause */ if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) @@ -1466,9 +1469,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) /* pre-start/stop - all running streams are changed to DRAINING state */ result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0); - if (result < 0) { - snd_pcm_stream_unlock_irq(substream); - goto _error; + if (result < 0) + goto unlock; + /* in non-blocking, we don't wait in ioctl but let caller poll */ + if (nonblock) { + result = -EAGAIN; + goto unlock; } for (;;) { @@ -1504,18 +1510,18 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } + unlock: snd_pcm_stream_unlock_irq(substream); - _error: - for (i = 0; i < num_drecs; i++) { - d = &drec[i]; - runtime = d->substream->runtime; - remove_wait_queue(&runtime->sleep, &d->wait); - runtime->stop_threshold = d->stop_threshold; + if (!nonblock) { + for (i = 0; i < num_drecs; i++) { + d = &drec[i]; + runtime = d->substream->runtime; + remove_wait_queue(&runtime->sleep, &d->wait); + } + if (drec != &drec_tmp) + kfree(drec); } - - if (drec != &drec_tmp) - kfree(drec); snd_power_unlock(card); return result; @@ -2544,7 +2550,7 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_hw_params_old_user(substream, arg); #endif case SNDRV_PCM_IOCTL_DRAIN: - return snd_pcm_drain(substream); + return snd_pcm_drain(substream, file); case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_PAUSE: -- 1.6.3.3

1 0

[alsa-devel] [PATCHv5 01/20] OMAP: McBSP: Provide functions for ASoC frame syncronization
by Eduardo Valentin 20 Aug '09

20 Aug '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> ASoC has an annoying bug letting either L or R channel to be played on L channel. In other words, L and R channels can switch at random. This provides McBSP funtionality that may be used to fix this feature. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> --- arch/arm/plat-omap/include/mach/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 52 +++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index 57249bb..b85e6e6 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -389,6 +389,8 @@ int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); void omap_mcbsp_start(unsigned int id, int tx, int rx); void omap_mcbsp_stop(unsigned int id, int tx, int rx); +void omap_mcbsp_xmit_enable(unsigned int id, u8 enable); +void omap_mcbsp_recv_enable(unsigned int id, u8 enable); void omap_mcbsp_xmit_word(unsigned int id, u32 word); u32 omap_mcbsp_recv_word(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index a3d2313..0aa2524 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -412,6 +412,58 @@ void omap_mcbsp_stop(unsigned int id, int tx, int rx) } EXPORT_SYMBOL(omap_mcbsp_stop); +void omap_mcbsp_xmit_enable(unsigned int id, u8 enable) +{ + struct omap_mcbsp *mcbsp; + void __iomem *io_base; + u16 w; + + if (!(cpu_is_omap2430() || cpu_is_omap34xx())) + return; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + + mcbsp = id_to_mcbsp_ptr(id); + io_base = mcbsp->io_base; + + w = OMAP_MCBSP_READ(io_base, XCCR); + + if (enable) + OMAP_MCBSP_WRITE(io_base, XCCR, w & ~(XDISABLE)); + else + OMAP_MCBSP_WRITE(io_base, XCCR, w | XDISABLE); +} +EXPORT_SYMBOL(omap_mcbsp_xmit_enable); + +void omap_mcbsp_recv_enable(unsigned int id, u8 enable) +{ + struct omap_mcbsp *mcbsp; + void __iomem *io_base; + u16 w; + + if (!(cpu_is_omap2430() || cpu_is_omap34xx())) + return; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + + mcbsp = id_to_mcbsp_ptr(id); + io_base = mcbsp->io_base; + + w = OMAP_MCBSP_READ(io_base, RCCR); + + if (enable) + OMAP_MCBSP_WRITE(io_base, RCCR, w & ~(RDISABLE)); + else + OMAP_MCBSP_WRITE(io_base, RCCR, w | RDISABLE); +} +EXPORT_SYMBOL(omap_mcbsp_recv_enable); + /* polled mcbsp i/o operations */ int omap_mcbsp_pollwrite(unsigned int id, u16 buf) { -- 1.6.2.GIT -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 19

[alsa-devel] [PATCH v2 1/4] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This driver is very simple. It support playback only now. This patch is tested by ms7724se board. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o it direct use clk_enable/disable o rename fsi.h -> sh_fsi.h include/sound/sh_fsi.h | 83 ++++ sound/soc/sh/Kconfig | 7 +- sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi.c | 1004 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1095 insertions(+), 1 deletions(-) create mode 100644 include/sound/sh_fsi.h create mode 100644 sound/soc/sh/fsi.c diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h new file mode 100644 index 0000000..c022736 --- /dev/null +++ b/include/sound/sh_fsi.h @@ -0,0 +1,83 @@ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.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. + */ + +/* flags format + + * 0xABCDEEFF + * + * A: channel size for TDM (input) + * B: channel size for TDM (ooutput) + * C: inversion + * D: mode + * E: input format + * F: output format + */ + +#include <linux/clk.h> +#include <sound/soc.h> + +/* TDM channel */ +#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) +#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) + +#define SH_FSI_CH_IMASK 0xF0000000 +#define SH_FSI_CH_OMASK 0x0F000000 +#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) +#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) + +/* clock inversion */ +#define SH_FSI_INVERSION_MASK 0x00F00000 +#define SH_FSI_LRM_INV (1 << 20) +#define SH_FSI_BRM_INV (1 << 21) +#define SH_FSI_LRS_INV (1 << 22) +#define SH_FSI_BRS_INV (1 << 23) + +/* mode */ +#define SH_FSI_MODE_MASK 0x000F0000 +#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ +#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ + +/* DI format */ +#define SH_FSI_FMT_MASK 0x000000FF +#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) +#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) +#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) +#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) + +#define SH_FSI_FMT_MONO (1 << 0) +#define SH_FSI_FMT_MONO_DELAY (1 << 1) +#define SH_FSI_FMT_PCM (1 << 2) +#define SH_FSI_FMT_I2S (1 << 3) +#define SH_FSI_FMT_TDM (1 << 4) +#define SH_FSI_FMT_TDM_DELAY (1 << 5) + +#define SH_FSI_IFMT_TDM_CH(x) \ + (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) +#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ + (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) + +#define SH_FSI_OFMT_TDM_CH(x) \ + (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) +#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ + (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +struct sh_fsi_platform_info { + unsigned long porta_flags; + unsigned long portb_flags; +}; + +extern struct snd_soc_dai fsi_soc_dai[2]; +extern struct snd_soc_platform fsi_soc_platform; + +#endif /* __SOUND_FSI_H */ diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 54bd604..01943a1 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -20,7 +20,12 @@ config SND_SOC_SH4_HAC config SND_SOC_SH4_SSI tristate - +config SND_SOC_SH4_FSI + tristate "SH4 FSI support" + depends on CPU_SUBTYPE_SH7724 + select SH_DMA + help + This option enables FSI sound support ## ## Boards diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a8e8ab8..9fbcc4a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -5,8 +5,10 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o ## audio units found on some SH-4 snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o +snd-soc-fsi-objs := fsi.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o +obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c new file mode 100644 index 0000000..4412324 --- /dev/null +++ b/sound/soc/sh/fsi.c @@ -0,0 +1,1004 @@ +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ssi.c + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/list.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/sh_fsi.h> +#include <asm/atomic.h> +#include <asm/dma.h> +#include <asm/dma-sh.h> + +#define DO_FMT 0x0000 +#define DOFF_CTL 0x0004 +#define DOFF_ST 0x0008 +#define DI_FMT 0x000C +#define DIFF_CTL 0x0010 +#define DIFF_ST 0x0014 +#define CKG1 0x0018 +#define CKG2 0x001C +#define DIDT 0x0020 +#define DODT 0x0024 +#define MUTE_ST 0x0028 +#define REG_END MUTE_ST + +#define INT_ST 0x0200 +#define IEMSK 0x0204 +#define IMSK 0x0208 +#define MUTE 0x020C +#define CLK_RST 0x0210 +#define SOFT_RST 0x0214 +#define MREG_START INT_ST +#define MREG_END SOFT_RST + +/* DO_FMT */ +/* DI_FMT */ +#define CR_FMT(param) ((param) << 4) +# define CR_MONO 0x0 +# define CR_MONO_D 0x1 +# define CR_PCM 0x2 +# define CR_I2S 0x3 +# define CR_TDM 0x4 +# define CR_TDM_D 0x5 + +/* DOFF_CTL */ +/* DIFF_CTL */ +#define IRQ_HALF 0x00100000 +#define FIFO_CLR 0x00000001 + +/* DOFF_ST */ +#define ERR_OVER 0x00000010 +#define ERR_UNDER 0x00000001 + +/* CLK_RST */ +#define B_CLK 0x00000010 +#define A_CLK 0x00000001 + +/* INT_ST */ +#define INT_B_IN (1 << 12) +#define INT_B_OUT (1 << 8) +#define INT_A_IN (1 << 4) +#define INT_A_OUT (1 << 0) + +#define FSI_RATES SNDRV_PCM_RATE_8000_96000 + +#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/************************************************************************ + + + struct + + +************************************************************************/ +struct fsi_priv { + void __iomem *base; + struct snd_pcm_substream *substream; + + int fifo_max; + int chan; + int dma_chan; + + int byte_offset; + int period_len; + int buffer_len; + int periods; +}; + +struct fsi_master { + void __iomem *base; + int irq; + struct clk *clk; + struct fsi_priv fsia; + struct fsi_priv fsib; + struct sh_fsi_platform_info *info; +}; + +static struct fsi_master *master; + +/************************************************************************ + + + basic read write function + + +************************************************************************/ +static int __fsi_reg_write(u32 reg, u32 data) +{ + /* valid data area is 24bit */ + data &= 0x00ffffff; + + return ctrl_outl(data, reg); +} + +static u32 __fsi_reg_read(u32 reg) +{ + return ctrl_inl(reg); +} + +static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +{ + u32 val = __fsi_reg_read(reg); + + val &= ~mask; + val |= data & mask; + + return __fsi_reg_write(reg, val); +} + +static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_write((u32)(fsi->base + reg), data); +} + +static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) +{ + if (reg > REG_END) + return 0; + + return __fsi_reg_read((u32)(fsi->base + reg)); +} + +static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); +} + +static int fsi_master_write(u32 reg, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_write((u32)(master->base + reg), data); +} + +static u32 fsi_master_read(u32 reg) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return 0; + + return __fsi_reg_read((u32)(master->base + reg)); +} + +static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); +} + +/************************************************************************ + + + basic function + + +************************************************************************/ +static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd; + struct fsi_priv *fsi = NULL; + + if (!substream || !master) + return NULL; + + rtd = substream->private_data; + switch (rtd->dai->cpu_dai->id) { + case 0: + fsi = &master->fsia; + break; + case 1: + fsi = &master->fsib; + break; + } + + return fsi; +} + +static int fsi_is_port_a(struct fsi_priv *fsi) +{ + /* return + * 1 : port a + * 0 : port b + */ + + if (fsi == &master->fsia) + return 1; + + return 0; +} + +static u32 fsi_get_info_flags(struct fsi_priv *fsi) +{ + int is_porta = fsi_is_port_a(fsi); + + return is_porta ? master->info->porta_flags : + master->info->portb_flags; +} + +static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) +{ + u32 mode; + u32 flags = fsi_get_info_flags(fsi); + + mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; + + /* return + * 1 : master mode + * 0 : slave mode + */ + + return (mode & flags) != mode; +} + +static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play) +{ + int is_porta = fsi_is_port_a(fsi); + u32 data; + + if (is_porta) + data = is_play ? (1 << 0) : (1 << 4); + else + data = is_play ? (1 << 8) : (1 << 12); + + return data; +} + +static void fsi_stream_push(struct fsi_priv *fsi, + struct snd_pcm_substream *substream, + u32 buffer_len, + u32 period_len) +{ + fsi->substream = substream; + fsi->buffer_len = buffer_len; + fsi->period_len = period_len; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static void fsi_stream_pop(struct fsi_priv *fsi) +{ + fsi->substream = NULL; + fsi->buffer_len = 0; + fsi->period_len = 0; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) +{ + u32 status; + u32 reg = is_play ? DOFF_ST : DIFF_ST; + int residue; + + status = fsi_reg_read(fsi, reg); + residue = 0x1ff & (status >> 8); + residue *= fsi->chan; + + return residue; +} + +static int fsi_get_residue(struct fsi_priv *fsi, int is_play) +{ + int residue; + int width; + struct snd_pcm_runtime *runtime; + + runtime = fsi->substream->runtime; + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + if (2 == width) + residue = fsi_get_fifo_residue(fsi, is_play); + else + residue = get_dma_residue(fsi->dma_chan); + + return residue; +} + +/************************************************************************ + + + basic dma function + + +************************************************************************/ +#define PORTA_DMA 0 +#define PORTB_DMA 1 + +static int fsi_get_dma_chan(void) +{ + if (0 != request_dma(PORTA_DMA, "fsia")) + return -EIO; + + if (0 != request_dma(PORTB_DMA, "fsib")) { + free_dma(PORTA_DMA); + return -EIO; + } + + master->fsia.dma_chan = PORTA_DMA; + master->fsib.dma_chan = PORTB_DMA; + + return 0; +} + +static void fsi_free_dma_chan(void) +{ + dma_wait_for_completion(PORTA_DMA); + dma_wait_for_completion(PORTB_DMA); + free_dma(PORTA_DMA); + free_dma(PORTB_DMA); + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; +} + +/************************************************************************ + + + ctrl function + + +************************************************************************/ +static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, data); + fsi_master_mask_set(IEMSK, data, data); +} + +static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, 0); + fsi_master_mask_set(IEMSK, data, 0); +} + +static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) +{ + u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + + if (enable) + fsi_master_mask_set(CLK_RST, val, val); + else + fsi_master_mask_set(CLK_RST, val, 0); +} + +static void fsi_irq_init(struct fsi_priv *fsi, int is_play) +{ + u32 data; + u32 ctrl; + + data = fsi_port_ab_io_bit(fsi, is_play); + ctrl = is_play ? DOFF_CTL : DIFF_CTL; + + /* set IMSK */ + fsi_irq_disable(fsi, is_play); + + /* set interrupt generation factor */ + fsi_reg_write(fsi, ctrl, IRQ_HALF); + + /* clear FIFO */ + fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); + + /* clear interrupt factor */ + fsi_master_mask_set(INT_ST, data, 0); +} + +static void fsi_soft_all_reset(void) +{ + u32 status = fsi_master_read(SOFT_RST); + + /* port AB reset */ + status &= 0x000000ff; + fsi_master_write(SOFT_RST, status); + mdelay(10); + + /* soft reset */ + status &= 0x000000f0; + fsi_master_write(SOFT_RST, status); + status |= 0x00000001; + fsi_master_write(SOFT_RST, status); + mdelay(10); +} + +static void fsi_16data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u16 *dma_start; + u32 snd; + int i; + + /* get dma start position for FSI */ + dma_start = (u16 *)runtime->dma_area; + dma_start += fsi->byte_offset / 2; + + /* + * soft dma + * FSI can not use DMA when 16bpp + */ + for (i = 0; i < send; i++) { + snd = (u32)dma_start[i]; + fsi_reg_write(fsi, DODT, snd << 8); + } +} + +static void fsi_32data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u32 *dma_start; + + /* get dma start position for FSI */ + dma_start = (u32 *)runtime->dma_area; + dma_start += fsi->byte_offset / 4; + + dma_wait_for_completion(fsi->dma_chan); + dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); + dma_write(fsi->dma_chan, (u32)dma_start, + (u32)(fsi->base + DODT), send * 4); +} + +/* playback interrupt */ +static int fsi_data_push(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int send; + int fifo_free; + int width; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get send size for alsa */ + send = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get FIFO free size */ + fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1); + + /* size check */ + if (fifo_free < send) + send = fifo_free; + + if (2 == width) + fsi_16data_push(fsi, runtime, send); + else if (4 == width) + fsi_32data_push(fsi, runtime, send); + else + return -EINVAL; + + fsi->byte_offset += send * width; + + fsi_irq_enable(fsi, 1); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + +static irqreturn_t fsi_interrupt(int irq, void *data) +{ + u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(INT_ST); + + /* clear irq status */ + fsi_master_write(SOFT_RST, status); + fsi_master_write(SOFT_RST, status | 0x00000010); + + if (int_st & INT_A_OUT) + fsi_data_push(&master->fsia); + if (int_st & INT_B_OUT) + fsi_data_push(&master->fsib); + + fsi_master_write(INT_ST, 0x0000000); + + return IRQ_HANDLED; +} + +/************************************************************************ + + + dai ops + + +************************************************************************/ +static int fsi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + const char *msg; + u32 flags = fsi_get_info_flags(fsi); + u32 fmt; + u32 reg; + u32 data; + int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int is_master; + int ret = 0; + + clk_enable(master->clk); + + /* CKG1 */ + data = is_play ? (1 << 0) : (1 << 4); + is_master = fsi_is_master_mode(fsi, is_play); + if (is_master) + fsi_reg_mask_set(fsi, CKG1, data, data); + else + fsi_reg_mask_set(fsi, CKG1, data, 0); + + /* clock inversion (CKG2) */ + data = 0; + switch (SH_FSI_INVERSION_MASK & flags) { + case SH_FSI_LRM_INV: + data = 1 << 12; + break; + case SH_FSI_BRM_INV: + data = 1 << 8; + break; + case SH_FSI_LRS_INV: + data = 1 << 4; + break; + case SH_FSI_BRS_INV: + data = 1 << 0; + break; + } + fsi_reg_write(fsi, CKG2, data); + + /* do fmt, di fmt */ + data = 0; + reg = is_play ? DO_FMT : DI_FMT; + fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); + switch (fmt) { + case SH_FSI_FMT_MONO: + msg = "MONO"; + data = CR_FMT(CR_MONO); + fsi->chan = 1; + break; + case SH_FSI_FMT_MONO_DELAY: + msg = "MONO Delay"; + data = CR_FMT(CR_MONO_D); + fsi->chan = 1; + break; + case SH_FSI_FMT_PCM: + msg = "PCM"; + data = CR_FMT(CR_PCM); + fsi->chan = 2; + break; + case SH_FSI_FMT_I2S: + msg = "I2S"; + data = CR_FMT(CR_I2S); + fsi->chan = 2; + break; + case SH_FSI_FMT_TDM: + msg = "TDM"; + data = CR_FMT(CR_TDM) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + case SH_FSI_FMT_TDM_DELAY: + msg = "TDM Delay"; + data = CR_FMT(CR_TDM_D) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + default: + dev_err(dai->dev, "unknown format.\n"); + return -EINVAL; + } + + switch (fsi->chan) { + case 1: + fsi->fifo_max = 256; + break; + case 2: + fsi->fifo_max = 128; + break; + case 3: + case 4: + fsi->fifo_max = 64; + break; + case 5: + case 6: + case 7: + case 8: + fsi->fifo_max = 32; + break; + default: + dev_err(dai->dev, "channel size error.\n"); + return -EINVAL; + } + + fsi_reg_write(fsi, reg, data); + dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n", + msg, fsi->chan, fsi->dma_chan); + + /* + * clear clk reset if master mode + */ + if (is_master) + fsi_clk_ctrl(fsi, 1); + + /* irq setting */ + fsi_irq_init(fsi, is_play); + + return ret; +} + +static void fsi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + fsi_irq_disable(fsi, is_play); + fsi_clk_ctrl(fsi, 0); + + clk_disable(master->clk); +} + +static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int ret = 0; + + /* capture not supported */ + if (!is_play) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + fsi_stream_push(fsi, substream, + frames_to_bytes(runtime, runtime->buffer_size), + frames_to_bytes(runtime, runtime->period_size)); + ret = fsi_data_push(fsi); + break; + case SNDRV_PCM_TRIGGER_STOP: + fsi_irq_disable(fsi, is_play); + fsi_stream_pop(fsi); + break; + } + + return ret; +} + +static struct snd_soc_dai_ops fsi_dai_ops = { + .startup = fsi_dai_startup, + .shutdown = fsi_dai_shutdown, + .trigger = fsi_dai_trigger, +}; + +/************************************************************************ + + + pcm ops + + +************************************************************************/ +static struct snd_pcm_hardware fsi_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = FSI_FMTS, + .rates = FSI_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int fsi_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int fsi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int fsi_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + long location; + + location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + if (location < 0) + location = 0; + + return bytes_to_frames(runtime, location); +} + +static struct snd_pcm_ops fsi_pcm_ops = { + .open = fsi_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsi_hw_params, + .hw_free = fsi_hw_free, + .pointer = fsi_pointer, +}; + +/************************************************************************ + + + snd_soc_platform + + +************************************************************************/ +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static void fsi_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int fsi_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* + * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + return snd_pcm_lib_preallocate_pages_for_all( + pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +/************************************************************************ + + + alsa struct + + +************************************************************************/ +struct snd_soc_dai fsi_soc_dai[] = { + { + .name = "FSIA", + .id = 0, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + }, + { + .name = "FSIB", + .id = 1, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(fsi_soc_dai); + +struct snd_soc_platform fsi_soc_platform = { + .name = "fsi-pcm", + .pcm_ops = &fsi_pcm_ops, + .pcm_new = fsi_pcm_new, + .pcm_free = fsi_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsi_soc_platform); + +/************************************************************************ + + + platform function + + +************************************************************************/ +static int fsi_probe(struct platform_device *pdev) +{ + struct resource *res; + char clk_name[8]; + unsigned int irq; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + irq = platform_get_irq(pdev, 0); + if (!res || !irq) { + dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); + ret = -ENODEV; + goto exit; + } + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) { + dev_err(&pdev->dev, "Could not allocate master\n"); + ret = -ENOMEM; + goto exit; + } + + master->base = ioremap_nocache(res->start, resource_size(res)); + if (!master->base) { + ret = -ENXIO; + dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); + goto exit_kfree; + } + + master->irq = irq; + master->info = pdev->dev.platform_data; + master->fsia.base = master->base; + master->fsib.base = master->base + 0x40; + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; + + ret = fsi_get_dma_chan(); + if (ret < 0) { + dev_err(&pdev->dev, "cannot get dma api\n"); + goto exit_iounmap; + } + + /* FSI is based on SPU mstp */ + snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); + master->clk = clk_get(NULL, clk_name); + if (IS_ERR(master->clk)) { + dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); + ret = -EIO; + goto exit_free_dma; + } + + fsi_soc_dai[0].dev = &pdev->dev; + fsi_soc_dai[1].dev = &pdev->dev; + + fsi_soft_all_reset(); + + ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); + if (ret) { + dev_err(&pdev->dev, "irq request err\n"); + goto exit_free_dma; + } + + ret = snd_soc_register_platform(&fsi_soc_platform); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd soc register\n"); + goto exit_free_irq; + } + + return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + +exit_free_irq: + free_irq(irq, master); +exit_free_dma: + fsi_free_dma_chan(); +exit_iounmap: + iounmap(master->base); +exit_kfree: + kfree(master); + master = NULL; +exit: + return ret; +} + +static int fsi_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&fsi_soc_platform); + + clk_put(master->clk); + + fsi_free_dma_chan(); + + free_irq(master->irq, master); + + iounmap(master->base); + kfree(master); + master = NULL; + return 0; +} + +static struct platform_driver fsi_driver = { + .driver = { + .name = "sh_fsi", + }, + .probe = fsi_probe, + .remove = fsi_remove, +}; + +static int __init fsi_mobile_init(void) +{ + return platform_driver_register(&fsi_driver); +} + +static void __exit fsi_mobile_exit(void) +{ + platform_driver_unregister(&fsi_driver); +} +module_init(fsi_mobile_init); +module_exit(fsi_mobile_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

4 5

[alsa-devel] [PATCH 1/3] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This driver is very simple. It support playback only now. It is tested by ms7724se board. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- include/sound/fsi.h | 83 ++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/Makefile | 1 + sound/soc/sh/fsi.c | 1001 +++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1090 insertions(+), 0 deletions(-) create mode 100644 include/sound/fsi.h create mode 100644 sound/soc/sh/fsi.c diff --git a/include/sound/fsi.h b/include/sound/fsi.h new file mode 100644 index 0000000..c022736 --- /dev/null +++ b/include/sound/fsi.h @@ -0,0 +1,83 @@ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.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. + */ + +/* flags format + + * 0xABCDEEFF + * + * A: channel size for TDM (input) + * B: channel size for TDM (ooutput) + * C: inversion + * D: mode + * E: input format + * F: output format + */ + +#include <linux/clk.h> +#include <sound/soc.h> + +/* TDM channel */ +#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) +#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) + +#define SH_FSI_CH_IMASK 0xF0000000 +#define SH_FSI_CH_OMASK 0x0F000000 +#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) +#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) + +/* clock inversion */ +#define SH_FSI_INVERSION_MASK 0x00F00000 +#define SH_FSI_LRM_INV (1 << 20) +#define SH_FSI_BRM_INV (1 << 21) +#define SH_FSI_LRS_INV (1 << 22) +#define SH_FSI_BRS_INV (1 << 23) + +/* mode */ +#define SH_FSI_MODE_MASK 0x000F0000 +#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ +#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ + +/* DI format */ +#define SH_FSI_FMT_MASK 0x000000FF +#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) +#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) +#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) +#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) + +#define SH_FSI_FMT_MONO (1 << 0) +#define SH_FSI_FMT_MONO_DELAY (1 << 1) +#define SH_FSI_FMT_PCM (1 << 2) +#define SH_FSI_FMT_I2S (1 << 3) +#define SH_FSI_FMT_TDM (1 << 4) +#define SH_FSI_FMT_TDM_DELAY (1 << 5) + +#define SH_FSI_IFMT_TDM_CH(x) \ + (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) +#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ + (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) + +#define SH_FSI_OFMT_TDM_CH(x) \ + (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) +#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ + (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +struct sh_fsi_platform_info { + unsigned long porta_flags; + unsigned long portb_flags; +}; + +extern struct snd_soc_dai fsi_soc_dai[2]; +extern struct snd_soc_platform fsi_soc_platform; + +#endif /* __SOUND_FSI_H */ diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 54bd604..afb0e60 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -20,6 +20,11 @@ config SND_SOC_SH4_HAC config SND_SOC_SH4_SSI tristate +config SND_SOC_SH4_FSI + tristate "SH4 FSI support" + depends on CPU_SUBTYPE_SH7724 + help + This option enables FSI sound support ## diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a8e8ab8..dfa4949 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -7,6 +7,7 @@ snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o +obj-$(CONFIG_SND_SOC_SH4_FSI) += fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c new file mode 100644 index 0000000..017f170 --- /dev/null +++ b/sound/soc/sh/fsi.c @@ -0,0 +1,1001 @@ +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ssi.c + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/list.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/fsi.h> +#include <asm/atomic.h> +#include <asm/dma.h> +#include <asm/dma-sh.h> + +#define DO_FMT 0x0000 +#define DOFF_CTL 0x0004 +#define DOFF_ST 0x0008 +#define DI_FMT 0x000C +#define DIFF_CTL 0x0010 +#define DIFF_ST 0x0014 +#define CKG1 0x0018 +#define CKG2 0x001C +#define DIDT 0x0020 +#define DODT 0x0024 +#define MUTE_ST 0x0028 +#define REG_END MUTE_ST + +#define INT_ST 0x0200 +#define IEMSK 0x0204 +#define IMSK 0x0208 +#define MUTE 0x020C +#define CLK_RST 0x0210 +#define SOFT_RST 0x0214 +#define MREG_START INT_ST +#define MREG_END SOFT_RST + +/* DO_FMT */ +/* DI_FMT */ +#define CR_FMT(param) ((param) << 4) +# define CR_MONO 0x0 +# define CR_MONO_D 0x1 +# define CR_PCM 0x2 +# define CR_I2S 0x3 +# define CR_TDM 0x4 +# define CR_TDM_D 0x5 + +/* DOFF_CTL */ +/* DIFF_CTL */ +#define IRQ_HALF 0x00100000 +#define FIFO_CLR 0x00000001 + +/* DOFF_ST */ +#define ERR_OVER 0x00000010 +#define ERR_UNDER 0x00000001 + +/* CLK_RST */ +#define B_CLK 0x00000010 +#define A_CLK 0x00000001 + +/* INT_ST */ +#define INT_B_IN (1 << 12) +#define INT_B_OUT (1 << 8) +#define INT_A_IN (1 << 4) +#define INT_A_OUT (1 << 0) + +#define FSI_RATES SNDRV_PCM_RATE_8000_96000 + +#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/************************************************************************ + + + struct + + +************************************************************************/ +struct fsi_priv { + void __iomem *base; + struct snd_pcm_substream *substream; + + int fifo_max; + int chan; + int dma_chan; + + int byte_offset; + int period_len; + int buffer_len; + int periods; +}; + +struct fsi_master { + void __iomem *base; + atomic_t user; + int irq; + struct clk *clk; + struct fsi_priv fsia; + struct fsi_priv fsib; + struct sh_fsi_platform_info *info; +}; + +struct fsi_master *master; + +/************************************************************************ + + + basic read write function + + +************************************************************************/ +static int __fsi_reg_write(u32 reg, u32 data) +{ + /* valid data area is 24bit */ + data &= 0x00ffffff; + + return ctrl_outl(data, reg); +} + +static u32 __fsi_reg_read(u32 reg) +{ + return ctrl_inl(reg); +} + +static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +{ + u32 val = __fsi_reg_read(reg); + + val &= ~mask; + val |= data & mask; + + return __fsi_reg_write(reg, val); +} + +static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_write((u32)(fsi->base + reg), data); +} + +static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) +{ + if (reg > REG_END) + return 0; + + return __fsi_reg_read((u32)(fsi->base + reg)); +} + +static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); +} + +static int fsi_master_write(u32 reg, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_write((u32)(master->base + reg), data); +} + +static u32 fsi_master_read(u32 reg) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return 0; + + return __fsi_reg_read((u32)(master->base + reg)); +} + +static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); +} + +/************************************************************************ + + + basic function + + +************************************************************************/ +static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd; + struct fsi_priv *fsi = NULL; + + if (!substream || !master) + return NULL; + + rtd = substream->private_data; + switch (rtd->dai->cpu_dai->id) { + case 0: + fsi = &master->fsia; + break; + case 1: + fsi = &master->fsib; + break; + } + + return fsi; +} + +static int fsi_is_port_a(struct fsi_priv *fsi) +{ + /* return + * 1 : port a + * 0 : port b + */ + + if (fsi == &master->fsia) + return 1; + + return 0; +} + +static u32 fsi_get_info_flags(struct fsi_priv *fsi) +{ + int is_porta = fsi_is_port_a(fsi); + + return is_porta ? master->info->porta_flags : + master->info->portb_flags; +} + +static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) +{ + u32 mode; + u32 flags = fsi_get_info_flags(fsi); + + mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; + + /* return + * 1 : master mode + * 0 : slave mode + */ + + return (mode & flags) != mode; +} + +static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play) +{ + int is_porta = fsi_is_port_a(fsi); + u32 data; + + if (is_porta) + data = is_play ? (1 << 0) : (1 << 4); + else + data = is_play ? (1 << 8) : (1 << 12); + + return data; +} + +static void fsi_stream_push(struct fsi_priv *fsi, + struct snd_pcm_substream *substream, + u32 buffer_len, + u32 period_len) +{ + fsi->substream = substream; + fsi->buffer_len = buffer_len; + fsi->period_len = period_len; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static void fsi_stream_pop(struct fsi_priv *fsi) +{ + fsi->substream = NULL; + fsi->buffer_len = 0; + fsi->period_len = 0; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) +{ + u32 status; + u32 reg = is_play ? DOFF_ST : DIFF_ST; + int residue; + + status = fsi_reg_read(fsi, reg); + residue = 0x1ff & (status >> 8); + residue *= fsi->chan; + + return residue; +} + +static int fsi_get_residue(struct fsi_priv *fsi, int is_play) +{ + int residue; + int width; + struct snd_pcm_runtime *runtime; + + runtime = fsi->substream->runtime; + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + if (2 == width) + residue = fsi_get_fifo_residue(fsi, is_play); + else + residue = get_dma_residue(fsi->dma_chan); + + return residue; +} + +/************************************************************************ + + + basic dma function + + +************************************************************************/ +#define PORTA_DMA 0 +#define PORTB_DMA 1 + +static int fsi_get_dma_chan(void) +{ + if (0 != request_dma(PORTA_DMA, "fsia")) + return -EIO; + + if (0 != request_dma(PORTB_DMA, "fsib")) { + free_dma(PORTA_DMA); + return -EIO; + } + + master->fsia.dma_chan = PORTA_DMA; + master->fsib.dma_chan = PORTB_DMA; + + return 0; +} + +static void fsi_free_dma_chan(void) +{ + dma_wait_for_completion(PORTA_DMA); + dma_wait_for_completion(PORTB_DMA); + free_dma(PORTA_DMA); + free_dma(PORTB_DMA); + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; +} + +/************************************************************************ + + + ctrl function + + +************************************************************************/ +static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, data); + fsi_master_mask_set(IEMSK, data, data); +} + +static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, 0); + fsi_master_mask_set(IEMSK, data, 0); +} + +static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) +{ + u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + + if (enable) + fsi_master_mask_set(CLK_RST, val, val); + else + fsi_master_mask_set(CLK_RST, val, 0); +} + +static void fsi_irq_init(struct fsi_priv *fsi, int is_play) +{ + u32 data; + u32 ctrl; + + data = fsi_port_ab_io_bit(fsi, is_play); + ctrl = is_play ? DOFF_CTL : DIFF_CTL; + + /* set IMSK */ + fsi_irq_disable(fsi, is_play); + + /* set interrupt generation factor */ + fsi_reg_write(fsi, ctrl, IRQ_HALF); + + /* clear FIFO */ + fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); + + /* clear interrupt factor */ + fsi_master_mask_set(INT_ST, data, 0); +} + +static void fsi_soft_all_reset(void) +{ + u32 status = fsi_master_read(SOFT_RST); + + /* port AB reset */ + status &= 0x000000ff; + fsi_master_write(SOFT_RST, status); + mdelay(10); + + /* soft reset */ + status &= 0x000000f0; + fsi_master_write(SOFT_RST, status); + status |= 0x00000001; + fsi_master_write(SOFT_RST, status); + mdelay(10); +} + +static void fsi_16data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u16 *dma_start; + u32 snd; + int i; + + /* get dma start position for FSI */ + dma_start = (u16 *)runtime->dma_area; + dma_start += fsi->byte_offset / 2; + + /* + * soft dma + * FSI can not use DMA when 16bpp + */ + for (i = 0; i < send; i++) { + snd = (u32)dma_start[i]; + fsi_reg_write(fsi, DODT, snd << 8); + } +} + +static void fsi_32data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u32 *dma_start; + + /* get dma start position for FSI */ + dma_start = (u32 *)runtime->dma_area; + dma_start += fsi->byte_offset / 4; + + dma_wait_for_completion(fsi->dma_chan); + dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); + dma_write(fsi->dma_chan, (u32)dma_start, + (u32)(fsi->base + DODT), send * 4); +} + +/* playback interrupt */ +static int fsi_data_push(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int send; + int fifo_free; + int width; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get send size for alsa */ + send = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get FIFO free size */ + fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1); + + /* size check */ + if (fifo_free < send) + send = fifo_free; + + if (2 == width) + fsi_16data_push(fsi, runtime, send); + else if (4 == width) + fsi_32data_push(fsi, runtime, send); + else + return -EINVAL; + + fsi->byte_offset += send * width; + + fsi_irq_enable(fsi, 1); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + +static irqreturn_t fsi_interrupt(int irq, void *data) +{ + u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(INT_ST); + + /* clear irq status */ + fsi_master_write(SOFT_RST, status); + fsi_master_write(SOFT_RST, status | 0x00000010); + + if (int_st & INT_A_OUT) + fsi_data_push(&master->fsia); + if (int_st & INT_B_OUT) + fsi_data_push(&master->fsib); + + fsi_master_write(INT_ST, 0x0000000); + + return IRQ_HANDLED; +} + +/************************************************************************ + + + dai ops + + +************************************************************************/ +static int fsi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + const char *msg; + u32 flags = fsi_get_info_flags(fsi); + u32 fmt; + u32 reg; + u32 data; + int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int is_master; + int ret = 0; + + if (1 == atomic_inc_return(&master->user)) + clk_enable(master->clk); + + /* CKG1 */ + data = is_play ? (1 << 0) : (1 << 4); + is_master = fsi_is_master_mode(fsi, is_play); + if (is_master) + fsi_reg_mask_set(fsi, CKG1, data, data); + else + fsi_reg_mask_set(fsi, CKG1, data, 0); + + /* clock inversion (CKG2) */ + data = 0; + switch (SH_FSI_INVERSION_MASK & flags) { + case SH_FSI_LRM_INV: + data = 1 << 12; + break; + case SH_FSI_BRM_INV: + data = 1 << 8; + break; + case SH_FSI_LRS_INV: + data = 1 << 4; + break; + case SH_FSI_BRS_INV: + data = 1 << 0; + break; + } + fsi_reg_write(fsi, CKG2, data); + + /* do fmt, di fmt */ + data = 0; + reg = is_play ? DO_FMT : DI_FMT; + fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); + switch (fmt) { + case SH_FSI_FMT_MONO: + msg = "MONO"; + data = CR_FMT(CR_MONO); + fsi->chan = 1; + break; + case SH_FSI_FMT_MONO_DELAY: + msg = "MONO Delay"; + data = CR_FMT(CR_MONO_D); + fsi->chan = 1; + break; + case SH_FSI_FMT_PCM: + msg = "PCM"; + data = CR_FMT(CR_PCM); + fsi->chan = 2; + break; + case SH_FSI_FMT_I2S: + msg = "I2S"; + data = CR_FMT(CR_I2S); + fsi->chan = 2; + break; + case SH_FSI_FMT_TDM: + msg = "TDM"; + data = CR_FMT(CR_TDM) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + case SH_FSI_FMT_TDM_DELAY: + msg = "TDM Delay"; + data = CR_FMT(CR_TDM_D) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + default: + dev_err(dai->dev, "unknown format.\n"); + return -EINVAL; + } + + switch (fsi->chan) { + case 1: + fsi->fifo_max = 256; + break; + case 2: + fsi->fifo_max = 128; + break; + case 3: + case 4: + fsi->fifo_max = 64; + break; + case 5: + case 6: + case 7: + case 8: + fsi->fifo_max = 32; + break; + default: + dev_err(dai->dev, "channel size error.\n"); + return -EINVAL; + } + + fsi_reg_write(fsi, reg, data); + dev_info(dai->dev, "use %s format (%d channel) use %d DMAC\n", + msg, fsi->chan, fsi->dma_chan); + + /* + * clear clk reset if master mode + */ + if (is_master) + fsi_clk_ctrl(fsi, 1); + + /* irq setting */ + fsi_irq_init(fsi, is_play); + + return ret; +} + +static void fsi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + fsi_irq_disable(fsi, is_play); + fsi_clk_ctrl(fsi, 0); + + if (atomic_dec_and_test(&master->user)) + clk_disable(master->clk); +} + +static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int ret = 0; + + /* capture not supported */ + if (!is_play) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + fsi_stream_push(fsi, substream, + frames_to_bytes(runtime, runtime->buffer_size), + frames_to_bytes(runtime, runtime->period_size)); + ret = fsi_data_push(fsi); + break; + case SNDRV_PCM_TRIGGER_STOP: + fsi_irq_disable(fsi, is_play); + fsi_stream_pop(fsi); + break; + } + + return ret; +} + +static struct snd_soc_dai_ops fsi_dai_ops = { + .startup = fsi_dai_startup, + .shutdown = fsi_dai_shutdown, + .trigger = fsi_dai_trigger, +}; + +/************************************************************************ + + + pcm ops + + +************************************************************************/ +static struct snd_pcm_hardware fsi_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = FSI_FMTS, + .rates = FSI_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int fsi_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int fsi_pcm_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static int fsi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int fsi_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + long location; + + location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + if (location < 0) + location = 0; + + return bytes_to_frames(runtime, location); +} + +static struct snd_pcm_ops fsi_pcm_ops = { + .open = fsi_pcm_open, + .close = fsi_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsi_hw_params, + .hw_free = fsi_hw_free, + .pointer = fsi_pointer, +}; + +/************************************************************************ + + + snd_soc_platform + + +************************************************************************/ +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static void fsi_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int fsi_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* + * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + return snd_pcm_lib_preallocate_pages_for_all( + pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +/************************************************************************ + + + alsa struct + + +************************************************************************/ +struct snd_soc_dai fsi_soc_dai[] = { + { + .name = "FSI0", + .id = 0, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + } +}; +EXPORT_SYMBOL_GPL(fsi_soc_dai); + +struct snd_soc_platform fsi_soc_platform = { + .name = "fsi-pcm", + .pcm_ops = &fsi_pcm_ops, + .pcm_new = fsi_pcm_new, + .pcm_free = fsi_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsi_soc_platform); + +/************************************************************************ + + + platform function + + +************************************************************************/ +static int fsi_probe(struct platform_device *pdev) +{ + struct resource *res; + char clk_name[8]; + unsigned int irq; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + irq = platform_get_irq(pdev, 0); + if (!res || !irq) { + dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); + ret = -ENODEV; + goto exit; + } + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) { + dev_err(&pdev->dev, "Could not allocate master\n"); + ret = -ENOMEM; + goto exit; + } + + master->base = ioremap_nocache(res->start, resource_size(res)); + if (!master->base) { + ret = -ENXIO; + dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); + goto exit_kfree; + } + + master->irq = irq; + master->info = pdev->dev.platform_data; + master->fsia.base = master->base; + master->fsib.base = master->base + 0x40; + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; + + ret = fsi_get_dma_chan(); + if (ret < 0) + goto exit_iounmap; + + /* FSI is based on SPU mstp */ + snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); + master->clk = clk_get(&pdev->dev, clk_name); + if (IS_ERR(master->clk)) { + dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); + ret = -EIO; + goto exit_free_dma; + } + + /* FSIA */ + fsi_soc_dai[0].dev = &pdev->dev; + + master->user = ATOMIC_INIT(0); + + fsi_soft_all_reset(); + + ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); + if (ret) { + dev_err(&pdev->dev, "irq request err\n"); + goto exit_free_dma; + } + + ret = snd_soc_register_platform(&fsi_soc_platform); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd soc register\n"); + goto exit_free_irq; + } + + return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + +exit_free_irq: + free_irq(irq, master); +exit_free_dma: + fsi_free_dma_chan(); +exit_iounmap: + iounmap(master->base); +exit_kfree: + kfree(master); + master = NULL; +exit: + return ret; +} + +static int fsi_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&fsi_soc_platform); + + clk_put(master->clk); + + fsi_free_dma_chan(); + + free_irq(master->irq, master); + + iounmap(master->base); + kfree(master); + master = NULL; + return 0; +} + +static struct platform_driver fsi_driver = { + .driver = { + .name = "sh_fsi", + }, + .probe = fsi_probe, + .remove = fsi_remove, +}; + +static int __init fsi_mobile_init(void) +{ + return platform_driver_register(&fsi_driver); +} + +static void __exit fsi_mobile_exit(void) +{ + platform_driver_unregister(&fsi_driver); +} +module_init(fsi_mobile_init); +module_exit(fsi_mobile_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

4 6

[alsa-devel] [PATCH v3 2/4] Add ak4642 codec support
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v3 I'm so sorry. I took miss. v2 patch still use ak464x name. This patch modify name o ak464x -> ak4642 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 516 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 25 +++ 4 files changed, 547 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..ee794f1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -74,6 +75,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..c8c567e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..1b439b4 --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,516 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#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 <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 "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4642_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..28c267a --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,25 @@ +/* + * ak4642.h -- AK4642 Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * 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. + */ + +#ifndef _AK4642_H +#define _AK4642_H + +struct ak4642_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

1 0

[alsa-devel] [PATCH v2 0/3] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 20 Aug '09

20 Aug '09

These patches are FSI driver for ALSA v2 1) Add SuperH FSI driver support for ALSA 2) Add ak464x codec support 3) Add FSI-AK4642 sound support for SuperH 4) sh: add FSI driver support for ms7724se Best regards -- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 3/3] Add FSI-AK464x sound support for ms7724se
by Kuninori Morimoto 20 Aug '09

20 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- arch/sh/boards/Kconfig | 9 +++ arch/sh/boards/mach-se/7724/Makefile | 3 +- arch/sh/boards/mach-se/7724/fsi-ak464x.c | 90 ++++++++++++++++++++++++++++++ arch/sh/boards/mach-se/7724/setup.c | 90 ++++++++++++++++++++++++++++++ 4 files changed, 191 insertions(+), 1 deletions(-) create mode 100644 arch/sh/boards/mach-se/7724/fsi-ak464x.c diff --git a/arch/sh/boards/Kconfig b/arch/sh/boards/Kconfig index db04c85..44f4c8b 100644 --- a/arch/sh/boards/Kconfig +++ b/arch/sh/boards/Kconfig @@ -55,6 +55,15 @@ config SH_7724_SOLUTION_ENGINE Select 7724 SolutionEngine if configuring for a Hitachi SH7724 evaluation board. +config SND_SH4_FSI_AK464X + bool "FSI-AK464X sound support" + depends on SND_SOC_SH4_FSI + depends on SH_7724_SOLUTION_ENGINE + select SND_SOC_AK464X + help + This option enables generic sound support for the + FSI - AK464x unit of the SH4. + config SH_7751_SOLUTION_ENGINE bool "SolutionEngine7751" select SOLUTION_ENGINE diff --git a/arch/sh/boards/mach-se/7724/Makefile b/arch/sh/boards/mach-se/7724/Makefile index 349cbd6..e0f5366 100644 --- a/arch/sh/boards/mach-se/7724/Makefile +++ b/arch/sh/boards/mach-se/7724/Makefile @@ -7,4 +7,5 @@ # # -obj-y := setup.o irq.o \ No newline at end of file +obj-y := setup.o irq.o +obj-$(CONFIG_SND_SH4_FSI_AK464X) += fsi-ak464x.o diff --git a/arch/sh/boards/mach-se/7724/fsi-ak464x.c b/arch/sh/boards/mach-se/7724/fsi-ak464x.c new file mode 100644 index 0000000..bab118e --- /dev/null +++ b/arch/sh/boards/mach-se/7724/fsi-ak464x.c @@ -0,0 +1,90 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/io.h> + +#include <sound/fsi.h> +#include <../sound/soc/codecs/ak464x.h> + +static int machine_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK464x", + .stream_name = "AK464x", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak464x_dai, + .init = machine_init, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "SH4 I2S (FSI)", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +struct ak464x_setup_data ak464x_setup = { + .i2c_bus = 0, + .i2c_address = 0x12, /* 0x13 */ +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak464x, + .codec_data = &ak464x_setup, +}; + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak464x_init(void) +{ + int ret; + + ret = -ENOMEM; + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak464x_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak464x_init); +module_exit(fsi_ak464x_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK464x sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..0099a79 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,11 +22,13 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> #include <asm/clock.h> #include <asm/sh_keysc.h> +#include <asm/dma-sh.h> #include <cpu/sh7724.h> #include <mach-se/mach/se7724.h> @@ -246,6 +248,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +412,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +480,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +622,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

3 9

[alsa-devel] unable to do simultaneous playback and recording
by prashant chopade 20 Aug '09

20 Aug '09

Hello, We using linux-2.6.27 on target board-AT91SAM9263EK. We are using ALSA library utilities (aplay & arecord). But by using these utilities we can’t do simultaneous playback and recording. We performed following activity to come to this conclusion: //At the background we were recording #arecord –f cd record_test_file & //At the forefront we were playing pre-recorded audio file #aplay –f cd pre_recorded_file Our observation is that, recording stops when start playing file. Kindly need your guidance for simultaneous playback and recording of an audio file. Appreciates your help if you provide sample test programs (using ALSA API) for recording. Thanks & Regards, Prashant C. Love Cricket? Check out live scores, photos, video highlights and more. Click here http://cricket.yahoo.com

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Re: [alsa-devel] unable to do simultaneous playback and recording
by prashant chopade 20 Aug '09

20 Aug '09

Hello, My understandings: >Define your meaning of "simultaneous playback and recording" more >exactly. I mean: Playing audio file stored on disk on speaker and at the same time recording audio from MIC and storing it on disk. I don,t want to do multiplayback, I mean i don,t want to play two or more audio files at a time. So you are saying that my driver(ATMEL-AT91SAM9263-AC97c) is not supporting the full duplex mode? Thanks & Regards, Prashant C. >prashant chopade wrote: > > Hello, > > As per discussion, I know that ALSA driver does not have support for > simultaneous playback and recording. Huh? The full-duplex is of course supported. Your example shown in the thread doesn't indicate the multi playbacks or so. > We are using AD1981B codec, so our hardware has a support for > simultaneous playback and recording of an audio. I think you are confused. > So what exactly I have to do to get a support for simultaneous > playback and recording? > Shall I have to use sound server demons? > > Is it really possible to do simultaneous playback and recording in > linux? Define your meaning of "simultaneous playback and recording" more exactly. If it's about full-duplex, the answer is definitely yes. If it's multi-streams, the answer is yes, but there are choices, e.g. dmix/dsnoop plugin or a sound server. Takashi > prashant chopade wrote: > > > > > > > > > > > > Hello, > > > > We using linux-2.6.27 on target board-AT91SAM9263EK. > > We are using ALSA library utilities (aplay & > > arecord). But by using these utilities we can’t do simultaneous playback and > > recording. > > > > > > > > We performed following activity to come to this conclusion: > > > > > > > > //At the background we were recording > > > > #arecord –f cd record_test_file & > > > > > > > > //At the forefront we were playing pre-recorded audio file > > > > #aplay –f cd pre_recorded_file > > > > > > > > Our observation is that, recording stops when start playing > > file. > > In the API side, the above should work fine. It must be the driver > that doesn't work with full-duplex. > > > Takashi Yahoo! recommends that you upgrade to the new and safer Internet Explorer 8. http://downloads.yahoo.com/in/internetexplorer/

2 1

Re: [alsa-devel] Nvidia MSI Geforce 9500
by Clemens Ladisch 20 Aug '09

20 Aug '09

Shane W wrote: > think this might belong on the devel list. After some > searching, looks like my Geforce 9500 card is not > recognized by the hda_intel driver. I tried adding: > { PCI_DEVICE(0x10de, 0x0640), .driver_data = AZX_DRIVER_NVIDIA }, > >> lspci -vvv: >> 01:00.0 VGA compatible controller: nVidia Corporation G96 [GeForce 9500 GT] >> (rev a1) (prog-if 00 [VGA controller]) This PCI ID is for the video part; the HDMI interface should have a second entry in the lspci output. HTH Clemens

1 0

[alsa-devel] Using 8bit mono at 8000Hz with arecord and aplay
by Shilpa Kedar Walvekar 20 Aug '09

20 Aug '09

Hello, We tried to record and play the recorded file using arecord and aplay as follows: arecord -f cd /home/test.wav aplay -f cd /home/test.wav The file gets recorded and played correctly. But if I try to record file of format S8 or U8, I get error: # arecord -f U8 /home/test_alsa_record_2 Recording WAVE '/home/test_alsa_record_2' : Unsigned 8 bit, Rate 8000 Hz, Mono arecord: set_params:979: Sample format non available # arecord -f S8 /home/alsa_record_2 Recording WAVE '/home/alsa_record_2' : Signed 8 bit, Rate 8000 Hz, Mono arecord: set_params:979: Sample format non available Is there any other way to set the format as S8 or U8? Or only cd/cdr/dat formats can be played using alsa? Regards, --Shilpa.

3 4

[alsa-devel] Support for ALC662 on FUJITSU SIEMENS AMILO Li3710
by Mads Kiilerich 20 Aug '09

20 Aug '09

On a friends FUJITSU SIEMENS AMILO Li3710 10601011427 sound didn't work with kernel-PAE-2.6.29.6-213.fc11.i686 Out of the box we get no sound and HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: Unknown model for ALC662, trying auto-probe from BIOS... ALSA sound/pci/hda/hda_codec.c:3507: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3511: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3515: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3516: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3524: inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 But with /etc/modprobe.d/local.conf (from http://ubuntuforums.org/archive/index.php/t-940689.html) options snd-hda-intel model=3stack-dig options snd-hda-intel single_cmd=1 options snd-hda-intel enable_msi=1 we get sound and HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 32 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA sound/pci/hda/hda_codec.c:2698: hda_codec: model '3stack-dig' is selected I would like to have this fixed so Linux sound works on my friends machine without manual configuration. After having read some documentation I assume that single_cmd and enable_msi perhaps shouldn't be needed. I will test that next time I get access to that machine. HAL reports the sound card as /org/freedesktop/Hal/devices/pci_8086_293e, so should this patch be sufficient? diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1563f4..0385353 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16699,6 +16699,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x8086, 0x293e, "FUJITSU SIEMENS AMILO Li3710", + ALC662_3ST_2ch_DIG), {} }; Untested and hard to test because I don't have direct access to the machine. This issue was also reported on https://bugzilla.redhat.com/show_bug.cgi?id=487889 Please CC me. /Mads

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Re: [alsa-devel] [PATCH] ALSA: hda: partially revert ALC268 quirk collapse
by Takashi Iwai 19 Aug '09

19 Aug '09

At Sat, 15 Aug 2009 20:53:38 +0400, Alexey Dobriyan wrote: > > On Sat, Aug 15, 2009 at 05:41:04PM +0200, Takashi Iwai wrote: > > At Sat, 15 Aug 2009 16:38:46 +0400, > > Alexey Dobriyan wrote: > > > > > > Fix alsamixer inability to start: > > > > > > ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Master Playback Switch',0,3,0) appears twice or more > > > > > > ALC268 codec shipped with Toshiba Satellite X200 has ids 1179:ff00 and > > > began to match quirk table. > > > > > > If it doesn't match and auto-probe from BIOS is used, alsamixer works: > > > > > > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > > > > > > 8871e5b91518a47284b6bc2603b44dbc79c85446 is first bad commit > > > commit 8871e5b91518a47284b6bc2603b44dbc79c85446 > > > Author: Takashi Iwai <tiwai(a)suse.de> > > > Date: Tue Jun 2 01:02:50 2009 +0200 > > > > > > ALSA: hda - Reorder and clean-up ALC268 quirk table > > > > > > Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some > > > entries using SND_PCI_QUIRK_MASK(). > > > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > :040000 040000 2497bb6fc5305acf3191b8e9b82956ec0ea1aee9 6f1d9ff9fed20550a1b2b398c7d2d5fec705fcd9 M sound > > > > > > Signed-off-by: Alexey Dobriyan <adobriyan(a)gmail.com> > > > > Well, it's no right fix. The symptom implies that something wrong > > with the current model=toshiba, so fixing the quirk is the way to go > > at this moment. > > > > Could you give alsa-info.sh output (run with --no-upload option) > > as vanilla kernel? Run with both model=auto and without model option > > and attach both results to compare. The former means actually the > > condition as your patch does. > > Here is diff for quick look: Thanks. The problem is that this laptop has two ALC268 codecs, and these conflict. Could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fea9767..6f683e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12521,8 +12521,6 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), - SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", - ALC268_TOSHIBA), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), @@ -12530,6 +12528,15 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { {} }; +/* Toshiba laptops have no unique PCI SSID but only codec SSID */ +static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO), + SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO), + SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", + ALC268_TOSHIBA), + {} +}; + static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, @@ -12696,6 +12703,10 @@ static int patch_alc268(struct hda_codec *codec) alc268_models, alc268_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) + board_config = snd_hda_check_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { printk(KERN_INFO "hda_codec: Unknown model for %s, " "trying auto-probe from BIOS...\n", codec->chip_name);

2 2

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 19 Aug '09

19 Aug '09

The following changes since commit d3c9e9a1390f8a34da8b69e09fa1afa90f5067f4: Mark Brown (1): ASoC: Implement TDM configuration for WM8993 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 Joonyoung Shim (1): ASoC: Fix data format configuration for S3C64XX IISv2 Mark Brown (3): ASoC: Reenable S3C64xx I2S support ASoC: Remove duplicate ADC/DAC widgets from wm_hubs.c ASoC: Provide default set_bias_level() implementation arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 5 +++++ sound/soc/codecs/wm_hubs.c | 6 ------ sound/soc/s3c24xx/Kconfig | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 6 +++--- sound/soc/soc-dapm.c | 8 ++++++-- 5 files changed, 15 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH 2/3] Add ak464x codec support
by Kuninori Morimoto 19 Aug '09

19 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak464x.c | 525 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak464x.h | 25 +++ 4 files changed, 556 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak464x.c create mode 100644 sound/soc/codecs/ak464x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..f66689c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK464X if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -74,6 +75,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK464X + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..4400ad4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak464x-objs := ak464x.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK464X) += snd-soc-ak464x.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/ak464x.c b/sound/soc/codecs/ak464x.c new file mode 100644 index 0000000..d2ef80d --- /dev/null +++ b/sound/soc/codecs/ak464x.c @@ -0,0 +1,525 @@ +/* + * ak464x.c -- AK464x ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * Copyright 2005 Openedhand Ltd. + * Author: Richard Purdie <richard(a)openedhand.com> + * + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It is able to use headphone output + */ + +#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 <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 "ak464x.h" + +#define AK464X_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK464X_CACHEREGNUM 0x25 + +#define DPRINTK(param...) printk(param) + +struct snd_soc_codec_device soc_codec_dev_ak464x; + +/* codec private data */ +struct ak464x_priv { + unsigned int sysclk; +}; + +/* + * ak464x register cache + */ +static const u16 ak464x_reg[AK464X_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak464x register cache + */ +static inline unsigned int ak464x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK464X_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak464x_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak464x register cache + */ +static inline void ak464x_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK464X_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK464X register space + */ +static int ak464x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK464X register + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak464x_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak464x_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK464X_CACHEREGNUM; i++) + r |= ak464x_write(codec, i, cache[i]); + + return r; +}; + +static int ak464x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + codec->bias_level = level; + return 0; +} + +static int ak464x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* start headphone output */ + ak464x_write(codec, 0x0f, 0x09); + ak464x_write(codec, 0x0e, 0x19); + ak464x_write(codec, 0x09, 0x91); + ak464x_write(codec, 0x0c, 0x91); + ak464x_write(codec, 0x0a, 0x28); + ak464x_write(codec, 0x0d, 0x28); + ak464x_write(codec, 0x00, 0x64); + ak464x_write(codec, 0x01, 0x3b); + ak464x_write(codec, 0x01, 0x7b); + } else { + /* start stereo input */ + ak464x_write(codec, 0x02, 0x05); + ak464x_write(codec, 0x06, 0x3c); + ak464x_write(codec, 0x08, 0xe1); + ak464x_write(codec, 0x0b, 0x00); + ak464x_write(codec, 0x07, 0x21); + ak464x_write(codec, 0x00, 0x41); + ak464x_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak464x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak464x_write(codec, 0x01, 0x3b); + ak464x_write(codec, 0x01, 0x0b); + ak464x_write(codec, 0x00, 0x40); + ak464x_write(codec, 0x0e, 0x11); + ak464x_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak464x_write(codec, 0x00, 0x40); + ak464x_write(codec, 0x10, 0x00); + ak464x_write(codec, 0x07, 0x01); + } +} + +static int ak464x_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak464x_priv *ak464x = codec->private_data; + + ak464x->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak464x_dai_ops = { + .startup = ak464x_dai_startup, + .shutdown = ak464x_dai_shutdown, + .set_sysclk = ak464x_dai_set_sysclk, +}; + +struct snd_soc_dai ak464x_dai = { + .name = "AK464X", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,}, + .ops = &ak464x_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak464x_dai); + +static int ak464x_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; + + ak464x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak464x_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak464x_sync(codec); + ak464x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ak464x_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the AK464X driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak464x_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + int ret = 0; + + codec->name = "AK464X"; + codec->owner = THIS_MODULE; + codec->read = ak464x_read_reg_cache; + codec->write = ak464x_write; + codec->set_bias_level = ak464x_set_bias_level; + codec->dai = &ak464x_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(ak464x_reg); + codec->reg_cache = kmemdup(ak464x_reg, + sizeof(ak464x_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak464x: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak464x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak464x: failed to register card\n"); + goto card_err; + } + + /* clock setting */ + ak464x_write(codec, 0x01, 0x08); + ak464x_write(codec, 0x04, 0x4a); + ak464x_write(codec, 0x05, 0x27); + ak464x_write(codec, 0x00, 0x40); + ak464x_write(codec, 0x01, 0x0b); + + 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 *ak464x_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static int ak464x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = ak464x_socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = ak464x_init(socdev); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK464X\n"); + + return ret; +} + +static int ak464x_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id ak464x_i2c_id[] = { + { "ak464x", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak464x_i2c_id); + +static struct i2c_driver ak464x_i2c_driver = { + .driver = { + .name = "AK464X I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak464x_i2c_probe, + .remove = ak464x_i2c_remove, + .id_table = ak464x_i2c_id, +}; + +static int ak464x_add_i2c_device(struct platform_device *pdev, + const struct ak464x_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ak464x_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ak464x", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&ak464x_i2c_driver); + return -ENODEV; +} +#endif + +static int ak464x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak464x_setup_data *setup; + struct snd_soc_codec *codec; + struct ak464x_priv *ak464x; + int ret; + + printk(KERN_INFO "AK464X Audio Codec %s", AK464X_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak464x = kzalloc(sizeof(struct ak464x_priv), GFP_KERNEL); + if (ak464x == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak464x; + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak464x_socdev = socdev; + ret = -ENODEV; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = ak464x_add_i2c_device(pdev, setup); + } +#endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } + + return ret; +} + +/* power down chip */ +static int ak464x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec->control_data) + ak464x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&ak464x_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak464x = { + .probe = ak464x_probe, + .remove = ak464x_remove, + .suspend = ak464x_suspend, + .resume = ak464x_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak464x); + +static int __init ak464x_modinit(void) +{ + return snd_soc_register_dai(&ak464x_dai); +} +module_init(ak464x_modinit); + +static void __exit ak464x_exit(void) +{ + snd_soc_unregister_dai(&ak464x_dai); +} +module_exit(ak464x_exit); + +MODULE_DESCRIPTION("Soc AK464x driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak464x.h b/sound/soc/codecs/ak464x.h new file mode 100644 index 0000000..07cd1bf --- /dev/null +++ b/sound/soc/codecs/ak464x.h @@ -0,0 +1,25 @@ +/* + * ak464x.h -- AK464x Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * 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. + */ + +#ifndef _AK464X_H +#define _AK464X_H + +struct ak464x_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak464x_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak464x; + +#endif -- 1.6.0.4

3 3

[alsa-devel] [PATCH] snd-usb-audio: bootup sequence provided for new device - Digidesign Mbox 2
by Damien Zammit 19 Aug '09

19 Aug '09

Digidesign Mbox 2 support for Linux - Damien Zammit 19/08/2009 dczammit@flop:~$ dmesg [132064.931715] ALSA sound/usb/usbaudio.c:3279: Sending Digidesign Mbox 2 boot sequence... [132064.934474] ALSA sound/usb/usbaudio.c:3306: device initialised! [132064.937473] ALSA sound/usb/usbaudio.c:3315: mbox2_boot: new boot length = 646 [132064.945769] ALSA sound/usb/usbaudio.c:2814: 36:2:2: add audio endpoint 0x3 [132064.950765] ALSA sound/usb/usbaudio.c:2814: 36:2:3: add audio endpoint 0x3 [132064.957314] ALSA sound/usb/usbaudio.c:2814: 36:3:2: add audio endpoint 0x4 [132064.961622] ALSA sound/usb/usbaudio.c:2814: 36:3:3: add audio endpoint 0x4 [132064.964575] ALSA sound/usb/usbaudio.c:2814: 36:3:6: add audio endpoint 0x4 [132064.971046] ALSA sound/usb/usbaudio.c:2814: 36:4:2: add audio endpoint 0x85 [132064.973545] ALSA sound/usb/usbaudio.c:2814: 36:4:3: add audio endpoint 0x85 [132064.979576] ALSA sound/usb/usbaudio.c:2814: 36:5:2: add audio endpoint 0x86 [132064.982522] ALSA sound/usb/usbaudio.c:2814: 36:5:3: add audio endpoint 0x86 [132064.986584] ALSA sound/usb/usbmidi.c:1357: created 1 output and 1 input ports [132064.986955] usbcore: registered new interface driver snd-usb-audio dczammit@flop:~$ dczammit@flop:~$ ls /proc/asound/M2/ id midi0 pcm0c pcm0p pcm1c pcm1p pcm2p stream0 stream1 stream2 usbbus usbid dczammit@flop:~$ dczammit@flop:~$ cat /proc/asound/M2/stream0 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio Playback: Status: Stop Interface 2 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Interface 2 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 5 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 6 IN (ASYNC) Rates: 8000 - 48000 (continuous) Interface 5 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 6 IN (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:~$ dczammit@flop:~$ cat /proc/asound/M2/stream1 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio #1 Playback: Status: Stop Interface 3 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 4 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Interface 3 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 4 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 4 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 8000 - 48000 (continuous) Interface 4 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 5 IN (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:~$ dczammit@flop:~$ cat /proc/asound/M2/stream2 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio #2 Playback: Status: Stop Interface 3 Altset 6 Format: 0x2 (16 bits) Channels: 2 Endpoint: 4 OUT (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:~$ I fixed the broken pipes!! I can get speaker-test working, but I dont get any sound... 1) Is format 0x20 (S24_3LE) supported yet? I can't get it to work with jackd version 0.116.2 2) Which interfaces/altsettings should I disable? I can't quite get the sound to come out. Damien

2 1

[alsa-devel] [PATCH 0/3] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 19 Aug '09

19 Aug '09

These patches are ALSA driver support for SuperH SH7724 1: Add SuperH FSI driver support for ALSA 2: Add ak464x codec support for ALSA 3: Add FSI-AK464x sound support for ms7724se 1 and 2 are very simple driver and these for ALSA. 3 is FSI-AK464x combination patch for ms7724se board. Best regards -- Kuninori Morimoto

1 0

[alsa-devel] implement delay in ioplug alsa plugin
by Stefan Schoenleitner 19 Aug '09

19 Aug '09

Hi, I managed to write an ioplug plugin which dumps a raw PCM stream to a file (similar to the way the file or tee plugins do, but without a slave device). However, instead of immediately writing all the samples to the file, I would like to add some delay so that the samples are written to the file in real time at the frequency of the sampling rate. Thus if there is a single channel stream with 16bits being played at a sampling rate of 8kHz, each 1/8000sec exactly 16 bits should be written to the file. The reason for this is that I have a DSP attached over SPI which needs exactly one audio sample per interval. In order to talk to the DSP there is a SPI device-file which my ioplugin would use to write the samples to. What is the best way to implement such a delay ? I'm pretty sure that just adding some usleep calls() won't work very well, since the write operations to the device file probably don't take the same amount of time every time. Maybe it would work with non-blocking writes to the device file and a high precision timer ? cheers, Stefan

2 1

[alsa-devel] [PATCH] hda: enable speaker output for Compaq 6530s/6531s
by Wu Fengguang 19 Aug '09

19 Aug '09

There are 3 causes to the silence of Compaq 6530s' internal speaker: - HP pin 0x11 always return 0xffff,ffff (pin_presence=1), which automutes the speaker at module loading time. - HP pin 0x11 won't emit unsol events on jack plug/unplug, so the speaker have no chance to be unmuted. - pin 0x1c must be set to PIN_OUT mode to make the speaker work this is the most weird part - line-in pin 0x1c and speaker pin 0x16 seem to be unrelated. Fix the problems by - add "Speaker" and "Headphone" mixers to allow hand tuning by user - add the 0x1c PIN_OUT quirk Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_analog.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/patch_analog.c 2009-08-05 09:40:13.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_analog.c 2009-08-16 16:49:39.000000000 +0800 @@ -3717,6 +3717,8 @@ static struct snd_kcontrol_new ad1884a_l HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), @@ -3837,8 +3839,10 @@ static struct hda_verb ad1884a_laptop_ve {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* required for compaq 6530s speaker output */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-C pin - internal mic-in */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ @@ -4003,6 +4007,8 @@ static struct snd_pci_quirk ad1884a_cfg_ SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x30e8, "HP Compaq 6530s", AD1884A_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30e9, "HP Compaq 6531s", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),

2 15

[alsa-devel] [PATCH] hda: add model for Intel DG45ID/DG45FC boards
by Wu Fengguang 19 Aug '09

19 Aug '09

The BIOS pin configs are in fact correct and shall not be overwritten. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) --- sound-2.6.orig/sound/pci/hda/patch_sigmatel.c 2009-08-19 16:59:50.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_sigmatel.c 2009-08-19 16:59:51.000000000 +0800 @@ -78,6 +78,7 @@ enum { STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, + STAC_92HD73XX_INTEL, STAC_DELL_M6_AMIC, STAC_DELL_M6_DMIC, STAC_DELL_M6_BOTH, @@ -1775,6 +1776,7 @@ static const char *stac92hd73xx_models[S [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", + [STAC_92HD73XX_INTEL] = "intel", [STAC_DELL_M6_AMIC] = "dell-m6-amic", [STAC_DELL_M6_DMIC] = "dell-m6-dmic", [STAC_DELL_M6_BOTH] = "dell-m6", @@ -1787,6 +1789,10 @@ static struct snd_pci_quirk stac92hd73xx "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5002, + "Intel DG45ID", STAC_92HD73XX_INTEL), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5003, + "Intel DG45FC", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255,

2 1

Re: [alsa-devel] Old photos
by tglx＠linutronix.de 18 Aug '09

18 Aug '09

Greetings from france, your friend.

1 0

[alsa-devel] SB Audigy 4 No Front speakers, only Rear / SigmaTel STAC9750, 51 codec
by David P 18 Aug '09

18 Aug '09

Hi, after some time trying to trouble shoot this sound issue, I was encuraged to post a message explaining my situation in hopes maybe I would get some help. I have a SB Audigy 4. Surround Speakers work, Mic works. I do not have sound from front speakers. I'm not trying to do 5.1 surround or anything fancy. Just have headphones and a pair of speakers. I find it strange I have sound coming from the Rear and not the Front. All the searching I have done for this card seems people are having the oposite issue. I have checked all connections, they are all fine. I enabled all mixer controls and turned all the volumes up. Running various command line speaker-test's always reports no sound from Fronts, only Rears. So here I am asking for help. Here is a copy of my Alsa-info I hope this helps . upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Tue Aug 18 15:42:27 UTC 2009 !!Linux Distribution !!------------------ Welcome to openSUSE 11.1 - Kernel \r (\l). openSUSE 11.1 (i586) LSB_VERSION="core-2.0-noarch:core-3.2-noarch:core-2.0-ia32:core-3.2-ia32" !!DMI Information !!--------------- Manufacturer: Gigabyte Technology Co., Ltd. Product Name: P31-S3G !!Kernel Information !!------------------ Kernel release: 2.6.27.25-0.1-pae Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.17 Library version: Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_emu10k1 !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/opt/kde3/bin/artsd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [Audigy2 ]: Audigy2 - Audigy 4 [SB0610] Audigy 4 [SB0610] (rev.0, serial:0x10211102) at 0xd000, irq 18 !!PCI Soundcards installed in the system !!-------------------------------------- 04:02.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 04:02.0 0401: 1102:0008 Subsystem: 1102:1021 !!Modprobe options (Sound related) !!-------------------------------- snd-emu10k1: extin=0x3fff extout=0x7fff !!Loaded sound module options !!-------------------------- !!Module: snd_emu10k1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_ir : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N extin : 16383,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 extout : 32767,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 max_buffer_size : 128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128 max_synth_voices : 64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64 seq_ports : 4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4 subsystem : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: SigmaTel STAC9750,51 PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 20-bit 3D enhancement : SigmaTel 3D Enhancement Current setup Mic gain : +20dB [+20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=1 AMAP DSA=0 SPDIF VRA Extended status : SPCV SPDIF=7/8 PCM front DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Copyright Category=0x0 Generation=0 Rate=48kHz 0:00 = 6a90 0:02 = 0505 0:04 = 9f1f 0:06 = 801f 0:08 = 0000 0:0a = 801e 0:0c = 8009 0:0e = 0047 0:10 = 0a0a 0:12 = 8808 0:14 = 8808 0:16 = 8b0b 0:18 = 9f1f 0:1a = 0000 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 000f 0:28 = 0605 0:2a = 0410 0:2c = bb80 0:2e = 0000 0:30 = 0000 0:32 = bb80 0:34 = 0000 0:36 = 0000 0:38 = 0000 0:3a = 2000 0:3c = 0000 0:3e = 0100 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0003 0:4e = ffff 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0002 0:6e = 1000 0:70 = 0000 0:72 = 0000 0:74 = 0800 0:76 = 0000 0:78 = 0000 0:7a = 0000 0:7c = 8384 0:7e = 7650 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 12 Aug 18 05:04 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 3 Aug 18 05:04 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 14 Aug 18 10:04 /dev/snd/hwC0D2 crw-rw----+ 1 root audio 116, 5 Aug 18 05:04 /dev/snd/midiC0D0 crw-rw----+ 1 root audio 116, 4 Aug 18 05:04 /dev/snd/midiC0D1 crw-rw----+ 1 root audio 116, 15 Aug 18 10:04 /dev/snd/midiC0D2 crw-rw----+ 1 root audio 116, 16 Aug 18 10:04 /dev/snd/midiC0D3 crw-rw----+ 1 root audio 116, 11 Aug 18 05:04 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 10 Aug 18 10:20 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 9 Aug 18 05:04 /dev/snd/pcmC0D1c crw-rw----+ 1 root audio 116, 8 Aug 18 05:04 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 7 Aug 18 05:04 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 6 Aug 18 05:04 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 13 Aug 18 10:04 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Aug 18 05:04 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Audigy2 [Audigy 4 [SB0610]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 31/32 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Subdevice #8: subdevice #8 Subdevice #9: subdevice #9 Subdevice #10: subdevice #10 Subdevice #11: subdevice #11 Subdevice #12: subdevice #12 Subdevice #13: subdevice #13 Subdevice #14: subdevice #14 Subdevice #15: subdevice #15 Subdevice #16: subdevice #16 Subdevice #17: subdevice #17 Subdevice #18: subdevice #18 Subdevice #19: subdevice #19 Subdevice #20: subdevice #20 Subdevice #21: subdevice #21 Subdevice #22: subdevice #22 Subdevice #23: subdevice #23 Subdevice #24: subdevice #24 Subdevice #25: subdevice #25 Subdevice #26: subdevice #26 Subdevice #27: subdevice #27 Subdevice #28: subdevice #28 Subdevice #29: subdevice #29 Subdevice #30: subdevice #30 Subdevice #31: subdevice #31 card 0: Audigy2 [Audigy 4 [SB0610]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Audigy2 [Audigy 4 [SB0610]], device 3: emu10k1 [Multichannel Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Audigy2 [Audigy 4 [SB0610]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audigy2 [Audigy 4 [SB0610]], device 1: emu10k1 mic [Mic Capture] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audigy2 [Audigy 4 [SB0610]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Audigy2] Card hw:0 'Audigy2'/'Audigy 4 [SB0610] (rev.0, serial:0x10211102) at 0xd000, irq 18' Mixer name : 'SigmaTel STAC9750,51' Components : 'AC97a:83847650' Controls : 201 Simple ctrls : 36 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined cvolume cswitch cswitch-joined Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 31 Mono: Playback 68 [68%] [-12.80dB] Front Left: Capture 26 [84%] [-7.50dB] [on] Front Right: Capture 26 [84%] [-7.50dB] [on] Simple mixer control 'Tone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Bass',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 40 Front Left: 27 [68%] Front Right: 27 [68%] Simple mixer control 'Treble',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 40 Front Left: 27 [68%] Front Right: 27 [68%] Simple mixer control 'PCM',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 100 Front Left: Capture 65 [65%] [-14.00dB] Front Right: Capture 65 [65%] [-14.00dB] Simple mixer control 'PCM Center',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 75 [75%] [-10.00dB] Simple mixer control 'PCM Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 75 [75%] [-10.00dB] Front Right: Playback 75 [75%] [-10.00dB] Simple mixer control 'PCM LFE',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 75 [75%] [-10.00dB] Simple mixer control 'PCM Out Path & Mute',0 Capabilities: enum Items: 'pre 3D' 'post 3D' Item0: 'pre 3D' Simple mixer control 'PCM Side',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 72 [72%] [-11.20dB] Front Right: Playback 72 [72%] [-11.20dB] Simple mixer control 'PCM Surround',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 75 [75%] [-10.00dB] Front Right: Playback 75 [75%] [-10.00dB] Simple mixer control 'Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 82 [82%] [-7.20dB] Front Right: Playback 82 [82%] [-7.20dB] Simple mixer control 'Surround',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 84 [84%] [-6.40dB] Front Right: Playback 84 [84%] [-6.40dB] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 70 [70%] [-12.00dB] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 64 [64%] [-14.40dB] Simple mixer control 'Side',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 82 [82%] [-7.20dB] Front Right: Playback 82 [82%] [-7.20dB] Simple mixer control 'Synth',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 81 [81%] [-7.60dB] Capture 100 [100%] [0.00dB] Front Right: Playback 81 [81%] [-7.60dB] Capture 100 [100%] [0.00dB] Simple mixer control 'Wave',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 82 [82%] [-7.20dB] Front Right: Playback 82 [82%] [-7.20dB] Simple mixer control 'Line',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 21 [68%] [-3.00dB] [on] Front Right: Capture 21 [68%] [-3.00dB] [on] Simple mixer control 'CD',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Mic',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 31 Mono: Capture 24 [77%] [1.50dB] [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Phone',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 31 Mono: Capture 22 [71%] [-1.50dB] [off] Simple mixer control 'IEC958 Optical',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 75 [75%] [-10.00dB] Capture 73 [73%] [-10.80dB] Front Right: Playback 75 [75%] [-10.00dB] Capture 73 [73%] [-10.80dB] Simple mixer control 'IEC958 Optical Raw',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PC Speaker',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 0 [0%] [-45.00dB] [off] Simple mixer control 'Aux',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 20 [65%] [-4.50dB] [off] Front Right: Capture 20 [65%] [-4.50dB] [off] Simple mixer control 'Aux2',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'AMic',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Simple mixer control 'Analog Mix',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Simple mixer control 'Audigy Analog/Digital Output Jack',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Audigy CD',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 76 [76%] [-9.60dB] Capture 74 [74%] [-10.40dB] Front Right: Playback 76 [76%] [-9.60dB] Capture 74 [74%] [-10.40dB] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output !!------------- --startcollapse-- state.Audigy2 { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Front Playback Volume' value.0 75 value.1 75 } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Surround Playback Volume' value.0 75 value.1 75 } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Side Playback Volume' value.0 72 value.1 72 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Center Playback Volume' value 75 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM LFE Playback Volume' value 75 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Wave Playback Volume' value.0 82 value.1 82 } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Synth Playback Volume' value.0 81 value.1 81 } control.8 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Capture Volume' value.0 65 value.1 65 } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Synth Capture Volume' value.0 100 value.1 100 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'AMic Playback Volume' value.0 0 value.1 0 } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Audigy CD Playback Volume' value.0 76 value.1 76 } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Audigy CD Capture Volume' value.0 74 value.1 74 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'IEC958 Optical Playback Volume' value.0 75 value.1 75 } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'IEC958 Optical Capture Volume' value.0 73 value.1 73 } control.18 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Analog Mix Playback Volume' value.0 0 value.1 0 } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Analog Mix Capture Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Aux2 Playback Volume' value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Aux2 Capture Volume' value.0 0 value.1 0 } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 82 value.1 82 } control.23 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 84 value.1 84 } control.24 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Center Playback Volume' value 70 } control.25 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'LFE Playback Volume' value 64 } control.26 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 82 value.1 82 } control.27 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 40' iface MIXER name 'Tone Control - Bass' value.0 27 value.1 27 } control.28 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 40' iface MIXER name 'Tone Control - Treble' value.0 27 value.1 27 } control.29 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Tone Control - Switch' value.0 true value.1 true } control.30 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 68 } control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'IEC958 Optical Raw Playback Switch' value.0 true value.1 true } control.32 { comment.access 'read write' comment.type BOOLEAN comment.count 64 iface PCM device 2 name 'Captured FX8010 Outputs' value.0 false value.1 false value.2 false value.3 false value.4 false value.5 false value.6 false value.7 false value.8 false value.9 false value.10 false value.11 false value.12 false value.13 false value.14 false value.15 false value.16 false value.17 false value.18 false value.19 false value.20 false value.21 false value.22 false value.23 false value.24 false value.25 false value.26 false value.27 false value.28 false value.29 false value.30 false value.31 false value.32 true value.33 true value.34 true value.35 true value.36 true value.37 true value.38 true value.39 true value.40 true value.41 true value.42 true value.43 true value.44 true value.45 true value.46 true value.47 true value.48 false value.49 false value.50 false value.51 false value.52 false value.53 false value.54 false value.55 false value.56 false value.57 false value.58 false value.59 false value.60 false value.61 false value.62 false value.63 false } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Capture Switch' value true } control.34 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Capture Volume' value.0 26 value.1 26 } control.39 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Speaker Capture Switch' value false } control.40 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' comment.dbmin -4500 comment.dbmax 0 iface MIXER name 'PC Speaker Capture Volume' value 0 } control.41 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Capture Switch' value false } control.42 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Phone Capture Volume' value 22 } control.43 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Capture Switch' value true } control.44 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Capture Volume' value 24 } control.45 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost (+20dB)' value true } control.46 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Capture Switch' value true } control.47 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Capture Volume' value.0 21 value.1 21 } control.48 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'CD Capture Switch' value false } control.49 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'CD Capture Volume' value.0 23 value.1 23 } control.50 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Video Capture Switch' value false } control.51 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Video Capture Volume' value.0 23 value.1 23 } control.52 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Aux Capture Switch' value false } control.53 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Aux Capture Volume' value.0 20 value.1 20 } control.59 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'pre 3D' comment.item.1 'post 3D' iface MIXER name 'PCM Out Path & Mute' value 'pre 3D' } control.61 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.62 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic1 } control.65 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value true } control.66 { comment.access 'read write' comment.type INTEGER comment.count 24 comment.range '0 - 63' iface PCM name 'EMU10K1 PCM Send Routing' value.0 0 value.1 1 value.2 2 value.3 3 value.4 0 value.5 0 value.6 0 value.7 0 value.8 0 value.9 1 value.10 2 value.11 3 value.12 0 value.13 0 value.14 0 value.15 0 value.16 0 value.17 1 value.18 2 value.19 3 value.20 0 value.21 0 value.22 0 value.23 0 } control.98 { comment.access 'read write' comment.type INTEGER comment.count 24 comment.range '0 - 255' iface PCM name 'EMU10K1 PCM Send Volume' value.0 255 value.1 255 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 value.8 255 value.9 0 value.10 0 value.11 0 value.12 0 value.13 0 value.14 0 value.15 0 value.16 0 value.17 255 value.18 0 value.19 0 value.20 0 value.21 0 value.22 0 value.23 0 } control.130 { comment.access 'read write' comment.type INTEGER comment.count 3 comment.range '0 - 65535' iface PCM name 'EMU10K1 PCM Volume' value.0 65535 value.1 65535 value.2 65535 } control.210 { comment.access read comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Mask' value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.211 { comment.access read comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Mask' index 1 value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.212 { comment.access read comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Mask' index 2 value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.213 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Default' value '0482000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.214 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Default' index 1 value '0492100200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.215 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Default' index 2 value '0492100200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.216 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Audigy Analog/Digital Output Jack' value true } } --endcollapse-- !!All Loaded Modules !!------------------ Module ip6t_LOG xt_tcpudp xt_pkttype ipt_LOG xt_limit binfmt_misc snd_pcm_oss snd_mixer_oss snd_seq_midi snd_emu10k1_synth snd_emux_synth snd_seq_virmidi snd_seq_midi_event snd_seq_midi_emul snd_seq af_packet ip6t_REJECT nf_conntrack_ipv6 ip6table_raw xt_NOTRACK ipt_REJECT xt_state iptable_raw iptable_filter ip6table_mangle nf_conntrack_netbios_ns nf_conntrack_ipv4 nf_conntrack ip_tables ip6table_filter ip6_tables x_tables cpufreq_conservative ipv6 cpufreq_userspace cpufreq_powersave acpi_cpufreq speedstep_lib fuse loop dm_mod snd_emu10k1 snd_rawmidi snd_ac97_codec ac97_bus snd_pcm snd_seq_device snd_timer snd_page_alloc snd_util_mem snd_hwdep snd iTCO_wdt nvidia iTCO_vendor_support rtc_cmos r8169 ppdev i2c_i801 sr_mod intel_agp soundcore rtc_core cdrom mii pcspkr rtc_lib i2c_core parport_pc button joydev parport agpgart floppy sg usbhid hid ff_memless uhci_hcd ehci_hcd sd_mod crc_t10dif usbcore edd ext3 mbcache jbd fan ide_pci_generic ide_core ata_generic ata_piix libata scsi_mod dock thermal processor thermal_sys hwmon !!ALSA/HDA dmesg !!------------------ EMU10K1_Audigy 0000:04:02.0: PCI INT A -> GSI 18 (level, low) -> IRQ 18 ALSA sound/pci/emu10k1/emu10k1_main.c:219: Audigy2 value: Special config. Adding 2104472k swap on /dev/sda5. Priority:-1 extents:1 across:2104472

1 0

[alsa-devel] [PATCH] ASoC: Fix data format configuration for S3C64XX IISv2
by Joonyoung Shim 18 Aug '09

18 Aug '09

The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD register. This patch changes to the defined values it. And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 5 +++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 6 +++--- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index 0fad757..07659da 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -33,6 +33,11 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + #define S3C64XX_IISMOD_IMS_PCLK (0 << 10) #define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 1a28317..ebfb2f6 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -357,19 +357,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, #endif #ifdef CONFIG_PLAT_S3C64XX - iismod &= ~0x606; + iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK); /* Sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: /* 8 bit sample, 16fs BCLK */ - iismod |= 0x2004; + iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS); break; case SNDRV_PCM_FORMAT_S16_LE: /* 16 bit sample, 32fs BCLK */ break; case SNDRV_PCM_FORMAT_S24_LE: /* 24 bit sample, 48fs BCLK */ - iismod |= 0x4002; + iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS); break; } #endif -- 1.6.0.4

2 4

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 18 Aug '09

18 Aug '09

The following changes since commit d2a382143b35a486b8089861790d6b3ec4e46932: Mark Brown (1): ASoC: Update AD1938 for new TDM slot API 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 (3): ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM work new ad1836 codec driver based on asoc ASoC: delete -spi suffix in ad1938 and free private data while registers fail Chaithrika U S (3): ASoC: DaVinci: McASP driver enhacements ASoC: DaVinci: Add a DAI format to McASP driver ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM Janusz Krzysztofik (2): ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 ASoC: OMAP: Enhance OMAP1510 DMA progress software counter Marek Vasut (1): ALSA: Allow passing platform_data for pxa2xx-ac97 Mark Brown (20): ASoC: Add WM8974 CODEC driver ASoC: WM8974 checkpatch cleanups ASoC: Update WM8974 to use standard I2C device probe methods ASoC: Remove unreferenced wm8974_add_controls() ASoC: Refresh WM8974 bias configuration ASoC: Declare 2 channels for WM8974 ASoC: Refresh WM8974 PLL configuration ASoC: Add WM8974 TLV information ASoC: Use symmetric rates for WM8974 ASoC: WM8974 cosmetic cleanups ASoC: WM8974 DAPM cleanups ASoC: Minor cleanups to AD1938 driver ASoC: Factor out shared code from WM8993 Merge branch 'for-2.6.32' into mxc ASoC: Remove unneeded i.MX dependency on SND ASoC: Staticise unexported variables ASoC: Hook i.MX into build Merge branch 'wm8974-upstream' into for-2.6.32 ASoC: Factor out cache I/O from WM8974 ASoC: Fix handling of bias levels for non-DAPM codecs Peter Ujfalusi (2): ASoC: TWL4030: Introduce PGAs for outputs ASoC: TWL4030: Fix for capture mixer strings Shine Liu (1): ASoC: UDA134X: Fix mistaken mute/unmute code javier Martin (5): ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec. ASoC: add DMA platform driver for MX1x and MX2x ASoC: add DAI platform ssi driver for MXC ASoC: add machine driver for i.mx27_visstrim_m10 board ASoC: Fix review issues in i.MX2x PCM driver arch/arm/mach-pxa/include/mach/audio.h | 3 + include/sound/ac97_codec.h | 3 + sound/arm/pxa2xx-ac97.c | 10 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad1836.c | 446 ++++++++++++++++ sound/soc/codecs/ad1836.h | 64 +++ sound/soc/codecs/ad1938.c | 16 +- sound/soc/codecs/twl4030.c | 155 ++++--- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/wm8974.c | 808 +++++++++++++++++++++++++++++ sound/soc/codecs/wm8974.h | 99 ++++ sound/soc/codecs/wm8993.c | 716 +------------------------- sound/soc/codecs/wm_hubs.c | 758 ++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.h | 24 + sound/soc/davinci/Kconfig | 13 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +- sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 113 ++++- sound/soc/davinci/davinci-mcasp.h | 5 + sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/imx/Kconfig | 21 + sound/soc/imx/Makefile | 10 + sound/soc/imx/mx1_mx2-pcm.c | 488 ++++++++++++++++++ sound/soc/imx/mx1_mx2-pcm.h | 26 + sound/soc/imx/mx27vis_wm8974.c | 317 ++++++++++++ sound/soc/imx/mxc-ssi.c | 868 ++++++++++++++++++++++++++++++++ sound/soc/imx/mxc-ssi.h | 238 +++++++++ sound/soc/omap/omap-pcm.c | 37 +- sound/soc/pxa/pxa2xx-ac97.c | 12 +- sound/soc/soc-dapm.c | 16 + 35 files changed, 4532 insertions(+), 792 deletions(-) create mode 100644 sound/soc/codecs/ad1836.c create mode 100644 sound/soc/codecs/ad1836.h create mode 100644 sound/soc/codecs/wm8974.c create mode 100644 sound/soc/codecs/wm8974.h create mode 100644 sound/soc/codecs/wm_hubs.c create mode 100644 sound/soc/codecs/wm_hubs.h create mode 100644 sound/soc/imx/Kconfig create mode 100644 sound/soc/imx/Makefile create mode 100644 sound/soc/imx/mx1_mx2-pcm.c create mode 100644 sound/soc/imx/mx1_mx2-pcm.h create mode 100644 sound/soc/imx/mx27vis_wm8974.c create mode 100644 sound/soc/imx/mxc-ssi.c create mode 100644 sound/soc/imx/mxc-ssi.h

2 1

[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 18 Aug '09

18 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..2eb1a69 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x3); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x2); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Mark Brown 17 Aug '09

17 Aug '09

From: Daniel Ribeiro <drwyrm(a)gmail.com> Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 25d62ac..2d3fa29 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 156f2a4..c965323 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1214,8 +1214,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1227,7 +1228,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 8889cd3..9f7c61e 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -190,7 +190,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..5b9ed64 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb8d7a7..e984a17 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2223,17 +2223,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- 1.6.3.3

3 12

[alsa-devel] Strange div-by-zero in alsa-lib 1.0.20's snd_pcm_mmap_begin()
by Daniel Chen 17 Aug '09

17 Aug '09

Hi, In https://bugs.launchpad.net/ubuntu/+source/alsa-lib/+bug/412677 we're debugging an issue where snd_pcm_mmap_begin(), at line 6409 with *offset = *pcm->appl.ptr % pcm->buffer_size; appears to have pcm->buffer_size == 0. What's the correct approach in handling this corner case? Thanks, Dan

3 4

[alsa-devel] How to use hda-verb to detect
by Ozan Çağlayan 17 Aug '09

17 Aug '09

Hi, I'm playing with an HP DV6(which produces sound only when model=hp-dv5, will send a patch for it after further testing) on which HP automuting is not working. The laptop has apparently two HP outputs which seems to behave completely parallel. One thing that I really don't get with hda-verb, is the last parameter. I used hda-emu for finding out an example call: Front Playback Volume:0 MIN/MAX: 0/64, VAL: [39] [39] dB min/max: -48.00/0.00, [-18.-75dB] [-18.-75dB] > set 1 64 64 send: NID=0x10, VERB=0x3a0(set_amp_gain_mute,O:L#0), PARM=0x7f receive: 0x0 send: NID=0x10, VERB=0x390(set_amp_gain_mute,O:R#0), PARM=0x7f receive: 0x0 > set 1 63 63 send: NID=0x10, VERB=0x3a0(set_amp_gain_mute,O:L#0), PARM=0x7e receive: 0x0 send: NID=0x10, VERB=0x390(set_amp_gain_mute,O:R#0), PARM=0x7e receive: 0x0 I can easily understand that I'm calling set_amp_gain_mute VERB for NID 0x10 with a level value 0x7f and 0x7e. Now when I run hda-verb without parameter, it gives me a bunch of possible values for PARM: VENDOR_ID, SUBSYSTEM_ID, REV_ID, NODE_COUNT, FUNCTION_TYPE AUDIO_FG_CAP, AUDIO_WIDGET_CAP, PCM, STREAM, PIN_CAP, AMP_IN_CAP CONNLIST_LEN, POWER_STATE, PROC_CAP, GPIO_CAP, AMP_OUT_CAP VOL_KNB_CA I can't easily understand which one I have to choose for example to send the verb SET_UNSOLICITED_ENABLE or SET_PIN_SENSE. I've found the widget below in HP DV6 (OUT HP Detect grabbed my attention and I've seen that the unsolicited events has something to do with HP sense/automute, just playling blindly), Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect ** Pin Default 0x40f100f0: [N/A] Other at Ext N/A Conn = 1/8, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=0c, enabled=0 Connection: 3 0x10 0x11 0x17* and say that I want to send SET_UNSOLICITED_ENABLE to 0x0a, which PARM should I choose? I'm also attaching alsa-info output, Thanks! Ozan

2 7

[alsa-devel] aplay: set_params:984: Channels count non available
by prashant chopade 17 Aug '09

17 Aug '09

Hello, I am using alsa library 1.0.20. I am using linux-2.6.27. But I have shown certain useful messages: # cat /etc/asound.conf pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] Subdevices: 1/1 Subdevice #0: subdevice #0 # aplay /home/Front_Center.wav -c 1 Playing WAVE '/home/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:984: Channels count non available To remove this error which channel i have to used or set and how? Thanks & Regards, Prashant C. Looking for local information? Find it on Yahoo! Local http://in.local.yahoo.com/

1 0

[alsa-devel] [PATCH] ASoC: UDA134X Codec: Fix mute/unmute code mistake and add ADC/DAC power control support
by Shine Liu 17 Aug '09

17 Aug '09

There is a mistake in current uda134x_mute function: mute_reg has been changed in line 162 or line 164, so uda134x_write should write "mute_reg" but not "mute_reg & ~(1<<2)" to UDA134X_DATA010. Besides, because there is no DAPM configuration for uda134x, when system starts up, snd_soc_int_card calls snd_soc_dapm_new_widgets, and snd_soc_dapm_new_widgets calls dapm_power_widgets. In function dapm_power_widgets, codec->dapm_widgets has no list entry, so sys_power retains it's original value zero. Then snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE) and snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY) are called sequentially. Finally, the uda134x codec goes to STANDBY mode, so the ADC/DAC power control bits of UDA134X_STATUS1 register keeps the 0 value. UDA134X has no trigger function currently, and the ADC/DAC power control bits of UDA134X_STATUS1 register are not exported in uda1341_snd_controls, so there's no way to enable the ADC/DAC power control bits when you want to use the codec. When playing with aplay or recording with arecord, there is no sound output or no wave input. I have added the uda134x_trigger function, which turns on/off the ADC/DAC power control bits according the substream type and the command. I also exported the ADC/DAC power control bits to uda1341_snd_controls, so we can also turn these bits on/off as we need via a mixer tool like alsamixer. The patch created against linux-2.6.31-rc3. Tested on a s3c2440 development board with UDA1341TS codec. Signed-off-by: Shine Liu <shinel(a)foxmail.com> --- sound/soc/codecs/uda134x.c.orig 2009-07-14 09:18:52.000000000 +0800 +++ sound/soc/codecs/uda134x.c 2009-08-17 13:46:57.000000000 +0800 @@ -163,7 +163,7 @@ else mute_reg &= ~(1<<2); - uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + uda134x_write(codec, UDA134X_DATA010, mute_reg); return 0; } @@ -339,6 +339,38 @@ return 0; } +static int uda134x_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int stream = substream->stream; + struct snd_soc_codec *codec = dai->codec; + u8 power_ctrl_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + + pr_debug("%s stream: %d, cmd: %d\n", __func__, stream, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg |= (1<<0); + else + power_ctrl_reg |= (1<<1); + break; + + case SNDRV_PCM_TRIGGER_STOP: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg &= ~(1<<0); + else + power_ctrl_reg &= ~(1<<1); + break; + + default: + return 0; + } + + uda134x_write(codec, UDA134X_STATUS1, power_ctrl_reg); + return 0; +} + static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -416,6 +448,8 @@ SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("ADC Power Switch", UDA134X_STATUS1, 1, 1, 0), +SOC_SINGLE("DAC Power Switch", UDA134X_STATUS1, 0, 1, 0), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; @@ -438,6 +472,7 @@ .digital_mute = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, .set_fmt = uda134x_set_dai_fmt, + .trigger = uda134x_trigger, }; struct snd_soc_dai uda134x_dai = {

2 8

[alsa-devel] [PATCHv3 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 17 Aug '09

17 Aug '09

Hello again guys, Here is version 3 of these changes. I've changed just 2 things: - all mcbsp instances are in element mode by default - moved all mcbsp related code to omap-mcbsp.c (although a single callback is still needed) Jarkko, I think we can add your patch for reading strings for dma op mode later on. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 77 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 13 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] custom ioplug plugin does not work
by Stefan Schoenleitner 17 Aug '09

17 Aug '09

Hi, I'm trying to write an ioplug plugin which dumps the raw PCM stream to a device file in real time. With real time I mean that a device available through a device file should receive the PCM samples exactly at the sampling rate. Thus for example with a sampling rate of 8kHz and 16bit linear samples, each 1/8000 = 125us the device should receive exactly one 16bit sample (if we assume that only one audio channel is used). This should all happen on an ARM9 target with the device being a fast SPI device that can handle the transfer speeds. In contrast to the existing file or tee plugins, this should also work if no slave device is specified. With the mentioned two existing plugins all PCM data is immediately written to the output file if the null device is specified. For this reason these plugins do not seem to be usable for my application. To solve the mentioned problem it was suggested to me that I should write an ALSA plugin which does the job. At the moment I already have some partially working code which is available at http://pastebin.com/m3666868b My problem with the code is that when playing audio over the plugin, it transfer some data chunks in the beginning and then it seems to be blocking forever. What I'm doing in my plugin code is basically the following: 1. open output file for writing (file descriptor is filedump->file_fd) 2. filedump->io.poll_fd = filedump->file_fd; filedump->io.poll_events = POLLOUT; filedump->io.mmap_rw = 0; 3. snd_pcm_ioplug_create() I have callbacks for start, stop, close, pointer, prepare and transfer: In the start and stop callbacks I don't do anything at the moment. In the close callback I close the output file. In the pointer callback I just return the current hw_ptr. In the prepare callback the hw_ptr is initialized with 0. However, the most imporant callback seems to be the transfer callback. At the moment it just prints a hexdump of the PCM buffer and returns the size of the buffer to indicate that the whole buffer has been transfered. * Is this correct so far ? * Why is the plugin blocking forever after some parts of the buffer have been transfered and the start callback is called ? The output of the plugin looks like this: ------------------------------------------------------------------------------------------ _snd_pcm_filedump_open:311 25740 SND_PCM_PLUGIN_DEFINE_FUNC(): done, err=0 Playing WAVE '/usr/share/sounds/card_shuffle.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo snd_pcm_filedump_prepare:134 25740 Preparing with io->period_size=5512 io->buffer_size=22050 snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=2 io->nonblock=0 <hexdump ...> snd_pcm_filedump_start:70 25740 snd_pcm_filedump_start() ^CAborted by signal Interrupt... snd_pcm_filedump_stop:76 25740 snd_pcm_filedump_stop() snd_pcm_filedump_close:59 25740 snd_pcm_filedump_close() ------------------------------------------------------------------------------------------ As you can see, after calling snd_pcm_filedump_start() the plugin seems to be blocking forever. * Why is data transfered at all before start is called ? Unfortunately the ALSA documentation seems to be really bad. Reading (and partially copying) code from other plugins brought me this far, but not further at the moment. I would appreciate any help to get rid of the problem mentioned above. cheers, stefan

1 0

[alsa-devel] [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510
by Janusz Krzysztofik 17 Aug '09

17 Aug '09

Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP models. Remove unnecessary DMA transfer restart from interrupt handler routine. The interrupt routine used to maintain a period index, originally needed for counting up periods up to a full buffer in order to restart the DMA transfer. For some time, this counter is also used as a replacement for hardware DMA progress counter that has been found unusable on OMAP1510 in case of playback. Thus, the period index calculation cannot be omitted completely. However, the accuracy of this counter can still suffer from missing DMA interrupts. In order to work correctly, it requires patch 1 from this series also applied: [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c.orig 2009-08-07 13:38:23.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c 2009-08-10 08:31:08.000000000 +0200 @@ -59,16 +59,18 @@ static void omap_pcm_dma_irq(int ch, u16 struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if (cpu_is_omap1510()) { + if ((cpu_is_omap1510()) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { /* - * OMAP1510 doesn't support DMA chaining so have to restart - * the transfer after all periods are transferred + * OMAP1510 doesn't fully support DMA progress counter + * and there is no software emulation implemented yet, + * so have to maintain our own playback progress counter + * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); if (prtd->period_index >= 0) { if (++prtd->period_index == runtime->periods) { prtd->period_index = 0; - omap_start_dma(prtd->dma_ch); } } spin_unlock_irqrestore(&prtd->lock, flags); @@ -100,7 +102,7 @@ static int omap_pcm_hw_params(struct snd prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!err && !cpu_is_omap1510()) { + if (!err) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -119,8 +121,7 @@ static int omap_pcm_hw_free(struct snd_p if (prtd->dma_data == NULL) return 0; - if (!cpu_is_omap1510()) - omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); + omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); omap_free_dma(prtd->dma_ch); prtd->dma_data = NULL;

2 1

[alsa-devel] [PATCH] ASoC: UDA134X Codec: Fix mute/unmute code mistake and add ADC/DAC power control support
by Shine Liu 17 Aug '09

17 Aug '09

There is a mistake in current uda134x_mute function: mute_reg has been changed in line 162 or line 164, so uda134x_write should write "mute_reg" but not "mute_reg & ~(1<<2)" to UDA134X_DATA010. Besides, because there is no DAPM configuration for uda134x, when system starts up, snd_soc_int_card calls snd_soc_dapm_new_widgets, and snd_soc_dapm_new_widgets calls dapm_power_widgets. In function dapm_power_widgets, codec->dapm_widgets has no list entry, so sys_power retains it's original value zero. Then snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE) and snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY) are called sequentially. Finally, the uda134x codec goes to STANDBY mode, so the ADC/DAC power control bits of UDA134X_STATUS1 register keeps the 0 value. UDA134X has no trigger function currently, and the ADC/DAC power control bits of UDA134X_STATUS1 register are not exported in uda1341_snd_controls, so there's no way to enable the ADC/DAC power control bits when you want to use the codec. When playing with aplay or recording with arecord, there is no sound output or no wave input. I have added the uda134x_trigger function, which turns on/off the ADC/DAC power control bits according the substream type and the command. I also exported the ADC/DAC power control bits to uda1341_snd_controls, so we can also turn these bits on/off as we need via a mixer tool like alsamixer. The patch created against linux-2.6.31-rc3. Tested on a s3c2440 development board with UDA1341TS codec. Signed-off-by: Shine Liu <shinel(a)foxmail.com> --- sound/soc/codecs/uda134x.c.orig 2009-07-14 09:18:52.000000000 +0800 +++ sound/soc/codecs/uda134x.c 2009-08-17 13:46:57.000000000 +0800 @@ -163,7 +163,7 @@ else mute_reg &= ~(1<<2); - uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + uda134x_write(codec, UDA134X_DATA010, mute_reg); return 0; } @@ -339,6 +339,38 @@ return 0; } +static int uda134x_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int stream = substream->stream; + struct snd_soc_codec *codec = dai->codec; + u8 power_ctrl_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + + pr_debug("%s stream: %d, cmd: %d\n", __func__, stream, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg |= (1<<0); + else + power_ctrl_reg |= (1<<1); + break; + + case SNDRV_PCM_TRIGGER_STOP: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg &= ~(1<<0); + else + power_ctrl_reg &= ~(1<<1); + break; + + default: + return 0; + } + + uda134x_write(codec, UDA134X_STATUS1, power_ctrl_reg); + return 0; +} + static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -416,6 +448,8 @@ SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("ADC Power Switch", UDA134X_STATUS1, 1, 1, 0), +SOC_SINGLE("DAC Power Switch", UDA134X_STATUS1, 0, 1, 0), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; @@ -438,6 +472,7 @@ .digital_mute = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, .set_fmt = uda134x_set_dai_fmt, + .trigger = uda134x_trigger, }; struct snd_soc_dai uda134x_dai = {

1 0

[alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 17 Aug '09

17 Aug '09

I compiled the latest stable ALSA and kernel, and all the utilities. !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Mon Aug 17 06:39:10 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 8.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04" !!DMI Information !!--------------- Manufacturer: Dell Computer Corporation Product Name: Latitude D600 !!Kernel Information !!------------------ Kernel release: 2.6.30.4 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_usb_audio !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 1 [M2 ]: USB-Audio - Mbox 2 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed !!PCI Soundcards installed in the system !!-------------------------------------- 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1f.5 0401: 8086:24c5 (rev 01) Subsystem: 1028:011d !!Modprobe options (Sound related) !!-------------------------------- snd-bt87x: index=-2 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=1 snd-bt87x: index=-2 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_usb_audio async_unlink : Y device_setup : 0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> ignore_ctl_error : N index : 1,-1,-1,-1,-1,-1,-1,-1 nrpacks : 8 pid : -1,-1,-1,-1,-1,-1,-1,-1 vid : -1,-1,-1,-1,-1,-1,-1,-1 !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 32 Aug 17 15:08 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 40 Aug 17 15:08 /dev/snd/midiC1D0 crw-rw----+ 1 root audio 116, 56 Aug 17 16:36 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 48 Aug 17 16:36 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 1 Aug 17 14:50 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 17 14:50 /dev/snd/timer !!ALSA configuration files !!------------------------ !!User specific config file (~/.asoundrc) # ALSA library configuration file # Include settings that are under the control of asoundconf(1). # (To disable these settings, comment out this line.) </home/dczammit/.asoundrc.asoundconf> !!asoundconf-generated config file # ALSA library configuration file managed by asoundconf(1). # # MANUAL CHANGES TO THIS FILE WILL BE OVERWRITTEN! # # Manual changes to the ALSA library configuration should be implemented # by editing the ~/.asoundrc file, not by editing this file. defaults.rawmidi.card 0 defaults.rawmidi.device 0 defaults.rawmidi.subdevice -1 defaults.hwdep.card 0 defaults.hwdep.device 0 defaults.timer.class 2 defaults.timer.sclass 0 defaults.timer.card 0 defaults.timer.device 0 defaults.timer.subdevice 0 defaults.namehint.showall off defaults.namehint.basic on defaults.namehint.extended off !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 1: M2 [Mbox 2], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 1: M2 [Mbox 2], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 1 [M2] Card hw:1 'M2'/'Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed' Mixer name : '' Components : 'USB0dba:3000' Controls : 0 Simple ctrls : 0 !!Alsactl output !!------------- --startcollapse-- state.M2 { control { } } --endcollapse-- jackd 0.116.2 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. loading driver .. apparent rate = 48000 creating alsa driver ... hw:1|hw:1|256|2|48000|2|2|nomon|swmeter|-|32bit control device hw:1 configuring for 48000Hz, period = 256 frames (5.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 24bit little-endian ALSA: use 2 periods for capture ALSA: cannot set hardware parameters for capture ALSA: cannot configure capture channel cannot load driver module alsa [ 6387.449014] WARNING: at sound/usb/usbaudio.c:1314 snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio]() [ 6387.449056] Hardware name: Latitude D600 [ 6387.449059] Modules linked in: snd_usb_audio snd_pcm snd_page_alloc snd_usb_lib snd_hwdep binfmt_misc radeon drm i2c_core rfcomm l2cap bluetooth ppdev sbs sbshc container acpi_cpufreq cpufreq_userspace cpufreq_stats cpufreq_powersave cpufreq_ondemand freq_table cpufreq_conservative ipv6 dm_crypt snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device snd soundcore lp af_packet joydev pcmcia yenta_socket rsrc_nonstatic video output pcmcia_core parport_pc parport psmouse serio_raw button battery ac processor pcspkr intel_agp shpchp agpgart pci_hotplug iTCO_wdt iTCO_vendor_support dcdbas evdev sr_mod sg cdrom sd_mod usbhid hid ata_piix pata_acpi ata_generic tg3 firmware_class libphy libata uhci_hcd ehci_hcd usbcore scsi_mod raid10 raid456 raid6_pq async_xor async_memcpy async_tx xor raid1 raid0 multipath linear md_mod dm_mirror dm_region_hash dm_log dm_snapshot dm_mod thermal fan thermal_sys fuse [ 6387.449148] Pid: 25866, comm: jackd Tainted: G W 2.6.30.4 #1 [ 6387.449151] Call Trace: [ 6387.449165] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449175] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449191] [<c012c03f>] ? warn_slowpath_common+0x7f/0xb0 [ 6387.449201] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449206] [<c012c083>] ? warn_slowpath_null+0x13/0x20 [ 6387.449216] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449237] [<f80eae90>] ? snd_pcm_hw_params_choose+0x30/0xa0 [snd_pcm] [ 6387.449248] [<f80e6fd0>] ? snd_pcm_hw_params+0xe0/0x320 [snd_pcm] [ 6387.449254] [<c0271dce>] ? copy_from_user+0x2e/0x70 [ 6387.449265] [<f80e7346>] ? snd_pcm_hw_params_user+0x36/0x80 [snd_pcm] [ 6387.449276] [<f80e7c06>] ? snd_pcm_common_ioctl1+0x7f6/0x1140 [snd_pcm] [ 6387.449281] [<c0129385>] ? check_preempt_wakeup+0xd5/0x160 [ 6387.449286] [<c01242e9>] ? try_to_wake_up+0xa9/0x200 [ 6387.449292] [<c01b80b5>] ? pollwake+0x45/0x50 [ 6387.449296] [<c0124440>] ? default_wake_function+0x0/0x10 [ 6387.449306] [<c011f0fb>] ? __wake_up_common+0x4b/0x80 [ 6387.449310] [<c01201f5>] ? __wake_up_sync_key+0x45/0x80 [ 6387.449314] [<c01bf8cd>] ? mnt_drop_write+0x1d/0xe0 [ 6387.449319] [<c01bbebb>] ? file_update_time+0x7b/0xc0 [ 6387.449325] [<c01b06ee>] ? pipe_write+0x32e/0x450 [ 6387.449337] [<f80e85de>] ? snd_pcm_capture_ioctl1+0x8e/0x400 [snd_pcm] [ 6387.449348] [<f80e8993>] ? snd_pcm_capture_ioctl+0x43/0x50 [snd_pcm] [ 6387.449359] [<f80e8950>] ? snd_pcm_capture_ioctl+0x0/0x50 [snd_pcm] [ 6387.449364] [<c01b629b>] ? vfs_ioctl+0x2b/0x90 [ 6387.449367] [<c01b65d0>] ? do_vfs_ioctl+0x2d0/0x510 [ 6387.449374] [<c0399075>] ? __schedule+0x235/0x730 [ 6387.449378] [<c01a9b0d>] ? vfs_write+0xfd/0x160 [ 6387.449382] [<c01b6866>] ? sys_ioctl+0x56/0x70 [ 6387.449387] [<c0102dd4>] ? sysenter_do_call+0x12/0x28 [ 6387.449391] ---[ end trace 8fe22d9a0585e3c1 ]---

1 0

Re: [alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 16 Aug '09

16 Aug '09

Using the following patch to usbquirks.h, I was able to get MIDI in and out working on the Digidesign Mbox2!! The capture/playback PCMs are also configured with alsa, but the status on these are "closed". I am guessing I need to snoop the commands on interface 1 (control device) to enable them. Also, there is some "magic" needed to initialise the device which I have captured. It seems that the device needs to be loaded with altsettings = 0 on interfaces 1-5 first, then some setup codes are transmitted, and finally the following altsettings are used. dczammit@flop:/proc/asound$ ls card0 cards devices hwdep M2 modules pcm seq timers version dczammit@flop:/proc/asound$ cat cards 0 [M2 ]: USB-Audio - Mbox 2 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed dczammit@flop:/proc/asound$ cat devices 0: [ 0] : control 1: : sequencer 8: [ 0- 0]: raw midi 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer dczammit@flop:/proc/asound$ cat pcm 00-00: USB Audio : USB Audio : playback 1 : capture 1 dczammit@flop:/proc/asound$ cat timers G0: system timer : 1000.000us (10000000 ticks) G1: RTC timer : 976.562us (100000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE dczammit@flop:/proc/asound/M2$ ls id midi0 pcm0c pcm0p stream0 usbbus usbid dczammit@flop:/proc/asound/M2$ cat midi0 Mbox 2 Output 0 Tx bytes : 212 Input 0 Rx bytes : 314 dczammit@flop:/proc/asound/M2$ cat stream0 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio Playback: Status: Stop Interface 3 Altset 3 Format: 0x20 Channels: 2 Endpoint: 4 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 5 Altset 3 Format: 0x20 Channels: 2 Endpoint: 6 IN (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:/proc/asound/seq$ aplay -Ddefault:CARD=M2 -f S16_LE ~/Desktop/tunes/dzjam.wav Playing WAVE '/home/dczammit/Desktop/tunes/dzjam.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Segmentation fault Please find the quirk below and the magic init codes. ################################## { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { /* Audio control interface - sets the sampling rate? - no idea how this works Possibly also enables the capture/playback using a command? */ .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 3, .altsetting = 3, /* 0 is also available, but has no endpoints */ .altset_idx = 1, .endpoint = 0x04, /* OUT or 0x84 IN (attr 1) */ .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 4, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 5, .altsetting = 3, .altset_idx = 1, .endpoint = 0x86, /* IN */ .ep_attr = 0x0D, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 6, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 2, .out_interval = 0x00, .in_ep = 2, .in_interval = 0x00, .out_cables = 0x0001, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ################################## Audio driver magic: [0 ms] UsbSnoop compiled on Jan 18 2003 22:41:32 loading [0 ms] UsbSnoop - DriverEntry(baca2c40) : Windows NT WDM version 1.32 [101 ms] UsbSnoop - AddDevice(baca2f50) : DriverObject 83499768, pdo 83338c10 [102 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_INTERFACE) [102 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (0x00000018) [127 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_RESOURCE_REQUIREMENTS) [127 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_FILTER_RESOURCE_REQUIREMENTS) [128 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_START_DEVICE) [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 1 going down >>> -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000012 TransferBuffer = bacef95c TransferBufferMDL = 00000000 Index = 00000001 DescriptorType = 00000001 (USB_DEVICE_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=832ff2e8, IRQL=0 [128 ms] <<< URB 1 coming back <<< -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000012 TransferBuffer = bacef95c TransferBufferMDL = 00000000 00000000: 12 01 00 01 00 00 00 08 ba 0d 00 30 38 01 01 02 00000010: 00 01 Index = 00000001 DescriptorType = 00000001 (USB_DEVICE_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 2 going down >>> -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000009 TransferBuffer = bacef9d0 TransferBufferMDL = 00000000 Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=833e2438, IRQL=0 [128 ms] <<< URB 2 coming back <<< -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000009 TransferBuffer = bacef9d0 TransferBufferMDL = 00000000 00000000: 09 02 58 02 05 01 00 80 fa Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 3 going down >>> -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000258 TransferBuffer = 83369540 TransferBufferMDL = 00000000 Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=83455a90, IRQL=0 [128 ms] <<< URB 3 coming back <<< -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000258 TransferBuffer = 83369540 TransferBufferMDL = 00000000 00000000: 09 02 58 02 05 01 00 80 fa 09 04 01 00 00 01 01 00000010: 00 02 0c 24 01 00 01 60 00 04 02 03 04 05 0c 24 00000020: 02 05 01 01 00 02 03 00 00 00 09 24 03 06 01 06 00000030: 00 05 00 0c 24 02 07 01 01 00 02 03 00 00 00 09 00000040: 24 03 08 01 06 00 07 00 0c 24 02 01 01 06 00 02 00000050: 03 00 00 00 09 24 03 02 01 01 00 01 00 0c 24 02 00000060: 03 01 06 00 02 03 00 00 00 09 24 03 04 01 01 00 00000070: 03 00 09 04 02 00 00 01 02 00 00 09 04 02 02 02 00000080: 01 02 00 00 07 24 01 05 01 01 00 0e 24 02 01 02 00000090: 03 18 00 40 1f 00 80 bb 00 09 05 03 05 28 01 01 000000a0: 00 83 07 25 01 00 00 00 00 09 05 83 01 03 00 01 000000b0: 02 00 09 04 02 03 02 01 02 00 00 07 24 01 05 01 000000c0: 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 000000d0: 09 05 03 05 28 01 01 00 83 07 25 01 00 00 00 00 000000e0: 09 05 83 01 03 00 01 02 00 09 04 03 00 00 01 02 000000f0: 00 00 09 04 03 02 02 01 02 00 00 07 24 01 07 01 00000100: 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 00000110: 09 05 04 05 28 01 01 00 84 07 25 01 00 00 00 00 00000120: 09 05 84 01 03 00 01 02 00 09 04 03 03 02 01 02 00000130: 00 00 07 24 01 07 01 01 00 0e 24 02 01 02 03 18 00000140: 00 40 1f 00 80 bb 00 09 05 04 05 28 01 01 00 84 00000150: 07 25 01 00 00 00 00 09 05 84 01 03 00 01 02 00 00000160: 09 04 03 06 01 01 02 00 00 07 24 01 07 01 01 20 00000170: 0e 24 02 03 02 02 10 00 40 1f 00 80 bb 00 09 05 00000180: 04 0d c0 00 01 00 00 07 25 01 01 00 00 00 09 04 00000190: 04 00 00 01 02 00 00 09 04 04 02 01 01 02 00 00 000001a0: 07 24 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 000001b0: 1f 00 80 bb 00 09 05 85 05 28 01 01 00 00 07 25 000001c0: 01 00 00 00 00 09 04 04 03 01 01 02 00 00 07 24 000001d0: 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 000001e0: 80 bb 00 09 05 85 0d 28 01 01 00 00 07 25 01 01 000001f0: 00 00 00 09 04 05 00 00 01 02 00 00 09 04 05 02 00000200: 01 01 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 00000210: 02 03 18 00 40 1f 00 80 bb 00 09 05 86 05 28 01 00000220: 01 00 00 07 25 01 00 00 00 00 09 04 05 03 01 01 00000230: 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 02 03 00000240: 18 00 40 1f 00 80 bb 00 09 05 86 0d 28 01 01 00 00000250: 00 07 25 01 01 00 00 00 Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 4 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [133 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=833573c0, IRQL=2 [133 ms] <<< URB 4 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83623978 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [433 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=83436508, IRQL=0 [433 ms] >>> URB 5 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [437 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=83436508, Context=83429560, IRQL=2 [437 ms] <<< URB 5 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 834a08e8 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [733 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=83436508, IRQL=0 [733 ms] >>> URB 6 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [738 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=83436508, Context=83625f08, IRQL=2 [738 ms] <<< URB 6 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 834a08e8 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1034 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=834a1398, IRQL=0 [1034 ms] >>> URB 7 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1038 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=834a1398, Context=832ff2e8, IRQL=2 [1038 ms] <<< URB 7 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 834a08e8 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1334 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [1334 ms] >>> URB 8 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1339 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [1339 ms] <<< URB 8 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1635 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [1635 ms] >>> URB 9 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1639 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832ff2e8, IRQL=2 [1639 ms] <<< URB 9 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1935 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [1935 ms] >>> URB 10 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1940 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [1940 ms] <<< URB 10 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [2236 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [2236 ms] >>> URB 11 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [2240 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832ff2e8, IRQL=2 [2240 ms] <<< URB 11 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [2536 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [2536 ms] >>> URB 12 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [2541 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [2541 ms] <<< URB 12 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [2837 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [2837 ms] >>> URB 13 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [2841 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832ff2e8, IRQL=2 [2841 ms] <<< URB 13 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [3137 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3137 ms] >>> URB 14 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [3142 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [3142 ms] <<< URB 14 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 02 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [3142 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3142 ms] >>> URB 15 going down >>> -- URB_FUNCTION_SELECT_CONFIGURATION: ConfigurationDescriptor = 0x83369540 (configure) ConfigurationDescriptor : bLength = 9 ConfigurationDescriptor : bDescriptorType = 0x00000002 ConfigurationDescriptor : wTotalLength = 0x00000258 ConfigurationDescriptor : bNumInterfaces = 0x00000005 ConfigurationDescriptor : bConfigurationValue = 0x00000001 ConfigurationDescriptor : iConfiguration = 0x00000000 ConfigurationDescriptor : bmAttributes = 0x00000080 ConfigurationDescriptor : MaxPower = 0x000000fa ConfigurationHandle = 0x00000000 Interface[0]: Length = 16 Interface[0]: InterfaceNumber = 1 Interface[0]: AlternateSetting = 0 Interface[1]: Length = 16 Interface[1]: InterfaceNumber = 2 Interface[1]: AlternateSetting = 0 Interface[2]: Length = 16 Interface[2]: InterfaceNumber = 3 Interface[2]: AlternateSetting = 0 Interface[3]: Length = 16 Interface[3]: InterfaceNumber = 4 Interface[3]: AlternateSetting = 0 Interface[4]: Length = 16 Interface[4]: InterfaceNumber = 5 Interface[4]: AlternateSetting = 0 [3142 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=83429560, IRQL=0 [3142 ms] <<< URB 15 coming back <<< -- URB_FUNCTION_SELECT_CONFIGURATION: ConfigurationDescriptor = 0x83369540 (configure) ConfigurationDescriptor : bLength = 9 ConfigurationDescriptor : bDescriptorType = 0x00000002 ConfigurationDescriptor : wTotalLength = 0x00000258 ConfigurationDescriptor : bNumInterfaces = 0x00000005 ConfigurationDescriptor : bConfigurationValue = 0x00000001 ConfigurationDescriptor : iConfiguration = 0x00000000 ConfigurationDescriptor : bmAttributes = 0x00000080 ConfigurationDescriptor : MaxPower = 0x000000fa ConfigurationHandle = 0x835a8978 Interface[0]: Length = 16 Interface[0]: InterfaceNumber = 1 Interface[0]: AlternateSetting = 0 Interface[0]: Class = 0x00000001 Interface[0]: SubClass = 0x00000001 Interface[0]: Protocol = 0x00000000 Interface[0]: InterfaceHandle = 0x8336b868 Interface[0]: NumberOfPipes = 0 Interface[1]: Length = 16 Interface[1]: InterfaceNumber = 2 Interface[1]: AlternateSetting = 0 Interface[1]: Class = 0x00000000 Interface[1]: SubClass = 0x00000000 Interface[1]: Protocol = 0x00000000 Interface[1]: InterfaceHandle = 0x00000000 Interface[1]: NumberOfPipes = 0 Interface[2]: Length = 16 Interface[2]: InterfaceNumber = 3 Interface[2]: AlternateSetting = 0 Interface[2]: Class = 0x00000000 Interface[2]: SubClass = 0x00000000 Interface[2]: Protocol = 0x00000000 Interface[2]: InterfaceHandle = 0x00000000 Interface[2]: NumberOfPipes = 0 Interface[3]: Length = 16 Interface[3]: InterfaceNumber = 4 Interface[3]: AlternateSetting = 0 Interface[3]: Class = 0x00000000 Interface[3]: SubClass = 0x00000000 Interface[3]: Protocol = 0x00000000 Interface[3]: InterfaceHandle = 0x00000000 Interface[3]: NumberOfPipes = 0 Interface[4]: Length = 16 Interface[4]: InterfaceNumber = 5 Interface[4]: AlternateSetting = 0 Interface[4]: Class = 0x00000000 Interface[4]: SubClass = 0x00000000 Interface[4]: Protocol = 0x00000000 Interface[4]: InterfaceHandle = 0x00000000 Interface[4]: NumberOfPipes = 0 [3142 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3142 ms] >>> URB 16 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 56 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 0 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3145 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832f0090, IRQL=0 [3145 ms] <<< URB 16 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 16 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 0 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83621580 Interface: NumberOfPipes = 0 [3145 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3145 ms] >>> URB 17 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 56 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 3 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3187 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=83539d80, IRQL=0 [3187 ms] <<< URB 17 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 56 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 3 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 833b9d68 Interface: NumberOfPipes = 2 Interface: Pipes[0] : MaximumPacketSize = 0x00000128 Interface: Pipes[0] : EndpointAddress = 0x00000003 Interface: Pipes[0] : Interval = 0x00000001 Interface: Pipes[0] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[0] : PipeHandle = 0x833b9d84 Interface: Pipes[0] : MaxTransferSize = 0x00000000 Interface: Pipes[0] : PipeFlags = 0x00000000 Interface: Pipes[1] : MaximumPacketSize = 0x00000003 Interface: Pipes[1] : EndpointAddress = 0x00000083 Interface: Pipes[1] : Interval = 0x00000001 Interface: Pipes[1] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[1] : PipeHandle = 0x833b9da4 Interface: Pipes[1] : MaxTransferSize = 0x00000000 Interface: Pipes[1] : PipeFlags = 0x00000000 [3187 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3187 ms] >>> URB 18 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 0 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3190 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=835fced8, IRQL=0 [3190 ms] <<< URB 18 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 16 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 0 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83621580 Interface: NumberOfPipes = 0 [3190 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3190 ms] >>> URB 19 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 3 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3207 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832f0090, IRQL=0 [3207 ms] <<< URB 19 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 36 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 3 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 834a36c8 Interface: NumberOfPipes = 1 Interface: Pipes[0] : MaximumPacketSize = 0x00000128 Interface: Pipes[0] : EndpointAddress = 0x00000085 Interface: Pipes[0] : Interval = 0x00000001 Interface: Pipes[0] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[0] : PipeHandle = 0x834a36e4 Interface: Pipes[0] : MaxTransferSize = 0x00000000 Interface: Pipes[0] : PipeFlags = 0x00000000 [3207 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3207 ms] >>> URB 20 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 0 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3210 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=83539d80, IRQL=0 [3210 ms] <<< URB 20 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 16 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 0 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83621580 Interface: NumberOfPipes = 0 [3210 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3210 ms] >>> URB 21 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 3 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3227 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=835fced8, IRQL=0 [3227 ms] <<< URB 21 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 36 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 3 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83631b58 Interface: NumberOfPipes = 1 Interface: Pipes[0] : MaximumPacketSize = 0x00000128 Interface: Pipes[0] : EndpointAddress = 0x00000086 Interface: Pipes[0] : Interval = 0x00000001 Interface: Pipes[0] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[0] : PipeHandle = 0x83631b74 Interface: Pipes[0] : MaxTransferSize = 0x00000000 Interface: Pipes[0] : PipeFlags = 0x00000000 [3228 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_CAPABILITIES) [3228 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_PNP_DEVICE_STATE) [3228 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_DEVICE_RELATIONS) [3288 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_DEVICE_RELATIONS) Audio driver "magic" successful response: usb_get_driver_np returned -61 1 get descriptor returned 18, bytes: 12 01 00 01 00 00 00 08 ba 0d 00 30 38 01 01 02 00 01 "...........08....." 2 get descriptor returned 9, bytes: 09 02 86 02 07 01 00 80 fa "........." 3 get descriptor returned 600, bytes: 09 02 86 02 07 01 00 80 fa 09 04 00 00 00 fe 01 00 00 07 21 05 ff ff 08 00 09 04 01 00 00 01 01 00 02 0c 24 01 00 01 60 00 04 02 03 04 05 0c 24 02 05 01 01 00 02 03 00 00 00 09 24 03 06 01 06 00 05 00 0c 24 02 07 01 01 00 02 03 00 00 00 09 24 03 08 01 06 00 07 00 0c 24 02 01 01 06 00 02 03 00 00 00 09 24 03 02 01 01 00 01 00 0c 24 02 03 01 06 00 02 03 00 00 00 09 24 03 04 01 01 00 03 00 09 04 02 00 00 01 02 00 00 09 04 02 02 02 01 02 00 00 07 24 01 05 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 03 05 28 01 01 00 83 07 25 01 00 00 00 00 09 05 83 01 03 00 01 02 00 09 04 02 03 02 01 02 00 00 07 24 01 05 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 03 05 28 01 01 00 83 07 25 01 00 00 00 00 09 05 83 01 03 00 01 02 00 09 04 03 00 00 01 02 00 00 09 04 03 02 02 01 02 00 00 07 24 01 07 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 04 05 28 01 01 00 84 07 25 01 00 00 00 00 09 05 84 01 03 00 01 02 00 09 04 03 03 02 01 02 00 00 07 24 01 07 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 04 05 28 01 01 00 84 07 25 01 00 00 00 00 09 05 84 01 03 00 01 02 00 09 04 03 06 01 01 02 00 00 07 24 01 07 01 01 20 0e 24 02 03 02 02 10 00 40 1f 00 80 bb 00 09 05 04 0d c0 00 01 00 00 07 25 01 01 00 00 00 09 04 04 00 00 01 02 00 00 09 04 04 02 01 01 02 00 00 07 24 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 85 05 28 01 01 00 00 07 25 01 00 00 00 00 09 04 04 03 01 01 02 00 00 07 24 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 85 0d 28 01 01 00 00 07 25 01 01 00 00 00 09 04 05 00 00 01 02 00 00 09 04 05 02 01 01 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 86 05 28 01 01 00 00 07 25 01 00 00 00 00 09 04 05 03 01 01 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 "...................!...............$...`.......$...........$........$...........$........$...........$........$...........$..........................$......$......@.........(.....%........................$......$......@.........(.....%.................................$......$......@.........(.....%........................$......$......@.........(.....%........................$.... .$......@...............%........................$......$......@.........(.....%...............$......$......@.........(.....%........................$......$......@.........(.....%...............$......$......@....." 4 control msg returned 1, bytes: 01 "." 5 control msg returned 1, bytes: 01 "." 6 control msg returned 1, bytes: 01 "." 7 control msg returned 1, bytes: 01 "." 8 control msg returned 1, bytes: 01 "." 9 control msg returned 1, bytes: 01 "." 10 control msg returned 1, bytes: 01 "." 11 control msg returned 1, bytes: 01 "." 12 control msg returned 1, bytes: 01 "." 13 control msg returned 1, bytes: 01 "." 14 control msg returned 1, bytes: 02 "." 15 set configuration returned 0 15 set alternate setting returned 0 16 set alternate setting returned 0 17 set alternate setting returned -22 18 set alternate setting returned 0 19 set alternate setting returned -22 20 set alternate setting returned 0 21 set alternate setting returned -22 I need help! How can I enable the capture/playback?

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[alsa-devel] snd_pcm_drain() and O_NONBLOCK
by Jon Smirl 16 Aug '09

16 Aug '09

I'm trying to fix brutefir to properly drain the driver before exiting. This sequence does not work on a running stream. snd_pcm_nonblock() removes the O_NONBLOCK flag from the file handle but the change does not get copied to stream->f_flags. snd_pcm_drain() returns -EAGAIN since the flags have not be copied. snd_pcm_nonblock(handles[IO][n], 0); snd_pcm_drain(handles[IO][n]); snd_pcm_nonblock(handles[IO][n], SND_PCM_NONBLOCK); static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { printk("snd_pcm_pre_drain_init flags %x\n", substream->f_flags); if (substream->f_flags & O_NONBLOCK) return -EAGAIN; printk("snd_pcm_pre_drain_init 1\n"); substream->runtime->trigger_master = substream; return 0; } Searching through the ALSA code the only way f_flags gets copied from the file handle to the stream handle is via snd_pcm_prepare(). But I can't do a snd_pcm_prepare() on a running stream. I can't find any way to change the NONBLOCK status on a running stream. I also tried implementing the wait for drain the the user app. snd_pcm_drain() always returns -EAGAIN of O_NONBLOCK is set. It can't be used to poll for the drain status. So how do I wait for a stream to drain? -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 16 Aug '09

16 Aug '09

Hi, I wrote an addition to usbquirks.h to provide support for "Digidesign Mbox 2" but I need someone to point me in the right direction as to which interfaces to use. I'm not sure if the MIDI interface is a standard device, and I'm not sure which interfaces to ignore. I am about to compile a new alsa + kernel with my changes to usbquirks.h to test the device. Here is my usbquirks.h addition: /* DIGIDESIGN MBOX 2 addition to usbquirks.h */ { /* Damien Zammit <damien.zammit(a)gmail.com> * 16 August 2009 * * Interface topology of Digidesign Mbox 2, it has 7 interfaces: * 0 - Proprietary firmware updates * 1 - Audio control device * 2 - Double endpoint IN-OUT PCM * 3 - Double endpoint OUT-IN PCM * 4 - Single endpoint IEC1937 AC3 / PCM (selectable by altsetting) * 5 - Single endpoint PCM (supports full 24bit 48000Hz) * 6 - Triple endpoint MIDI (Interrupt + 1 IN + 1 OUT) * * As I am unsure which interfaces to try, I have provided _all_ the settings * I can in this file to hand to the usb-audio developers. * I have no idea what .altset_idx does or where to find it, so I set it to 1 in all cases. * * I am currently testing with just interfaces 5 and 6 (not sure if it will work) */ USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { /* Proprietary firmware updater interface - ignore for now */ { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, /* Audio control interface - altsetting = 0 not sure what to do with this */ { .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 2, .altsetting = 3, /* 0 is also available but has no endpoints */ .altset_idx = 1, .endpoint = 0x03, /* IN or 0x83 OUT */ .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 3, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 3, .altsetting = 3, /* 0 is also available, but has no endpoints */ .altset_idx = 1, .endpoint = 0x04, /* OUT or 0x84 IN */ .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = <<<<< Type III - IEC1937_AC3 (16 bit) >>>>> .channels = 2, .iface = 3, .altsetting = 6, .altset_idx = 1, .endpoint = 0x04, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 5, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 5, .altsetting = 3, .altset_idx = 1, .endpoint = 0x86, /* Single endpoint only */ .ep_attr = 0x0D, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { /* MIDI INTERFACE DESCRIPTION * Is this a standard USB MIDI device? * It has 2 Interface descriptors for the same altsetting... Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 6 bAlternateSetting 0 bNumEndpoints 3 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 6 bAlternateSetting 0 bNumEndpoints 3 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 */ .ifnum = 6, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { /* I am not certain of the "cables" flags or how * to find out what they are. */ .out_ep = 2, .out_interval = 0x00, .in_ep = 2, .in_interval = 0x00, .out_cables = 0x0001, .in_cables = 0x0001 } }, { .ifnum = -1 } } } },

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Re: [alsa-devel] [PATCH] ALSA: hda: partially revert ALC268 quirk collapse
by Takashi Iwai 15 Aug '09

15 Aug '09

At Sat, 15 Aug 2009 16:38:46 +0400, Alexey Dobriyan wrote: > > Fix alsamixer inability to start: > > ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Master Playback Switch',0,3,0) appears twice or more > > ALC268 codec shipped with Toshiba Satellite X200 has ids 1179:ff00 and > began to match quirk table. > > If it doesn't match and auto-probe from BIOS is used, alsamixer works: > > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > > 8871e5b91518a47284b6bc2603b44dbc79c85446 is first bad commit > commit 8871e5b91518a47284b6bc2603b44dbc79c85446 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Jun 2 01:02:50 2009 +0200 > > ALSA: hda - Reorder and clean-up ALC268 quirk table > > Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some > entries using SND_PCI_QUIRK_MASK(). > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > :040000 040000 2497bb6fc5305acf3191b8e9b82956ec0ea1aee9 6f1d9ff9fed20550a1b2b398c7d2d5fec705fcd9 M sound > > Signed-off-by: Alexey Dobriyan <adobriyan(a)gmail.com> Well, it's no right fix. The symptom implies that something wrong with the current model=toshiba, so fixing the quirk is the way to go at this moment. Could you give alsa-info.sh output (run with --no-upload option) as vanilla kernel? Run with both model=auto and without model option and attach both results to compare. The former means actually the condition as your patch does. thanks, Takashi

1 0

[alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 15 Aug '09

15 Aug '09

I have a Digidesign Mbox2 and I am trying to write an open source alsa driver for it. It is a usb self powered device, and I used usbsnoop on the Windows XP drivers to capture the initialisation of the device. It doesn't seem to require any firmware, as I was able to playback the init code to the device over libusb and all the lights turned on (which hopefully means it's awaiting further instructions to play/record). I have written some C code which sends the init codes to the device using raw URB, but I really need someone who is familiar with snd-usb-audio to take a look at my lsusb output in case it is a standard device before I waste any more time on it. Please find the debugging output for my experimental initialisation driver here: http://cytdesign.com/mbox2/mbox2.tar.bz2 Note: After running my init codes, all the lights appeared on the device just as it does when you boot windows. Please contact me at damien.zammit [at] gmail.com if you are able to help. I'm not up to speed with mailing lists, I would prefer an email to my account if you are replying to me. Thanks, Damien Zammit

1 0

[alsa-devel] snd_pcm_delay, hw buffers and driver api (v2)
by Kai Vehmanen 15 Aug '09

15 Aug '09

Hi, as I got at least one reply, reposting this in a new thread to catch more people. Originally sent to alsa-devel thread: "[PATCH 11/20] OMAP: McBSP: Add link DMA mode selection)" http://kerneltrap.org/mailarchive/alsa-devel/2009/7/30/6272573 --- So not strictly related to the patch that started the original thread, but this touches on, and is a good example of, one question I've been wondering for some time now as an app developer. Could Takashi, Jaroslav, Mark, or others comment on this as well, perhaps? Dropping linux-omap as this is a generic ALSA question. On Wed, 12 Aug 2009, Jarkko Nikula wrote: [i.e. pcm_pointer == hw_ptr] > The threshold based transfer will cause that omap_pcm_pointer will > loose a bit its accuracy. Probably irrelevant but still better to play > safe at least over one kernel release before making it default. [...] > element: > 614 > 669 > 691 [...] > threshold: > 512 > 512 > 1024 > 1024 In both cases, this value (hw_ptr) shows the status of sending data to an separate entity, and one that has its own buffer (multiple ms) before the actual DAC/ADC. So when application developer uses snd_pcm_delay(), the result is not quite what the ALSA API says: alsa-lib.git/src/pcm/pcm.c: --cut-- * For playback the delay is defined as the time that a frame that is written * to the PCM stream shortly after this call will take to be actually * audible. It is as such the overall latency from the write call to the final * DAC. [...] int snd_pcm_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) --cut-- Of course, there are many other cases where the same happens, as the latency of the HW (or where it is connected) is not known. But we also have many cases where we _do_ know more about the latency, and we are just missing the means to expose this info. I think this applies to this OMAP patch as well. E.g. the HW can tell fairly accurately the current status (i.e. which sample is played-out/captured _now_). Now that brings us to my question -- how to expose this info in an ALSA driver. Elsewhere in alsa-lib docs, there is more accurate description of current implementation and behaviour of snd_pcm_delay(): alsa-lib.git/src/pcm/pcm.c: --cut-- The function #snd_pcm_delay() returns the delay in samples. For playback, it means count of samples in the ring buffer before the next sample will be sent to DAC. --cut-- So e.g. the OMAP patch that this thread started from, complies with this definition, and element/threshold modes just differ in accuracy (although threshold implementation should warn the application with the SNDRV_PCM_INFO_BATCH flag). But how would a driver expose both bits of info to apps (in a standard fashion): 1) the hw_ptr for shuffling bits out of the ringbuffer, and 2) the delay to next played-out/captured sample. For application developers, (2) is the piece of info we are mostly interested in (if I write a sample now, how long it will be until it's played out). One obvious solution (that has been used already in other ALSA drivers) is to virtualize the hw_ptr, but this way you lose the accurate info about ringbuffer status. Ideally both pieces of info could be exposed. But as I see it, drivers currently can only implement "pointer" to relay this info. Jaroslav proposed some ideas earlier this year, but then the discussion faded: "Re: Driver code with mpc5200 pointer problem.", 2009-04-28 http://article.gmane.org/gmane.linux.alsa.devel/62170 Takashi made basicly the same point I've tried to make above "Re: Driver code with mpc5200 pointer problem.", 2009-04-28 http://article.gmane.org/gmane.linux.alsa.devel/62172 Mark also commented to the same thread: http://article.gmane.org/gmane.linux.alsa.devel/62176, 2009-04-28 Any update/ideas on this topic? Personally I think adding a new driver callback would make most sense, as that would allow to take full benefit from hardware that allows to query sample-accurate position of playout (i.e. not just support exposing a fixed latency). Of course that's potentially a big change. In alsa-lib, snd_pcm_hwsync() could call this driver callback, and a new variant of snd_pcm_delay() could present the information to the applications.

3 5

[alsa-devel] [PATCH] delete -spi suffix in name and free private data while registers fail
by Barry Song 14 Aug '09

14 Aug '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad1938.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 051f68c..e62b277 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,7 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { - .name = "ad1938-spi", + .name = "ad1938", .bus = &spi_bus_type, .owner = THIS_MODULE, }, @@ -543,6 +543,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad1938); return ret; } @@ -550,6 +551,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); + kfree(ad1938); return ret; } -- 1.5.6.3

2 1

[alsa-devel] [PATCH 0/1 - v2] ASoC: TWL4030: Capture mixer string changes
by Peter Ujfalusi 14 Aug '09

14 Aug '09

Hello, Resending with the suggested changes: Before the patch the analog capture path strings ended up like this at the end: "Analog Left Capture Route Main mic Switch" These switches showed up in the playback tab of alsamixer, which was not nice. After the patch the strings will be like this: "Analog Left Main Mic Capture Switch" Which show up correctly in alsamixer as "Analog Left Main mic" under the capture tab. --- Peter Ujfalusi (1): ASoC: TWL4030: Fix for capture mixer strings sound/soc/codecs/twl4030.c | 46 +++++++++++++++++++++++-------------------- 1 files changed, 25 insertions(+), 21 deletions(-)

2 2

[alsa-devel] [PATCH] Allow passing platform_data to devices attached to AC97 bus
by Marek Vasut 14 Aug '09

14 Aug '09

Hi, I managed to put together this patch with huge help from Mark. It allows passing platform_data to devices on ac97 bus and implements this for pxa2xx- ac97. Please consider applying.

6 51

Re: [alsa-devel] [PATCHv3 14/20] OMAP: McBSP: Let element DMA mode hit retention also
by Eero Nurkkala 14 Aug '09

14 Aug '09

On Fri, 2009-08-14 at 05:19 +0200, ext Woodruff, Richard wrote: > > From: linux-omap-owner(a)vger.kernel.org [mailto:linux-omap- > > owner(a)vger.kernel.org] On Behalf Of Eduardo Valentin > > > The device no longer hits retention if element DMA > > mode is taken for at least the duration of the > > serial console timeout. Force element DMA mode to > > shut down through smartidle. > > Probably this behaves as you say but it seems possible that the bug is somewhere else in device handling. > For some reason unknown to us, if NO_IDLE mode is taken, and long enough of a arecord | aplay loop was tried (a minute or so), then, and when closing the loop, the device never hit retention anymore =) But after the fix, yes it does so. > The generic hope is you can set smart-idle + wakeups at init time and never need to touch them again. There are a few known cases where the hardware IP does have some bug which force you to dynamically play with bits. > Right. I did try to set them initially at device init (that was one of the first things I tried with McBSP), but things were not really working as expected. (Well, having iclk and fclk on and putting those at device init caused the whole HW device to jam). Another thing is, that if the McBSP is a slave (external clocking), then there will be unnecessary wakeups? If smart-idle + wakeups are always set? (and the slave clocks the McBSP WCLK and BCLK). BTW, are the wakeups active if McBSP is in reset state? > Clockactivity settings are usually static also, but depending on how you configure your device (slave or master clocking) you will need to dynamically set these. > It most you don't need all this dynamic twiddling of the idle protocol handshake bits. Doing it once at pm_init is what should be necessary. > One example which comes to mind is MMC with DMA mode. Early drivers were finding RET was gated once the driver was active. In the first pass port to 3430 the writer toggled smart/no/forced idle and the system went happily back to sleep. However, after some profiling it was seen that this method of putting the device down ended up causing the clock disable to the device to take a very long time. Another level of digging into documents was done and it was found that it was necessary to reset some command line as part of a clean shut down. Doing it this way allowed the system to idle, and the time to shut down clocks was just a few nS instead of uS. As I recall some status was not cleared until the reset of the line. It did happen that toggling through handshakes had a side effect of also clearing it, but not in the intended way. Whether a few uSec:s or nS:s, doesn't really matter in this case? (if that even was the case). > > Regards, > Richard W. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 13 Aug '09

13 Aug '09

Hellos guys, This series provides a way to allow DMA go to idle while performing playback/capture in OMAP3 devices using ASoC driver. The idea behind it is to do not use DMA element mode, but frame mode. In addition to that, it also configures McBSP threshold accordingly to alsa period. Also, McBSP wakeup signals are configured to generate wakeups for threshold events. Besides that, this series also include a way to allow choosing between element mode or the frame mode added here. Patches 1-15 are McBSP changes and, for this reason, are intended to be include in linux-omap tree. Patches 16-20 are OMAP ASoC driver changes, so, they are for alsa-devel. BR, Eduardo Valentin (11): OMAP: McBSP: Change wakeup signals OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold again ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Disable all wakeups upon exit OMAP: McBSP: Let element DMA mode hit retention also ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 54 +++++ arch/arm/plat-omap/mcbsp.c | 343 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 16 ++- sound/soc/omap/omap-pcm.c | 57 +++++- 5 files changed, 468 insertions(+), 7 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

8 66

[alsa-devel] [PATCH v2 1/2] ASoC: DaVinci: McASP driver enhacements
by Chaithrika U S 13 Aug '09

13 Aug '09

On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO support. This FIFO provides additional data buffering. It also provides tolerance to variation in host/DMA controller response times. The read and write FIFO sizes are 256 bytes each. If FIFO is enabled, the DMA events from McASP are sent to the FIFO which in turn sends DMA requests to the host CPU according to the thresholds programmed. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber= sprufm1&fileType=pdf This patch adds support for FIFO configuration. The platform data has a version field which differentiates the McASP on different SoCs. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- In this version of the patch, the davinci-i2s driver changes have been done to account acnt member introduced in the dma_params structure. Also, the DAI format addition has been removed from this patch. 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-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 105 ++++++++++++++++++++++++++++++++++--- sound/soc/davinci/davinci-mcasp.h | 5 ++ sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-pcm.h | 1 + 5 files changed, 107 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 2a56fb7..12a6c54 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -406,6 +406,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + dma_params->acnt = dma_params->data_type; rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f0c0347..e672f43 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -102,6 +102,11 @@ /* Receive Buffer for Serializer n */ #define DAVINCI_MCASP_RXBUF_REG 0x280 +/* McASP FIFO Registers */ +#define DAVINCI_MCASP_WFIFOCTL (0x1010) +#define DAVINCI_MCASP_WFIFOSTS (0x1014) +#define DAVINCI_MCASP_RFIFOCTL (0x1018) +#define DAVINCI_MCASP_RFIFOSTS (0x101C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -276,6 +281,13 @@ */ #define TXDATADMADIS BIT(0) +/* + * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits + */ +#define FIFO_ENABLE BIT(16) +#define NUMEVT_MASK (0xFF << 8) +#define NUMDMA_MASK (0xFF) + #define DAVINCI_MCASP_NUM_SERIALIZER 16 static inline void mcasp_set_bits(void __iomem *reg, u32 val) @@ -345,6 +357,9 @@ static void mcasp_start_rx(struct davinci_audio_dev *dev) static void mcasp_start_tx(struct davinci_audio_dev *dev) { + u8 offset = 0, i; + u32 cnt; + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); @@ -353,6 +368,19 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) { + offset = i; + break; + } + } + + /* wait for TX ready */ + cnt = 0; + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + TXSTATE) && (cnt < 100000)) + cnt++; + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); } @@ -362,6 +390,13 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) mcasp_start_tx(dev); else mcasp_start_rx(dev); + + /* enable FIFO */ + if (dev->txnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -382,6 +417,13 @@ static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) mcasp_stop_tx(dev); else mcasp_stop_rx(dev); + + /* disable FIFO */ + if (dev->txnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -401,7 +443,6 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); break; - case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); @@ -505,6 +546,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) { int i; + u8 tx_ser = 0; + u8 rx_ser = 0; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -525,12 +568,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) + if (dev->serial_dir[i] == TX_MODE) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); - else if (dev->serial_dir[i] == RX_MODE) + tx_ser++; + } else if (dev->serial_dir[i] == RX_MODE) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); + rx_ser++; + } + } + + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt * tx_ser > 64) + dev->txnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + + if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (dev->rxnumevt * rx_ser > 64) + dev->rxnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } @@ -543,6 +611,8 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < active_slots; i++) mask |= (1 << i); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ @@ -622,8 +692,16 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; int word_length; + u8 numevt; davinci_hw_common_param(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + numevt = dev->txnumevt; + else + numevt = dev->rxnumevt; + + if (!numevt) + numevt = 1; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -650,6 +728,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); return -EINVAL; } + + if (dev->version == MCASP_VERSION_2) { + dma_params->data_type *= numevt; + dma_params->acnt = 4 * numevt; + } else + dma_params->acnt = dma_params->data_type; + davinci_config_channel_size(dev, word_length); return 0; @@ -778,6 +863,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; dev->codec_fmt = pdata->codec_fmt; + dev->version = pdata->version; + dev->txnumevt = pdata->txnumevt; + dev->rxnumevt = pdata->rxnumevt; dma_data[count].name = "I2S PCM Stereo out"; dma_data[count].eventq_no = pdata->eventq_no; @@ -807,9 +895,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } dma_data[count].channel = res->start; - davinci_mcasp_dai[pdev->id].private_data = dev; - davinci_mcasp_dai[pdev->id].dev = &pdev->dev; - ret = snd_soc_register_dai(&davinci_mcasp_dai[pdev->id]); + davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) goto err_release_region; @@ -827,12 +915,13 @@ err_release_dev: static int davinci_mcasp_remove(struct platform_device *pdev) { + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_pcm_dma_params *dma_data; struct davinci_audio_dev *dev; struct resource *mem; - snd_soc_unregister_dai(&davinci_mcasp_dai[pdev->id]); - dev = davinci_mcasp_dai[pdev->id].private_data; + snd_soc_unregister_dai(&davinci_mcasp_dai[pdata->op_mode]); + dev = davinci_mcasp_dai[pdata->op_mode].private_data; clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 36b7104..554354c 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -50,6 +50,11 @@ struct davinci_audio_dev { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; }; #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ab43a53..091dacb 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -67,6 +67,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; unsigned int data_type; + unsigned short acnt; unsigned int count; period_size = snd_pcm_lib_period_bytes(substream); @@ -91,11 +92,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } + acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); edma_set_src_index(lch, src_bidx, 0); edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index eb4287f..63d9625 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -19,6 +19,7 @@ struct davinci_pcm_dma_params { char *name; /* stream identifier */ int channel; /* sync dma channel ID */ + unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ -- 1.5.6

2 1

[alsa-devel] [RFC] Break up the patch_sigmatel.c
by Maxim Levitsky 13 Aug '09

13 Aug '09

Hi folks, I own an stac9227 and with latest kernel, it doesn't output any sound via front headphones. (this a is desktop) I don't want to bother you with details, etc, I will fix that myself. I have good knowledge of the device. However, the mess that is patch_sigmatel.c seems to be doubled. its huge file, and there are loads on new codecs added. I have a suggestion, to break it up into smaller files. I especially want to move all 'data' from it to header file, and sort it by model. Maybe move pin configs to separate file. Also I know that stac9200 is quite old, and uses many different support functions. I could move these in another file as well. In other words, I want to break it up into several (5 maybe) files, while not touching the code itself (I so that later maybe) Since it is quite dirty and hard work, I want to ask if you agree. Btw, half of static data can be retrieved from codec by querying it (like dac,adc,pin node ids) Best regards, Maxim Levitsky

3 6

[alsa-devel] [PATCH V2 1/3] new ad1836 codec driver based on asoc
by Barry Song 13 Aug '09

13 Aug '09

There has been an ad1836 driver in sound/blackfin based on traditional alsa. The new driver is based on asoc. The architecture of ad1836 codec driver is very much like ad1938. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 451 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1836.h | 64 +++++++ 4 files changed, 520 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ad1836.c create mode 100644 sound/soc/codecs/ad1836.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 31a6d21..05e2279 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AD1836 + tristate + config SND_SOC_AD1938 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 78dce5d..9cbcf8f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ad1836-objs := ad1836.o snd-soc-ad1938-objs := ad1938.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o @@ -45,6 +46,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-max9877-objs := max9877.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c new file mode 100644 index 0000000..79b9d19 --- /dev/null +++ b/sound/soc/codecs/ad1836.c @@ -0,0 +1,451 @@ +/* + * File: sound/soc/codecs/ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 04 2009 + * Description: Driver for AD1836 sound chip + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-dapm.h> +#include <linux/spi/spi.h> +#include "ad1836.h" + +/* codec private data */ +struct ad1836_priv { + struct snd_soc_codec codec; + u16 reg_cache[AD1836_NUM_REGS]; +}; + +static struct snd_soc_codec *ad1836_codec; +struct snd_soc_codec_device soc_codec_dev_ad1836; +static int ad1836_register(struct ad1836_priv *ad1836); +static void ad1836_unregister(struct ad1836_priv *ad1836); + +/* + * AD1836 volume/mute/de-emphasis etc. controls + */ +static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; + +static const struct soc_enum ad1836_deemp_enum = + SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); + +static const struct snd_kcontrol_new ad1836_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL, + AD1836_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL, + AD1836_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL, + AD1836_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE, + AD1836_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE, + AD1836_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE, + AD1836_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE, + AD1836_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE, + AD1836_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1, + AD1836_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1, + AD1836_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1, + AD1836_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + /* at present, we support adc aux mode to interface with + * blackfin sport tdm mode + */ + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + /* ALCLK,ABCLK are both output, AD1836 can only be master */ + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ad1836_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, + AD1836_DAC_WORD_LEN_MASK, word_len); + + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + AD1836_ADC_WORD_LEN_MASK, word_len); + + return 0; +} + + +/* + * interface to read/write ad1836 register + */ +#define AD1836_SPI_REG_SHFT 12 +#define AD1836_SPI_READ (1 << 11) +#define AD1836_SPI_VAL_MSK 0x3FF + +/* + * write to the ad1836 register space + */ + +static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *reg_cache = codec->reg_cache; + int ret = 0; + + if (value != reg_cache[reg]) { + unsigned short buf; + struct spi_transfer t = { + .tx_buf = &buf, + .len = 2, + }; + struct spi_message m; + + buf = (reg << AD1836_SPI_REG_SHFT) | + (value & AD1836_SPI_VAL_MSK); + spi_message_init(&m); + spi_message_add_tail(&t, &m); + ret = spi_sync(codec->control_data, &m); + if (ret == 0) + reg_cache[reg] = value; + } + + return ret; +} + +/* + * read from the ad1836 register space cache + */ +static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +static int __devinit ad1836_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ad1836_priv *ad1836; + + ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL); + if (ad1836 == NULL) + return -ENOMEM; + + codec = &ad1836->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, ad1836); + + return ad1836_register(ad1836); +} + +static int __devexit ad1836_spi_remove(struct spi_device *spi) +{ + struct ad1836_priv *ad1836 = dev_get_drvdata(&spi->dev); + + ad1836_unregister(ad1836); + return 0; +} + +static struct spi_driver ad1836_spi_driver = { + .driver = { + .name = "ad1836-spi", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = ad1836_spi_probe, + .remove = __devexit_p(ad1836_spi_remove), +}; + +static struct snd_soc_dai_ops ad1836_dai_ops = { + .hw_params = ad1836_hw_params, + .set_fmt = ad1836_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad1836_dai = { + .name = "AD1836", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad1836_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad1836_dai); + +static int ad1836_register(struct ad1836_priv *ad1836) +{ + int ret; + struct snd_soc_codec *codec = &ad1836->codec; + + if (ad1836_codec) { + dev_err(codec->dev, "Another ad1836 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->private_data = ad1836; + codec->reg_cache = ad1836->reg_cache; + codec->reg_cache_size = AD1836_NUM_REGS; + codec->name = "AD1836"; + codec->owner = THIS_MODULE; + codec->dai = &ad1836_dai; + codec->num_dai = 1; + codec->write = ad1836_write_reg; + codec->read = ad1836_read_reg_cache; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ad1836_dai.dev = codec->dev; + ad1836_codec = codec; + + /* default setting for ad1836 */ + /* de-emphasis: 48kHz, power-on dac */ + codec->write(codec, AD1836_DAC_CTRL1, 0x300); + /* unmute dac channels */ + codec->write(codec, AD1836_DAC_CTRL2, 0x0); + /* high-pass filter enable, power-on adc */ + codec->write(codec, AD1836_ADC_CTRL1, 0x100); + /* unmute adc channles, adc aux mode */ + codec->write(codec, AD1836_ADC_CTRL2, 0x180); + /* left/right diff:PGA/MUX */ + codec->write(codec, AD1836_ADC_CTRL3, 0x3A); + /* volume */ + codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad1836); + return ret; + } + + ret = snd_soc_register_dai(&ad1836_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + kfree(ad1836); + return ret; + } + + return 0; +} + +static void ad1836_unregister(struct ad1836_priv *ad1836) +{ + snd_soc_unregister_dai(&ad1836_dai); + snd_soc_unregister_codec(&ad1836->codec); + kfree(ad1836); + ad1836_codec = NULL; +} + +static int ad1836_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ad1836_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ad1836_codec; + codec = ad1836_codec; + + /* 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: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ad1836_snd_controls, + ARRAY_SIZE(ad1836_snd_controls)); + snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, + ARRAY_SIZE(ad1836_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int ad1836_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad1836 = { + .probe = ad1836_probe, + .remove = ad1836_remove, + /* The power management of ad1836 is very simple. There are + * only adc&dac 2 components to control. Dapm handles them. + */ + .suspend = NULL, + .resume = NULL, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); + +static int __init ad1836_init(void) +{ + int ret; + + ret = spi_register_driver(&ad1836_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", + ret); + } + + return ret; +} +module_init(ad1836_init); + +static void __exit ad1836_exit(void) +{ + spi_unregister_driver(&ad1836_spi_driver); +} +module_exit(ad1836_exit); + +MODULE_DESCRIPTION("ASoC ad1836 driver"); +MODULE_AUTHOR("Barry Song <21cnbao(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h new file mode 100644 index 0000000..7660ee6 --- /dev/null +++ b/sound/soc/codecs/ad1836.h @@ -0,0 +1,64 @@ +/* + * File: sound/soc/codecs/ad1836.h + * Based on: + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 04, 2009 + * Description: definitions for AD1836 registers + * + * Modified: + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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. + */ + +#ifndef __AD1836_H__ +#define __AD1836_H__ + +#define AD1836_DAC_CTRL1 0 +#define AD1836_DAC_POWERDOWN 2 +#define AD1836_DAC_SERFMT_MASK 0xE0 +#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) +#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) +#define AD1836_DAC_WORD_LEN_MASK 0x18 + +#define AD1836_DAC_CTRL2 1 +#define AD1836_DACL1_MUTE 0 +#define AD1836_DACR1_MUTE 1 +#define AD1836_DACL2_MUTE 2 +#define AD1836_DACR2_MUTE 3 +#define AD1836_DACL3_MUTE 4 +#define AD1836_DACR3_MUTE 5 + +#define AD1836_DAC_L1_VOL 2 +#define AD1836_DAC_R1_VOL 3 +#define AD1836_DAC_L2_VOL 4 +#define AD1836_DAC_R2_VOL 5 +#define AD1836_DAC_L3_VOL 6 +#define AD1836_DAC_R3_VOL 7 + +#define AD1836_ADC_CTRL1 12 +#define AD1836_ADC_POWERDOWN 7 +#define AD1836_ADC_HIGHPASS_FILTER 8 + +#define AD1836_ADC_CTRL2 13 +#define AD1836_ADCL1_MUTE 0 +#define AD1836_ADCR1_MUTE 1 +#define AD1836_ADCL2_MUTE 2 +#define AD1836_ADCR2_MUTE 3 +#define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_SERFMT_MASK (7 << 6) +#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) +#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) + +#define AD1836_ADC_CTRL3 14 + +#define AD1836_NUM_REGS 16 + +extern struct snd_soc_dai ad1836_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad1836; +#endif -- 1.5.6.3

2 1

[alsa-devel] [PATCH 0/2] ASoC: TWL4030: Cosmetic and PM related changes
by Peter Ujfalusi 13 Aug '09

13 Aug '09

Hello, I'm not sure, if the first patch is acceptable by ASoC standards, but: Before the patch the analog capture path strings ended up like this at the end: "Analog Left Capture Route Main mic Switch" These switches showed up in the playback tab of alsamixer, which was not nice. After the patch the strings will be like this: "Analog Left Main mic Capture Route" Which show up correctly in alsamixer as "Analog Left Main mic" under the capture tab. The second patch cleans up the output amplifier mute/un-mute by adding PGA_E to all output pins, which were handled by the twl4030_codec_mute function. Since those gain controls attached to the outputs are not in DAPM domain (yet?) and they have to be muted/turned off when they are not needed. Previously all of these (Earpiece PreDrivL/R, CarkitL/R) muted/un-muted at the same time without checking if the output actually is needed, there could be cases when the output amplifier is enabled but no actual audio data is routed there, which could leak some noise on the output. I know about the changes in core that makes this hassle with the mute unnecesary (the core sets the BIAS_ON also when any bypass path is active), I'll be looking at that next. --- Peter Ujfalusi (2): ASoC: TWL4030: Fix for capture mixer strings ASoC: TWL4030: Introduce PGAs for outputs sound/soc/codecs/twl4030.c | 155 ++++++++++++++++++++++++-------------------- 1 files changed, 85 insertions(+), 70 deletions(-)

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Re: [alsa-devel] Audio support for TLV320AIC23 on Simtec S3C24XX products
by Mark Brown 13 Aug '09

13 Aug '09

On Thu, Aug 13, 2009 at 02:24:30PM +0100, Ben Dooks wrote: > Mark Brown wrote: > >Also, as I've mentioned to you on a number of previous occasions please > >do try to remember to CC subsystem maintainers when submitting patches. > I thought you where the ASoC maintainer? Who else should I be CC'ing? You'd only sent the patches to the list, there wasn't a CC to me. > >So I'm guessing that this is muting through powering off? In that case > >(and probably in general) this would be better implemented as a DAPM > >widget with a SND_SOC_DAPM_PIN_SWITCH() control set up to provide the > >mute functionality. Doing this will allow the speaker to be powered off > >when no audio is being played, saving power and helping avoid any pops > >or clicks being amplified by it. > I tried it that way first, unfortunately the AMP itself makes a POP > when powering up and down, the headphone output of the TLV320AIC23 > doesn't seem to have a problem with pops. But it doesn't pop when people use the mute control? OK, anyway.

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Re: [alsa-devel] tlv320aic3x: Allow driver to use already registered i2c device
by Mark Brown 13 Aug '09

13 Aug '09

On Thu, Aug 13, 2009 at 02:26:38PM +0100, Ben Dooks wrote: > Mark Brown wrote: > >Why would you want to do this? I suspect that perhaps you want to > >convert the driver to use the standard device model instead but without > >understanding your motivation for making this change it is very hard to > >comment in detail. > The device is registered by the code that sets the initial devices > up that have been passed from the bootloader, thus the tlv320aic33 > fails to initialise as there is already an i2c device on the i2c bus > when it comes along. OK, that's what I thought. You need to convert the driver to use the normal device model probe methods rather than using bodges like this. See drivers like wm8731 for an example of how the conversion looks.

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Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

At Thu, 13 Aug 2009 12:41:08 +0300, Mantas Pilipavičius wrote: > > I waited a bit longer and it worked with this command: > speaker-test -c2 > > But again, when I try to play my mp3 files with mp3blaster it doesn't affect > the sound when I use alsamixer. But with speaker-test it does work. It depends how app accesses the sound device. If the application uses "default" PCM, not "hw" or so, then it should work without changing. Or, if app uses OSS emulation over /dev/dsp, softvol doesn't work. Takashi

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[alsa-devel] [PATCH 0/3] patch sequence adds analog devices ad1836 asoc support
by Barry Song 13 Aug '09

13 Aug '09

Hi Mark, The patch sequence adds ad1836 support based on asoc patch 1: new ad1836 codec driver based on asoc patch 2: add set_tdm_slot entry in tdm-dai to define the relationship between audio channel and tdm slot number patch 3: new board driver to connect ad1836 codec and bf5xx patch 3 depends on patch 2. Thanks Barry

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Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

[Please don't drop Cc to alsa-devel ML] At Thu, 13 Aug 2009 11:27:29 +0300, Mantas Pilipavičius wrote: > > So I can play a sound with this command: > speaker-test -Dhw:0,0 -c2 -twav > > I edit my asound.rc: > nano -w ~/.asoundrc and nano -w /etc/asound.conf > > With this: > > pcm.softvol { > type softvol > slave { > pcm "hw:0,0" > } > control { > name "Garsas" > card 0 > } > } > > And the I try: > > speaker-test -Dsoftwol -c2 -twav > > And it doesn't work. Elaborate what doesn't work how. > Sorry to disturb you and consume your precious time, but > I am new to open source and really need this. What about if you run without -Dxxx option? Usually PCM volume should be created as a softvol if not exists. Takashi > On Thu, Aug 13, 2009 at 11:14 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Aug 2009 10:55:38 +0300, > Mantas Pilipavičius wrote: > > > > Can you explain more detailed? What is data manipulation? > > You need to change all the stream data multiplying the volume > gain/attenuation in software because you hardware gives no volume > control by itself. > > In the case of alsa-lib, it's softvol plugin. Or, if you use PA, > PA does it automatically. > > Takashi > > > On Thu, Aug 13, 2009 at 8:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Thu, 13 Aug 2009 00:39:48 +0300, > > Mantas Pilipavičius wrote: > > > > > > http://www.alsa-project.org/db/?f= > > afa01dd48ed59d895d18407af62dbd1fa138686c > > > Everything works great, I can play the sound, but PCM and Master > > channels > > > don't work, so I can't increase or decrease volume. Only thing I > can do > > is > > > mute/unmute PCM channel > > > > This hardware has no volume control indeed. > > The volume has to be adjusted in software by data manipulation. > > > > Takashi > > > > > >

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[alsa-devel] ALSA and MP3
by gammay 13 Aug '09

13 Aug '09

Hello, I am looking at making this product use case possible: Audio samples are streamed over wire/wireless from a source device to a sink device. The streaming format is in mp3. On the source device I need to capture locally played mp3 samples and stream to the sink device. The sink device takes care of uncompressing and playback. Can we capture compressed mp3 streams directly using ALSA? It is possible to capture uncompressed PCM samples using file plug or by other means in ALSA. While it is possible to directly read from the mp3 file, I would rather grap the samples that a media player sends out because a) it takes care of timing b) I can control the media player to play/pause the stream and for playlist management. For the user it is transperent - he/she will use only the local media player to control the streaming. I am not sure if this lies in the domain of ALSA at all - as I understand once we are in ALSA we are dealing with only PCM. In such case, please give me some pointers where I can look at to make this possible. I am using mplayer, but can also look at other types (non-GUI, command-line based). Can we do this in a non-media player specific way? Thank you, Girish

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Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

At Thu, 13 Aug 2009 10:55:38 +0300, Mantas Pilipavičius wrote: > > Can you explain more detailed? What is data manipulation? You need to change all the stream data multiplying the volume gain/attenuation in software because you hardware gives no volume control by itself. In the case of alsa-lib, it's softvol plugin. Or, if you use PA, PA does it automatically. Takashi > On Thu, Aug 13, 2009 at 8:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Aug 2009 00:39:48 +0300, > Mantas Pilipavičius wrote: > > > > http://www.alsa-project.org/db/?f= > afa01dd48ed59d895d18407af62dbd1fa138686c > > Everything works great, I can play the sound, but PCM and Master > channels > > don't work, so I can't increase or decrease volume. Only thing I can do > is > > mute/unmute PCM channel > > This hardware has no volume control indeed. > The volume has to be adjusted in software by data manipulation. > > Takashi > >

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[alsa-devel] [PATCH] new board driver to connect bfin-5xx with ad1836 codec
by Barry Song 13 Aug '09

13 Aug '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 139 +++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+), 0 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8a4de4d..ac927ff 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -88,6 +88,14 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f4d7607..87e3042 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -16,11 +16,13 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 0000000..c0ef965 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,139 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_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; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI slots, 8 channels, mask is defined as slot seq + * 0x73625140 means audio channel 0 uses slot 0 + * channel 1 uses slot 4 + * channel 2 uses slot 1 + * channel 3 uses slot 5 + * ... + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x73625140, 8, 32); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

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[alsa-devel] ASoC update for 2.6.32
by Mark Brown 13 Aug '09

13 Aug '09

The following changes since commit e0026beac0f1a6253488478b3f022ff243b012b9: Mark Brown (1): ASoC: Update WM9081 for tdm_slot() API change 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 Mark Brown (1): ASoC: Update AD1938 for new TDM slot API sound/soc/codecs/ad1938.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

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[alsa-devel] Intel card master and PCM channels doesn't work
by Mantas Pilipavičius 13 Aug '09

13 Aug '09

http://www.alsa-project.org/db/?f=afa01dd48ed59d895d18407af62dbd1fa138686c Everything works great, I can play the sound, but PCM and Master channels don't work, so I can't increase or decrease volume. Only thing I can do is mute/unmute PCM channel

2 1

[alsa-devel] [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM
by Chaithrika U S 12 Aug '09

12 Aug '09

Define resources for McASP1 used on DA830/OMAP-L137 EVM, add platform device defintion, initialization function. Additionally, this patch also adds version and FIFO related members to platform data structure. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to DaVinci GIT tree. In this version of the patch, the init fucntion, resource and platform device definitions have neen moved to the devices-da8xx.c file. arch/arm/mach-davinci/board-da830-evm.c | 27 +++++++++++++++++++++ arch/arm/mach-davinci/da830.c | 7 +++-- arch/arm/mach-davinci/devices-da8xx.c | 36 ++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++++ arch/arm/mach-davinci/include/mach/da8xx.h | 3 ++ 5 files changed, 85 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 9025669..a45340d 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -23,6 +23,7 @@ #include <mach/irqs.h> #include <mach/cp_intc.h> #include <mach/da8xx.h> +#include <mach/asp.h> #define DA830_EVM_PHY_MASK 0x0 #define DA830_EVM_MDIO_FREQUENCY 2200000 /* PHY bus frequency */ @@ -51,6 +52,25 @@ static struct davinci_uart_config da830_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +static u8 da830_iis_serializer_direction[] = { + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, TX_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da830_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da830_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + static __init void da830_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -93,6 +113,13 @@ static __init void da830_evm_init(void) davinci_serial_init(&da830_evm_uart_config); i2c_register_board_info(1, da830_evm_i2c_devices, ARRAY_SIZE(da830_evm_i2c_devices)); + + ret = da8xx_pinmux_setup(da830_mcasp1_pins); + if (ret) + pr_warning("da830_evm_init: mcasp1 mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(1, &da830_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da830.c b/arch/arm/mach-davinci/da830.c index 3a9b634..19b2748 100644 --- a/arch/arm/mach-davinci/da830.c +++ b/arch/arm/mach-davinci/da830.c @@ -23,6 +23,7 @@ #include <mach/common.h> #include <mach/time.h> #include <mach/da8xx.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -411,9 +412,9 @@ static struct davinci_clk da830_clks[] = { CLK("eqep.0", NULL, &eqep0_clk), CLK("eqep.1", NULL, &eqep1_clk), CLK("da830_lcdc", NULL, &lcdc_clk), - CLK("soc-audio.0", NULL, &mcasp0_clk), - CLK("soc-audio.1", NULL, &mcasp1_clk), - CLK("soc-audio.2", NULL, &mcasp2_clk), + CLK("davinci-mcasp.0", NULL, &mcasp0_clk), + CLK("davinci-mcasp.1", NULL, &mcasp1_clk), + CLK("davinci-mcasp.2", NULL, &mcasp2_clk), CLK("musb_hdrc", NULL, &usb20_clk), CLK(NULL, "aemif", &aemif_clk), CLK(NULL, "aintc", &aintc_clk), diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index 11c0971..fe0baaf 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -281,7 +281,43 @@ static struct platform_device da8xx_emac_device = { .resource = da8xx_emac_resources, }; +static struct resource da830_mcasp1_resources[] = { + { + .name = "mcasp1", + .start = DAVINCI_DA830_MCASP1_REG_BASE, + .end = DAVINCI_DA830_MCASP1_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .end = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AREVT, + .end = DAVINCI_DA830_DMA_MCASP1_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da830_mcasp1_device = { + .name = "davinci-mcasp", + .id = 1, + .num_resources = ARRAY_SIZE(da830_mcasp1_resources), + .resource = da830_mcasp1_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); } + +void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) +{ + if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + da830_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&da830_mcasp1_device); + } +} diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index cdf1f44..f3c97ac 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da830 McASP1 register banks */ +#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 + /* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 @@ -26,6 +29,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da830 McASP1 */ +#define DAVINCI_DA830_DMA_MCASP1_AREVT 2 +#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT @@ -43,6 +50,14 @@ struct snd_platform_data { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + u8 txnumevt; + u8 rxnumevt; +}; + +enum { + MCASP_VERSION_1 = 0, /* DM646x */ + MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; #define INACTIVE_MODE 0 diff --git a/arch/arm/mach-davinci/include/mach/da8xx.h b/arch/arm/mach-davinci/include/mach/da8xx.h index a8cb570..30c5c40 100644 --- a/arch/arm/mach-davinci/include/mach/da8xx.h +++ b/arch/arm/mach-davinci/include/mach/da8xx.h @@ -15,6 +15,7 @@ #include <mach/edma.h> #include <mach/i2c.h> #include <mach/emac.h> +#include <mach/asp.h> /* * The cp_intc interrupt controller for the da8xx isn't in the same @@ -65,6 +66,7 @@ int da8xx_register_edma(void); int da8xx_register_i2c(int instance, struct davinci_i2c_platform_data *pdata); int da8xx_register_watchdog(void); int da8xx_register_emac(void); +void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata); extern struct platform_device da8xx_serial_device; extern struct emac_platform_data da8xx_emac_pdata; @@ -100,6 +102,7 @@ extern const short da850_uart2_pins[]; extern const short da850_i2c0_pins[]; extern const short da850_i2c1_pins[]; extern const short da850_cpgmac_pins[]; +extern const short da850_mcasp_pins[]; int da8xx_pinmux_setup(const short pins[]); -- 1.5.6

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[alsa-devel] [PATCH] alsa: hdsp - allow proc reporting with disconnected io box
by Tim Blechmann 12 Aug '09

12 Aug '09

the hdsp driver refuses to report any information via the proc interface, if the io box is not connected. with this patch, the content of the control and status registers is printed before the iobox check. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 37 ++++++++++++++++++++----------------- 1 files changed, 20 insertions(+), 17 deletions(-)

2 3

[alsa-devel] [PATCH] alsa-info: Use sysfs if available instead of dmidecode
by Ozan Çağlayan 12 Aug '09

12 Aug '09

Using sysfs for acquiring DMI data requires no root privileges. Use it if available instead of dmidecode. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index 0ab1ddf..8f401c0 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -417,7 +417,12 @@ ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) DMIDECODE=$(which dmidecode 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) -if [ -x $DMIDECODE ]; then +#Check for DMI data +if [ -d /sys/class/dmi/id ]; then + # No root privileges are required when using sysfs method + DMI_SYSTEM_MANUFACTURER=$(cat /sys/class/dmi/id/sys_vendor 2>/dev/null) + DMI_SYSTEM_PRODUCT_NAME=$(cat /sys/class/dmi/id/product_name 2>/dev/null) +elif [ -x $DMIDECODE ]; then DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) fi -- 1.6.3.4

2 1

[alsa-devel] [PATCH] add output/input widgets in ad1938 to make dac/adc dynamic PM work
by Barry Song 12 Aug '09

12 Aug '09

According to the function dapm_dac_check_power() in sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any output widget as sink. And according to dapm_adc_check_power(), adc power can't be on/off stand-alone without any input widget as source. So we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC to hope their power can be managed dynamically. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad1938.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index aafda74..9b8cc5c 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -99,11 +99,23 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), }; static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC4OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, }; /* -- 1.5.6.3

2 1

[alsa-devel] [PATCHv2 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 12 Aug '09

12 Aug '09

Hello guys, Here is version 2 of this series. There were lots of little changes and also one significant change. Here is a changelog: - I tried to avoid additional ifdefs in mcbsp code - The WAKEUPEN_ALL macro was removed - I added a REVISIT prefix for the additional delay on mcbsp start - New added constants are now following mcbsp.h style - While reading DMA operation mode of mcbsp, now it prints also some help text to explain the meaning of each value (even though you still write numbers). Besides, I re-wrote the code for ASoC part, now the threshold and dma op mode usage are un-glued from mcbsp code. I tried to do not leave omap-pcm.c dirty, but, there was still one needed callback for setting the mcbsp threshold, as that needs to be done before DMA is started. In this series I've left THRESHOLD mode as default for mcbsp 2 only. But as I saw in previous comments, this is still an open issue :-) Comments, as usual are welcome. Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 375 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 67 +++++- sound/soc/omap/omap-pcm.c | 33 +++- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 517 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 23

[alsa-devel] [PATCH] ALSA: hda - fix noise issue when recording from digital mic with alc268
by Herton Ronaldo Krzesinski 12 Aug '09

12 Aug '09

With auto config model of alc268 realtek codec, it allows to select any of possible available digital microphone inputs when only one is available. For example, when only digital mic in nid 0x12 is available, on second input source it will allow you to select unavailable digital mic in nid 0x13. The problem is that selecting unavailable digital mic creates a source of noise when recording (I'm not sure if this happens on all machines with alc268 and only one digital mic input, but testing on a quanta uw1 netbook a lot of noise is introduced in recording from digital mic 0x12/first input source, when you select the unavailable digital mic 0x13 for capture source 0x24 in the second input source in mixer). Then to avoid noise when recording from digital mic with auto model in this case, prevent a digital mic input source to be selected if microphone is not available. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++----------------- 1 files changed, 36 insertions(+), 26 deletions(-) I'm attaching the codec info too here of the machine with the problem. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac3207e..7ff2934 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12232,32 +12232,40 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { struct hda_input_mux *imux = &spec->private_imux[0]; - int i, idx1; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - switch(cfg->input_pins[i]) { - case 0x18: - idx1 = 0; /* Mic 1 */ - break; - case 0x19: - idx1 = 1; /* Mic 2 */ - break; - case 0x1a: - idx1 = 2; /* Line In */ - break; - case 0x1c: - idx1 = 3; /* CD */ - break; - case 0x12: - case 0x13: - idx1 = 6; /* digital mics */ - break; - default: - continue; + int i, idx1, dmic_nid; + + dmic_nid = 0x12; + while (dmic_nid <= 0x13) { + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch (cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + case 0x12: + case 0x13: + if (cfg->input_pins[i] != dmic_nid) + continue; + idx1 = 6; /* digital mics */ + break; + default: + continue; + } + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; } - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx1; - imux->num_items++; + imux++; + dmic_nid++; } return 0; } @@ -12355,7 +12363,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); - spec->num_mux_defs = 1; + spec->num_mux_defs = 2; spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); @@ -12665,6 +12673,8 @@ static int patch_alc268(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], 0, AC_VERB_SET_CONNECT_SEL, + i < spec->num_mux_defs ? + spec->input_mux[i].items[0].index : spec->input_mux->items[0].index); } -- 1.6.4

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
by Chaithrika U S 11 Aug '09

11 Aug '09

There is one instance of McASP on DA850/OMAP-L138 SoC. This is connected to TLV320AIC3106 codec for audio playback and capture. This patch adds audio support on this platform. Some of the structure prefix names which are common for DA830/OMAP-L137 EVM and DA850/OMAP-L138 EVM have been renamed to da8xx from da830. 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/Kconfig | 13 +++++++++++-- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++------ 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 677a538..4dfd4ad 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -43,12 +43,21 @@ config SND_DAVINCI_SOC_SFFSDR Lyrtech SFFSDR board. config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAPL137 EVM" + tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM select SND_DAVINCI_SOC_MCASP select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DA830/OMAPL137 EVM + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 5e2195f..a6939d7 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -14,4 +14,5 @@ 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-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA850_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 a5a9212..46c1b0c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,7 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; - else if (machine_is_davinci_da830_evm()) + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) sysclk = 24576000; else @@ -165,7 +166,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; -static struct snd_soc_dai_link da830_evm_dai = { +static struct snd_soc_dai_link da8xx_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], @@ -191,8 +192,15 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { }; static struct snd_soc_card da830_snd_soc_card = { - .name = "DA830 EVM", - .dai_link = &da830_evm_dai, + .name = "DA830/OMAP-L137 EVM", + .dai_link = &da8xx_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + +static struct snd_soc_card da850_snd_soc_card = { + .name = "DA850/OMAP-L138 EVM", + .dai_link = &da8xx_evm_dai, .platform = &davinci_soc_platform, .num_links = 1, }; @@ -209,7 +217,7 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; -static struct aic3x_setup_data da830_evm_aic3x_setup = { +static struct aic3x_setup_data da8xx_evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x18, }; @@ -232,7 +240,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { static struct snd_soc_device da830_evm_snd_devdata = { .card = &da830_snd_soc_card, .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &da830_evm_aic3x_setup, + .codec_data = &da8xx_evm_aic3x_setup, +}; + +static struct snd_soc_device da850_evm_snd_devdata = { + .card = &da850_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &da8xx_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; @@ -255,6 +269,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_da830_evm()) { evm_snd_dev_data = &da830_evm_snd_devdata; index = 1; + } else if (machine_is_davinci_da850_evm()) { + evm_snd_dev_data = &da850_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.5.6

1 0

[alsa-devel] [PATCH v2 2/2] ASoC: DaVinci: Add a DAI format to McASP driver
by Chaithrika U S 11 Aug '09

11 Aug '09

The patch adds a DAI format: Codec bit clock master and frame sync slave, to the driver. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch has been separated out from the first patch of this series. In the previous version, this was part of the McASP driver FIFO support patch. 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 | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e672f43..eca22d7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -443,6 +443,16 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- 1.5.6

1 0

[alsa-devel] hda-emu codec information files
by Ozan Çağlayan 11 Aug '09

11 Aug '09

Hi, I'm able to collect a lot of alsa-info from a bunch of netbook/laptop and desktop boards. I can send them as a patch series if it will help debug problems Thanks Ozan

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 11 Aug '09

11 Aug '09

The following changes since commit e0c48a18f76415d3ec1ac28e5f7a0df10e2f3d8c: Mark Brown (1): ASoC: Drop unneeded declaration of removed wm8731 SPI write function 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 Daniel Ribeiro (1): ASoC: change set_tdm_slot api to allow slot_width override. Mark Brown (2): Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 ASoC: Update WM9081 for tdm_slot() API change Randy Dunlap (1): ASoC: fix I2C build errors include/sound/soc-dai.h | 5 ++- sound/soc/codecs/wm9081.c | 63 ++++++++++++++++++++++++++------------------ sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++----- sound/soc/soc-cache.c | 4 +- sound/soc/soc-core.c | 9 ++++-- 6 files changed, 69 insertions(+), 41 deletions(-)

2 1

[alsa-devel] hard freezes with hdsp with cardbus/pci adapter card
by Tim Blechmann 11 Aug '09

11 Aug '09

hi all, i have some issues with playback on one of my machines with different kernels, having my rme hammerfall hdsp connected via a cardbus/pci adapter card. after some time (between a few seconds and several minutes) of flawless playback, the system completely freezes, magic sysrq keys are not functional either ... the freeze occurs with rt and non-rt kernels, with jack running with and without realtime scheduling. i have the suspicion, that it is related to the combination of the hdsp with the cardbus controller of the adapter card ... the interesting part of lspci is: 06:01.0 CardBus bridge: ENE Technology Inc CB1410 Cardbus Controller (rev 01) Subsystem: ENE Technology Inc CB1410 Cardbus Controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at fb600000 (32-bit, non-prefetchable) [size=4K] Bus: primary=06, secondary=07, subordinate=0a, sec-latency=176 Memory window 0: c4000000-c7fff000 (prefetchable) Memory window 1: c8000000-cbfff000 I/O window 0: 0000e000-0000e0ff I/O window 1: 0000e400-0000e4ff BridgeCtl: Parity+ SERR+ ISA- VGA- MAbort- >Reset- 16bInt- PostWrite+ 16-bit legacy interface ports at 0001 Kernel driver in use: yenta_cardbus Kernel modules: yenta_socket 07:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 0f) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255 Interrupt: pin A routed to IRQ 17 Region 0: Memory at c8000000 (32-bit, non-prefetchable) [size=64K] Kernel driver in use: RME Hammerfall DSP Kernel modules: snd-hdsp the irq 17 is used by these two devices only. i am not sure, how to proceed debugging this, since i cannot get any useful debugging information ... thnx, tim -- tim(a)klingt.org http://tim.klingt.org A paranoid is a man who knows a little of what's going on. William S. Burroughs

3 7

[alsa-devel] (no subject)
by Chaithrika U S 11 Aug '09

11 Aug '09

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

2 1

[alsa-devel] [PATCH 1/5] ASoC: DaVinci: i2s: don't bounce through rtd to get dai
by Troy Kisky 11 Aug '09

11 Aug '09

dai is a parameter to the functions, so use it instead of looking it up. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 14 +++++--------- 1 files changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e5cd97b..2a56fb7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,9 +353,8 @@ static int davinci_i2s_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 davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; @@ -425,8 +424,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, static int davinci_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); davinci_mcbsp_stop(dev, playback); if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) { @@ -439,8 +437,7 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int ret = 0; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) @@ -466,8 +463,7 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); davinci_mcbsp_stop(dev, playback); } -- 1.5.6.3

2 11

[alsa-devel] Automatic mic-selection on Realtek and IDT/STAC codecs
by Takashi Iwai 11 Aug '09

11 Aug '09

Hi, recently I added the code for HD-audio Realtek and IDT/Sigmatel codecs to support the automatic capture-source selection via mic jack plugging. This feature is enabled for machines with one external and one internal mic, and only for auto-BIOS parser mode in the case of Realtek codecs. So, don't be surprised if you see neither "Capture Source" nor "Input Source" mixer elements after updates. Just plugging the mic jack will switch the input source from the internal to the external mic. If you encounter any problems regarding this change, please let me know. thanks, Takashi

1 0

[alsa-devel] NULL pointer dereference in ALSA triggered through saa7134-alsa
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, I've finally succesfully compiled and linked saa7134-alsa driver using an external alsa-driver and its Module.symvers file. Everything seems okay, no undefined symbol or something else: - An installed 2.6.30.4 kernel which only builds and brings soundcore and sound_firmware, - Latest alsa-driver built externally and installed, - Latest saa7134-alsa, cx88-alsa, etc. code from linus-2.6 (seen that they don't affected by some API/ABI changes) patched on top of the alsa-driver tarball, I'm copying Module.symvers file generated after building the alsa-driver into each directory for getting rid of undefined snd_*() symbols. When I plug a saa7134 PCI card and reboot, the -alsa driver is automatically probed but it immediately oopses. I wonder if this could be related to my weird-but-apparently-successful compilation experience above? [ 6.876120] Linux video capture interface: v2.00 [ 7.089550] saa7130/34: v4l2 driver version 0.2.15 loaded [ 7.089612] saa7134 0000:03:05.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20 [ 7.089622] saa7133[0]: found at 0000:03:05.0, rev: 209, irq: 20, latency: 64, mmio: 0xfdbff000 [ 7.089631] saa7133[0]: subsystem: 1461:f11d, board: Avermedia PCI pure analog (M135A) [card=149,autodetected] [ 7.089659] saa7133[0]: board init: gpio is 40000 [ 7.089744] input: saa7134 IR (Avermedia PCI pure as /devices/pci0000:00/0000:00:14.4/0000:03:05.0/input/input6 [ 7.089789] IRQ 20/saa7133[0]: IRQF_DISABLED is not guaranteed on shared IRQs [ 7.230265] saa7133[0]: i2c eeprom 00: 61 14 1d f1 54 20 1c 00 43 43 a9 1c 55 d2 b2 92 [ 7.230274] saa7133[0]: i2c eeprom 10: ff ff ff ff ff 20 ff ff ff ff ff ff ff ff ff ff [ 7.230282] saa7133[0]: i2c eeprom 20: 01 40 01 32 32 01 01 43 88 ff 00 56 ff ff ff ff [ 7.230289] saa7133[0]: i2c eeprom 30: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230296] saa7133[0]: i2c eeprom 40: ff 22 00 c0 96 ff 03 30 15 00 ff ff ff ff ff ff [ 7.230302] saa7133[0]: i2c eeprom 50: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230309] saa7133[0]: i2c eeprom 60: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230316] saa7133[0]: i2c eeprom 70: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230323] saa7133[0]: i2c eeprom 80: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230330] saa7133[0]: i2c eeprom 90: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230337] saa7133[0]: i2c eeprom a0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230344] saa7133[0]: i2c eeprom b0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230350] saa7133[0]: i2c eeprom c0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230357] saa7133[0]: i2c eeprom d0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230364] saa7133[0]: i2c eeprom e0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230371] saa7133[0]: i2c eeprom f0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.412095] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 7.512683] hda_codec: ALC883: BIOS auto-probing. [ 7.512886] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:14.2/input/input7 [ 7.516862] HDA Intel 0000:01:05.2: PCI INT B -> GSI 19 (level, low) -> IRQ 19 [ 7.689107] tuner 1-004b: chip found @ 0x96 (saa7133[0]) [ 7.724034] tda829x 1-004b: setting tuner address to 60 [ 7.800011] tda829x 1-004b: type set to tda8290+75a [ 8.294455] usblp0: USB Bidirectional printer dev 2 if 0 alt 0 proto 2 vid 0x04E8 pid 0x325B [ 8.294500] usbcore: registered new interface driver usblp [ 10.814379] saa7133[0]: dsp access error [ 10.845083] saa7133[0]: registered device video0 [v4l2] [ 10.845111] saa7133[0]: registered device vbi0 [ 10.845130] saa7133[0]: registered device radio0 [ 10.950359] EXT3 FS on sda5, internal journal [ 10.996203] saa7134 ALSA driver for DMA sound loaded [ 10.996217] IRQ 20/saa7133[0]: IRQF_DISABLED is not guaranteed on shared IRQs [ 10.996247] saa7133[0]/alsa: saa7133[0] at 0xfdbff000 irq 20 registered as card -1 <-- /* That's interesting? */ [ 10.996349] BUG: unable to handle kernel NULL pointer dereference at 00000002 <-- ********** 1st null ptr deref [ 10.996962] IP: [<f8834cd4>] snd_pcm_timer_init+0x26/0xd1 [snd_pcm] [ 10.997255] *pde = 00000000· [ 10.997255] Oops: 0000 [#1] SMP· [ 10.997255] last sysfs file: /sys/module/saa7134/initstate [ 10.997255] Modules linked in: saa7134_alsa(+) usblp tda827x tda8290 snd_hda_codec_atihdmi tuner snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss saa7134 snd_seq_m idi_event snd_seq snd_seq_device ir_common snd_pcm_oss snd_mixer_oss v4l2_common videodev v4l1_compat videobuf_dma_sg videobuf_core snd_pcm tveeprom snd_timer snd i2c_piix4 shpchp k8temp soundcore snd_pa ge_alloc pcspkr r8169 ati_agp agpgart mii brd ata_generic pata_acpi ahci pata_atiixp libata [ 10.997255]· [ 10.997255] Pid: 1141, comm: modprobe.bin Not tainted (2.6.30.4-126 #3) Unknow [ 10.997255] EIP: 0060:[<f8834cd4>] EFLAGS: 00010246 CPU: 1 [ 10.997255] EIP is at snd_pcm_timer_init+0x26/0xd1 [snd_pcm] [ 10.997255] EAX: 00000002 EBX: f65c0e0c ECX: f6b2fe98 EDX: 00000000 [ 10.997255] ESI: f65c0c00 EDI: 00000018 EBP: f6b2febc ESP: f6b2fe98 [ 10.997255] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 [ 10.997255] Process modprobe.bin (pid: 1141, ti=f6b2e000 task=f6a2caa0 task.ti=f6b2e000) [ 10.997255] Stack: [ 10.997255] 00000003 00000000 c030c2e9 f6b2feb8 f87ca221 00000000 f65c0e0c f6921400 [ 10.997255] 00000018 f6b2feec f882da8f 00000001 f6921438 f88358e4 436d6370 63304432 [ 10.997255] f65c0c00 f87cf9fe f69b42a0 f65c0d70 f65c0c00 f6b2fefc f87ce0bb f65c0e0c [ 10.997255] Call Trace: [ 10.997255] [<c030c2e9>] ? device_create_file+0x13/0x15 [ 10.997255] [<f87ca221>] ? snd_add_device_sysfs_file+0x67/0x6e [snd] [ 10.997255] [<f882da8f>] ? snd_pcm_dev_register+0x176/0x1de [snd_pcm] [ 10.997255] [<f87ce0bb>] ? snd_device_register_all+0x27/0x44 [snd] [ 10.997255] [<f87cadce>] ? snd_card_register+0x4c/0x131 [snd] [ 10.997255] [<f8d17687>] ? alsa_device_init+0x200/0x24b [saa7134_alsa] [ 10.997255] [<f8d1773d>] ? saa7134_alsa_init+0x6b/0x93 [saa7134_alsa] [ 10.997255] [<c0101137>] ? do_one_initcall+0x4a/0x115 [ 10.997255] [<f8d176d2>] ? saa7134_alsa_init+0x0/0x93 [saa7134_alsa] [ 10.997255] [<c0145205>] ? __blocking_notifier_call_chain+0x40/0x4c [ 10.997255] [<c015302d>] ? sys_init_module+0x87/0x18b [ 10.997255] [<c01031b4>] ? sysenter_do_call+0x12/0x28 [ 11.066012] Code: 00 00 5b 5d c3 55 89 e5 57 56 53 89 c3 83 ec 18 8b 53 30 c7 45 e0 00 00 00 00 8d 4d dc c7 45 dc 03 00 00 00 8b 30 83 e2 01 8b 06 <8b> 00 89 45 e4 8b 46 0c 89 45 e8 8b 43 0c 01 c0 09 c2 8d 45 f0· [ 11.066012] EIP: [<f8834cd4>] snd_pcm_timer_init+0x26/0xd1 [snd_pcm] SS:ESP 0068:f6b2fe98 [ 11.066012] CR2: 0000000000000002 [ 11.066318] ---[ end trace 000b75f3b1903e42 ]--- [ 14.079154] r8169: eth0: link up [ 14.095472] r8169: eth0: link up [ 14.335980] r8169: eth0: link up [ 17.074440] NET: Registered protocol family 10 [ 17.074879] lo: Disabled Privacy Extensions [ 20.520592] BUG: unable to handle kernel NULL pointer dereference at 00000002 <-- ********* 2nd nullptr deref [ 20.535353] IP: [<f8830abf>] snd_pcm_info+0x2e/0xe0 [snd_pcm] [ 20.535353] *pde = 00000000· [ 20.559932] Oops: 0000 [#2] SMP· [ 20.559932] last sysfs file: /sys/devices/pci0000:00/0000:00:14.4/0000:03:05.0/sound/card2/pcmC2D0c/pcm_class [ 20.586971] Modules linked in: ipv6 saa7134_alsa(+) usblp tda827x tda8290 snd_hda_codec_atihdmi tuner snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss saa7134 snd_ seq_midi_event snd_seq snd_seq_device ir_common snd_pcm_oss snd_mixer_oss v4l2_common videodev v4l1_compat videobuf_dma_sg videobuf_core snd_pcm tveeprom snd_timer snd i2c_piix4 shpchp k8temp soundcore s nd_page_alloc pcspkr r8169 ati_agp agpgart mii brd ata_generic pata_acpi ahci pata_atiixp libata [ 20.606741]· [ 20.606741] Pid: 1367, comm: hald Tainted: G D (2.6.30.4-126 #3) Unknow[ 20.704561] Process hald (pid: 1367, ti=f660e000 task=f6420c70 task.ti=f660e000) [ 20.704561] Stack: [ 20.704561] f65c0e0c f65c0c00 f6ada3f0 f642fe00 f6919ac0 f660fe34 f882e15a f65c0e0c [ 20.704561] 00000200 f6ada3f0 f6ada3c0 f6667000 f660fe3c f882e254 f660fe70 f87cb9e9 [ 20.704561] f680db00 0160fe70 ffffffff f64cb870 f6919ac0 00001000 00000020 00000020 [ 20.704561] Call Trace: [ 20.704561] [<f882e15a>] ? snd_pcm_proc_info_read+0x80/0x16c [snd_pcm] [ 20.704561] [<f882e254>] ? snd_pcm_stream_proc_info_read+0xe/0x10 [snd_pcm] [ 20.704561] [<f87cb9e9>] ? snd_info_entry_open+0x26b/0x2de [snd] [ 20.704561] [<c01df157>] ? proc_reg_open+0xb6/0x122 [ 20.704561] [<f87cb6ca>] ? snd_info_entry_release+0x0/0xa7 [snd] [ 20.704561] [<c01ac441>] ? __dentry_open+0x119/0x207 [ 20.704561] [<c01ac5c9>] ? nameidata_to_filp+0x2c/0x43 [ 20.704561] [<c01df0a1>] ? proc_reg_open+0x0/0x122 [ 20.704561] [<c01b656b>] ? do_filp_open+0x3ae/0x6c8 [ 20.704561] [<c019ff47>] ? free_pages_and_swap_cache+0x44/0x57 [ 20.704561] [<c01b6937>] ? getname+0x20/0xb7 [ 20.704561] [<c01be23c>] ? alloc_fd+0x55/0xbe [ 20.704561] [<c01ac253>] ? do_sys_open+0x44/0xba [ 20.704561] [<c01ac30b>] ? sys_open+0x1e/0x26 [ 20.704561] [<c01031b4>] ? sysenter_do_call+0x12/0x28 [ 20.704561] Code: 00 00 00 89 e5 57 56 53 89 d3 83 ec 08 89 df 89 45 ec 8b 00 8b 55 ec 89 45 f0 31 c0 8b 72 04 f3 ab 8b 55 f0 b9 40 00 00 00 8b 02 <8b> 00 89 43 0c 8b 42 0c 8b 55 ec 89 03 8b 42 30 89 43 08 8b 42· [ 20.704561] EIP: [<f8830abf>] snd_pcm_info+0x2e/0xe0 [snd_pcm] SS:ESP 0068:f660fe04 [ 21.199052] CR2: 0000000000000002 [ 21.222000] ---[ end trace 000b75f3b1903e43 ]--- [ 20.606741] EIP: 0060:[<f8830abf>] EFLAGS: 00010246 CPU: 1 [ 20.704561] EIP is at snd_pcm_info+0x2e/0xe0 [snd_pcm] [ 20.704561] EAX: 00000002 EBX: f642fe00 ECX: 00000040 EDX: f65c0c00 [ 20.704561] ESI: 00000000 EDI: f642ff20 EBP: f660fe18 ESP: f660fe04 [ 20.704561] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 The first dereference is happening at: (gdb) list *snd_pcm_timer_init+0x26 0x7cf8 is in snd_pcm_timer_init (/var/pisi/module-alsa-driver-1.0.20_20090808-49/work/alsa-driver/acore/../alsa-kernel/core/pcm_timer.c:131). 126 struct snd_timer_id tid; 127 struct snd_timer *timer; 128 129 tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; 130 tid.dev_class = SNDRV_TIMER_CLASS_PCM; 131 tid.card = substream->pcm->card->number; <---------- 132 tid.device = substream->pcm->device; 133 tid.subdevice = (substream->number << 1) | (substream->stream & 1); 134 if (snd_timer_new(substream->pcm->card, "PCM", &tid, &timer) < 0) 135 return;at the following place: Can it be related to a race condition between the sound card and the saa7134-alsa interface?

2 7

[alsa-devel] issue at 44.1 Khz capturing
by Harsha, Priya 10 Aug '09

10 Aug '09

Hi, I am using arecord for capturing data. When I try to record at 8 or 48 KHz, it's perfectly fine. But when I try to record at 44.1 Khz, there are spikes in the recorded data for every buffer. When I tried to tap the data captured at the firmware/hardware end, it's smooth but the same data when is checked from the user space after being recorded, it has spikes. On further analysis, looks like for every buffer recorded there is either 1 sample recorded extra or 1 sample dropped. But when I printed out the values passed from the pointer function, they are exactly at the intervals of 5513 bytes. Can anyone give me any clue as to what is going wrong? I am using indirect_* functions in the driver code. Is there any buffer size round off that is giving this error? Can I handle something in the driver to avoid this? Thanks, Harsha.

3 4

[alsa-devel] Implementing sync start
by Jon Smirl 10 Aug '09

10 Aug '09

I'm implementing sync start as Takashi described... --------------- If the hardware supports the simultaneous triggering of multiple streams, add the sync support to the driver. Namely, - Add SNDRV_PCM_INFO_SYNC_START flag to PCM info field - Call snd_pcm_set_sync(substream) at each open callback - In the trigger callback, check all bundled streams via snd_pcm_group_for_each_entry(). Mark all the streams but the current one via snd_pcm_trigger_done(). Then start all streams at once. Ditto for stop and pause. ------------- Is there a simple way to test this with something like a mono sine wav file? A aplay mono.wav and then use and ALSA conf to copy the sine to all channels and then sync start them. ----------- brutefit has this routine for sync start, does this look correct? int bfio_synch_start(void) { snd_pcm_status_t *status; int err, n; if (base_handle == NULL) { return 0; } /* FIXME: the SND_PCM_STATE_RUNNING code would not be needed if the bfio_write autostart hack was not there */ snd_pcm_status_alloca(&status); if (link_handles) { if ((err = snd_pcm_status(base_handle, status)) < 0) { fprintf(stderr, "ALSA I/O: Could not get status: %s.\n", snd_strerror(err)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_RUNNING) { return 0; } if ((err = snd_pcm_start(base_handle)) < 0) { fprintf(stderr, "ALSA I/O: Could not start audio: %s.\n", snd_strerror(err)); return -1; } return 0; } FOR_IN_AND_OUT { for (n = 0; n < n_handles[IO]; n++) { if ((err = snd_pcm_status(handles[IO][n], status)) < 0) { fprintf(stderr, "ALSA I/O: Could not get status: %s.\n", snd_strerror(err)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_RUNNING) { continue; } if ((err = snd_pcm_start(handles[IO][n])) < 0) { fprintf(stderr, "ALSA I/O: Could not start audio: %s.\n", snd_strerror(err)); return -1; } } } return 0; } -- Jon Smirl jonsmirl(a)gmail.com

3 12

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 10 Aug '09

10 Aug '09

The following changes since commit b7b8f9bf0cd73b90561e8123fd5ec28f4539c419: Janusz Krzysztofik (1): TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253 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 Chaithrika U S (1): ASoC: DaVinci: Support Audio on DA830 EVM Mark Brown (3): ASoC: Convert WM8776 to use factored out register cache code ASoC: Define more formats for the AC97 CODECs ASoC: Drop unneeded declaration of removed wm8731 SPI write function Uwe Kleine-König (1): ASoC: s3c2443-ac97: convert semaphore to mutex include/sound/soc-dai.h | 8 ++++- sound/soc/codecs/wm8731.c | 3 -- sound/soc/codecs/wm8776.c | 59 +++++++------------------------------ sound/soc/davinci/Kconfig | 11 +++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 33 +++++++++++++++++++++ sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++--- 7 files changed, 68 insertions(+), 57 deletions(-)

2 1

[alsa-devel] load_for_all_cards bug
by Kevin Goodsell 10 Aug '09

10 Aug '09

(I originally tried to send this via the moderator without subscribing to the list, but it never went through and I never received a moderator response. Apparently this list is set up as moderated, but nobody is actually moderating it.) Summary: load_for_all_cards fails with existing configuration elements In snd_config_hook_load_for_all_cards, the first call to snd_config_search attempts to locate an existing configuration node with the name of the driver. Typically none is found, and everything is good. However, if such a node is located, the next line assumes it is a leaf node with type 'string' and calls snd_config_get_string to fetch the string value. If this fails, the entire hook is abandoned. Because of this, setting something like the following in asoundrc: cards.<driver name>.foo 0 is sufficient to disable the entire card-specific configuration. As a concrete example, I have a HDA-Intel sound card. dmix.conf includes a way to set period_size, period_time, and periods by using configuration elements of the form cards.<driver name>.pcm.dmix.<var>. In ~/.asoundrc I add cards.HDA-Intel.pcm.dmix.period_size 1024 This will cause HDA-Intel.conf to fail to load, and the pcm defined in default.conf will fail to find the device-specific pcm cards.HDA-Intel.pcm.default, and fall back on the default pcm using plughw. By attempting to configure dmix, I have disabled it. Fixing this may just be a matter of changing the line in snd_config_hook_load_for_all_cards from: if (snd_config_search(root, fdriver, &n) >= 0) { to: if (snd_config_search(root, fdriver, &n) >= 0 && n->type != SND_CONFIG_TYPE_COMPOUND) { I have not tested this. -Kevin

2 1

[alsa-devel] [RFC][PATCH 0/3] Enhance ASoC on OMAP1510 by implementing DMA channel linking
by Janusz Krzysztofik 10 Aug '09

10 Aug '09

Current implementation of ASoC OMAP PCM makes use of DMA channel self linking feature for all machines but OMAP1510. This is caused by channel linking not supported on OMAP1510 in current OMAP DMA implementation. Instead, DMA interrputs are counted as periods, and after all periods in a buffer has been detected, DMA transfer is started again for a new buffer content. The period counter is also used as a software replacement for a hardware DMA progress counter that has been found unusable on OMAP1510 in case of playback. In my attempts to debug random PCM hangups, I was wondering what could happen if a single interrupt was missing, ie. the period index was not incremented for a period. It seems that the transfer would just hang after the current buffer processed, as DMA would not be started again, waiting for a missing period. Furthermore, software calculated PCM pointer accuracy can suffer from missing DMA interrupts. Even if OMAP1510 DMA hardware does not support channel linking in a way like newer models do, it has some other features that could be used for the purpose. Interrupt status flags, passed to OMAP PCM interrupt handler routine, can be used to get more accurate DMA progress counter software calculation. Even if these enhancements did not help me in eliminating random PCM hangups completely, I find them worth of sharing. There are three patches in the series: [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 [RFC][PATCH 3/3] ASoC: OMAP: Enhance OMAP1510 DMA progress software counter Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

1 0

[alsa-devel] alsamixer -a basic
by Elimar Riesebieter 10 Aug '09

10 Aug '09

alsa-lib 1.0.20 running alsamixer -a basic gives: ALSA lib python.c:991:(alsa_mixer_simple_finit) Unable to find python module '/usr/lib/alsa-lib/smixer/python/main.py' How do I build smixer/python/main.py ? Elimar -- On the keyboard of life you have always to keep a finger at the escape key;-)

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#539454: libasound2: Division by zero in pcm_rate.c when period_size->min == 0
by Elimar Riesebieter 10 Aug '09

10 Aug '09

forwarded 539454 alsa-devel(a)alsa-project.org tags 539454 + patch thanks Hi, this is Debian bug 539454. It would be better to fix it in general. Patching the Debian Package isn't a reasonable solution. Thanks Elimar * John Lindgren [090731 22:29 -0400] > Package: libasound2 > Version: 1.0.20-2 > Severity: important > Tags: patch > > > System: Dell Inspiron 1200 > Card: Intel ICH6 > Chip: SigmaTel STAC9752,53 > > > Steps to reproduce: > > 1. Disable dmix: mv /usr/share/alsa/cards/ICH4.conf /usr/share/alsa/cards/ICH4.conf.0 > 2. Restart ALSA: /etc/rcS.d/S50alsa-utils restart > 3. Clear Audacity configuration: rm -r ~/.audacity* > 4. Run Audacity 1.3.7-3 in Valgrind: valgrind --tool=none audacity > 5. Choose "OK" in the "Audacity First Run" dialog. > > > Valgrind output: > > ==26237== Process terminating with default action of signal 8 (SIGFPE) > ==26237== Integer divide by zero at address 0x64D1CD56 > ==26237== at 0x4CAD600: snd_pcm_rate_hw_refine_cchange (pcm_rate.c:189) > ==26237== by 0x4C97B2E: snd1_pcm_hw_refine_slave (pcm_params.c:2182) > ==26237== by 0x4CAD7D5: snd_pcm_rate_hw_refine (pcm_rate.c:222) > ==26237== by 0x4C96015: snd_pcm_hw_refine (pcm_params.c:2264) > ==26237== by 0x4CA116B: snd1_pcm_generic_hw_refine (pcm_generic.c:98) > ==26237== by 0x4C97B0D: snd1_pcm_hw_refine_slave (pcm_params.c:2158) > ==26237== by 0x4CC0E25: snd_pcm_lfloat_hw_refine (pcm_lfloat.c:261) > ==26237== by 0x4C96015: snd_pcm_hw_refine (pcm_params.c:2264) > ==26237== by 0x4C99068: sndrv_pcm_hw_params (pcm_params.c:2290) > ==26237== by 0x4CB209C: snd_pcm_plug_hw_params (pcm_plug.c:1045) > ==26237== by 0x4C9921D: sndrv_pcm_hw_params (pcm_params.c:2299) > ==26237== by 0x4C947B0: snd_pcm_hw_params (pcm.c:822) > > > Diagnosis: > > pcm_rate.c:189 reads: > > if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { > > When using the ICH4 driver without dmix, period_size->min == 0. Handle > this by changing pcm_rate.c:189 to: > > if (period_size->min > 0 && (buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { > > With this change, Audacity starts without problems. Patch against native 1.0.20: diff -Naurd a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c --- a/src/pcm/pcm_rate.c 2009-05-06 09:07:23.000000000 +0200 +++ b/src/pcm/pcm_rate.c 2009-08-06 21:34:23.000000000 +0200 @@ -186,7 +186,7 @@ if (!snd_interval_checkempty(period_size) && period_size->openmin && period_size->openmax && period_size->min + 1 == period_size->max) { - if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { + if (period_size->min > 0 && (buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { snd_interval_set_value(period_size, period_size->min); } else if ((buffer_size->max / period_size->max) * period_size->max == buffer_size->max) { snd_interval_set_value(period_size, period_size->max); -- Experience is something you don't get until just after you need it!

2 1

[alsa-devel] [BUG] NULL pointer dereference in patch_sigmatel.c
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, One of our users is having a NULL ptr dereference upon loading the snd_hda_intel module with 20090624's snapshot. There's only one commit after that date in patch_sigmatel.c so I didn't tell him to try with the latest snapshot but if you think that the bug may be related to another part of the ALSA codebase, I can make him try the latest snapshot. I'm attaching the alsa-info output on a working system with 1.0.18a. The kernel version in subject is 2.6.25.20. Thanks, Ozan ------------------------ First the BUG's trace: ------------------------ BUG: unable to handle kernel NULL pointer dereference at 00000074 IP: [<f8cdb83a>] :snd_hda_codec_idt:stac92xx_add_jack+0x84/0x9e *pde = 00000000 Oops: 0002 [#1] SMP Modules linked in: snd_hda_codec_idt snd_hda_intel(+) snd_hda_codec aes_i586 aes_generic ipv6 af_packet bridge bnep rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace cpufreq_conservative ndiswrapper vboxdrv pcmcia snd_hwdep rtc_cmos rtc_core yenta_socket tpm_infineon snd_seq_dummy rtc_lib tpm snd_seq_oss rsrc_nonstatic tpm_bios tifm_7xx1 iTCO_wdt snd_seq_midi_event tifm_core i2c_i801 snd_seq iTCO_vendor_support pcmcia_core r5u870 arc4 joydev snd_seq_device snd_pcm_oss nvidia(P) battery sg usbcam snd_mixer_oss ecb ac videobuf_dma_sg videobuf_core sony_laptop snd_pcm uvcvideo iwl4965 video compat_ioctl32 snd_timer output hci_usb iwlcore videodev thermal bluetooth snd processor v4l1_compat rfkill button led_class soundcore firmware_class snd_page_alloc mac80211 intel_agp i2c_core sky2 cfg80211 agpgart ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix uhci_hcd pata_acpi ehci_hcd usbcore ohci1394 ieee1394 ata_generic libata scsi_mod dock Pid: 1977, comm: modprobe Tainted: P (2.6.25.20-114 #1) EIP: 0060:[<f8cdb83a>] EFLAGS: 00210206 CPU: 0 EIP is at stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] EAX: 00000000 EBX: f8c7973e ECX: 00000004 EDX: f8cdeb1a ESI: 03211020 EDI: f675c600 EBP: f68d7d20 ESP: f68d7cd4 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process modprobe (pid: 1977, ti=f68d6000 task=f69ece60 task.ti=f68d6000) Stack: f68d7cf4 f8cdeb0a f8c796d5 f8c7973e f8c79793 00000001 000a69e0 f8c79793 4f205048 61207475 78452074 654c2074 4a207466 006b6361 f6898800 f6898800 00000000 f6898800 f6898800 f68d7d4c f8cdbac0 f65ef000 f6970000 00000001 Call Trace: [<f8cdbac0>] ? stac92xx_build_controls+0x26c/0x2d0 [snd_hda_codec_idt] [<f8c73aa9>] ? snd_hda_codec_build_controls+0x31/0x3d [snd_hda_codec] [<f8c750d3>] ? snd_hda_build_controls+0x18/0x67 [snd_hda_codec] [<f8c69138>] ? azx_probe+0x79e/0x840 [snd_hda_intel] [<f8c68857>] ? azx_send_cmd+0x0/0x101 [snd_hda_intel] [<f8c686a6>] ? azx_get_response+0x0/0x1b1 [snd_hda_intel] [<f8c67ece>] ? azx_attach_pcm_stream+0x0/0x15c [snd_hda_intel] [<f8c67b84>] ? azx_bus_reset+0x0/0x56 [snd_hda_intel] [<f8c67a40>] ? azx_power_notify+0x0/0x57 [snd_hda_intel] [<c01e7a37>] ? pci_device_probe+0x39/0x59 [<c024395f>] ? driver_probe_device+0xa0/0x136 [<c0243a50>] ? __driver_attach+0x5b/0x91 [<c024333c>] ? bus_for_each_dev+0x3b/0x63 [<c0243804>] ? driver_attach+0x14/0x16 [<c02439f5>] ? __driver_attach+0x0/0x91 [<c0242d3a>] ? bus_add_driver+0x9d/0x1ba [<c0243bc4>] ? driver_register+0x47/0xa7 [<c0168681>] ? __vunmap+0x93/0x9b [<c01e7bec>] ? __pci_register_driver+0x35/0x61 [<f8afd017>] ? alsa_card_azx_init+0x17/0x19 [snd_hda_intel] [<c0141f9c>] ? sys_init_module+0x18ad/0x19ca [<c0109c77>] ? do_syscall_trace+0x138/0x17f [<c0104a2e>] ? syscall_call+0x7/0xb ======================= Code: f9 ff 89 45 d0 89 f0 e8 29 68 f9 ff 89 c3 89 f0 e8 32 68 f9 ff ff 75 d0 53 50 68 0a eb cd f8 8d 45 d4 50 e8 45 25 50 c7 8b 47 08 <89> 78 74 8b 47 08 83 c4 14 c7 40 78 7f a2 cd f8 31 c0 8d 65 f4 EIP: [<f8cdb83a>] stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] SS:ESP 0068:f68d7cd4 ---[ end trace 4a628a5b4fc302e7 ]--- alsa-info output: ------------------- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Sun Feb 8 19:15:27 EET 2009 !!Linux Distribution !!------------------ Pardus 2008.2 Canis aureus !!Kernel Information !!------------------ Kernel release: 2.6.25.20-114 Operating System: GNU/Linux Architecture: i686 Processor: Intel(R) Core(TM)2 Duo CPU T8100 @ 2.10GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.18a Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 21 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) 09:04.2 Mass storage controller: Texas Instruments 5-in-1 Multimedia Card Reader (SD/MMC/MS/MS PRO/xD) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 104d:9008 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=vaio !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : vaio,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: SigmaTel STAC9872AK Address: 0 Vendor Id: 0x83847662 Subsystem Id: 0x104d1e00 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=5, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x07 Processing caps: benign=0, ncoeff=0 Node 0x07 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x0e Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x0d 0x0d] Connection: 1 0x15 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03211020: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x80: HP VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03a15030: [Jack] Mic at Ext Left Conn = 1/8, Color = Red DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x3e0]: 44100 48000 88200 96000 176400 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x11 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 2 0x10* 0x09 Node 0x12 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x13 Node 0x13 [Pin Complex] wcaps 0x440381: Stereo Digital Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Delay: 4 samples Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x90a7013e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=1 Amp-Out vals: [0x00 0x00] Connection: 4 0x0a 0x0d 0x14* 0x02 Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x17 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=0, val=127 Connection: 4 0x02* 0x03 0x04 0x05 Node 0x18 [Audio Output] wcaps 0x40201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Delay: 4 samples Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x104d1700 Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Şub 8 2009 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Şub 8 2009 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Şub 8 2009 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Şub 8 19:14 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Şub 8 19:15 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Şub 8 2009 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Şub 8 2009 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # PulseAudio plugin configuration # Let's create a virtual device "pulse" for mixer and PCM pcm.pulse { type pulse hint { description "PulseAudio Sound Server" } } ctl.pulse { type pulse hint { description "PulseAudio Sound Server" } } # Let's make it the default! pcm.!default { type pulse hint { description "Default" } } ctl.!default { type pulse hint { description "Default" } } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xfc300000 irq 21' Mixer name : 'SigmaTel STAC9872AK' Components : 'HDA:83847662,104d1e00,00100201 HDA:14f12c06,104d1700,00100000' Controls : 10 Simple ctrls : 7 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 255 Mono: Capture [off] Front Left: Playback 0 [0%] [-51.00dB] Front Right: Playback 0 [0%] [-51.00dB] Simple mixer control 'Mic Jack',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 13 [87%] [19.50dB] [on] Front Right: Capture 13 [87%] [19.50dB] [on] Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 127 value.1 127 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Speaker Playback Volume' value.0 127 value.1 127 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 13 value.1 13 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Mic Jack' comment.item.1 'Internal Mic' comment.item.2 PCM iface MIXER name 'Capture Source' value 'Internal Mic' } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 127 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.10 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module aes_i586 aes_generic ipv6 bridge bnep af_packet rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace ndiswrapper r5u870 usbcam videobuf_dma_sg snd_hda_intel snd_seq_dummy snd_seq_oss snd_seq_midi_event videobuf_core nvidia snd_seq uvcvideo snd_seq_device snd_pcm_oss arc4 compat_ioctl32 ecb snd_mixer_oss intel_agp snd_pcm iwl4965 snd_timer snd_page_alloc snd_hwdep snd pcmcia hci_usb rtc_cmos bluetooth yenta_socket rsrc_nonstatic tpm_infineon pcmcia_core tifm_7xx1 sky2 battery videodev iTCO_wdt agpgart ac thermal processor iwlcore rfkill led_class firmware_class i2c_i801 mac80211 video output tpm tpm_bios rtc_core rtc_lib joydev iTCO_vendor_support tifm_core sony_laptop v4l1_compat i2c_core sg button soundcore cfg80211 ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix ohci1394 ieee1394 uhci_hcd pata_acpi ata_generic libata scsi_mod dock ehci_hcd usbcore

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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: McASP driver enhacements
by Chaithrika U S 10 Aug '09

10 Aug '09

On DA830/OMAPL137 SoC, the McASP peripheral has internal FIFO support. It provides additional data buffering. It also provides tolerance to vairiation in host/DMA controller response times. The read and write FIFO sizes are 256 bytes each. If FIFO is enabled, the DMA events from McASP are sent to the FIFO which inturn sends DMA requests to the host CPU according to the thresholds programmed. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber= sprufm1&fileType=pdf This patch adds support for FIFO configuration.The platform member has a version field which differentiates the McASP on different SoCs. The patch also adds another DAI format to the driver. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 113 ++++++++++++++++++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 5 ++ sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-pcm.h | 1 + 4 files changed, 115 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f0c0347..eca22d7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -102,6 +102,11 @@ /* Receive Buffer for Serializer n */ #define DAVINCI_MCASP_RXBUF_REG 0x280 +/* McASP FIFO Registers */ +#define DAVINCI_MCASP_WFIFOCTL (0x1010) +#define DAVINCI_MCASP_WFIFOSTS (0x1014) +#define DAVINCI_MCASP_RFIFOCTL (0x1018) +#define DAVINCI_MCASP_RFIFOSTS (0x101C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -276,6 +281,13 @@ */ #define TXDATADMADIS BIT(0) +/* + * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits + */ +#define FIFO_ENABLE BIT(16) +#define NUMEVT_MASK (0xFF << 8) +#define NUMDMA_MASK (0xFF) + #define DAVINCI_MCASP_NUM_SERIALIZER 16 static inline void mcasp_set_bits(void __iomem *reg, u32 val) @@ -345,6 +357,9 @@ static void mcasp_start_rx(struct davinci_audio_dev *dev) static void mcasp_start_tx(struct davinci_audio_dev *dev) { + u8 offset = 0, i; + u32 cnt; + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); @@ -353,6 +368,19 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) { + offset = i; + break; + } + } + + /* wait for TX ready */ + cnt = 0; + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + TXSTATE) && (cnt < 100000)) + cnt++; + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); } @@ -362,6 +390,13 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) mcasp_start_tx(dev); else mcasp_start_rx(dev); + + /* enable FIFO */ + if (dev->txnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -382,6 +417,13 @@ static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) mcasp_stop_tx(dev); else mcasp_stop_rx(dev); + + /* disable FIFO */ + if (dev->txnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -401,7 +443,16 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); @@ -505,6 +556,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) { int i; + u8 tx_ser = 0; + u8 rx_ser = 0; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -525,12 +578,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) + if (dev->serial_dir[i] == TX_MODE) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); - else if (dev->serial_dir[i] == RX_MODE) + tx_ser++; + } else if (dev->serial_dir[i] == RX_MODE) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); + rx_ser++; + } + } + + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt * tx_ser > 64) + dev->txnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + + if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (dev->rxnumevt * rx_ser > 64) + dev->rxnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } @@ -543,6 +621,8 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < active_slots; i++) mask |= (1 << i); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ @@ -622,8 +702,16 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; int word_length; + u8 numevt; davinci_hw_common_param(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + numevt = dev->txnumevt; + else + numevt = dev->rxnumevt; + + if (!numevt) + numevt = 1; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -650,6 +738,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); return -EINVAL; } + + if (dev->version == MCASP_VERSION_2) { + dma_params->data_type *= numevt; + dma_params->acnt = 4 * numevt; + } else + dma_params->acnt = dma_params->data_type; + davinci_config_channel_size(dev, word_length); return 0; @@ -778,6 +873,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; dev->codec_fmt = pdata->codec_fmt; + dev->version = pdata->version; + dev->txnumevt = pdata->txnumevt; + dev->rxnumevt = pdata->rxnumevt; dma_data[count].name = "I2S PCM Stereo out"; dma_data[count].eventq_no = pdata->eventq_no; @@ -807,9 +905,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } dma_data[count].channel = res->start; - davinci_mcasp_dai[pdev->id].private_data = dev; - davinci_mcasp_dai[pdev->id].dev = &pdev->dev; - ret = snd_soc_register_dai(&davinci_mcasp_dai[pdev->id]); + davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) goto err_release_region; @@ -827,12 +925,13 @@ err_release_dev: static int davinci_mcasp_remove(struct platform_device *pdev) { + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_pcm_dma_params *dma_data; struct davinci_audio_dev *dev; struct resource *mem; - snd_soc_unregister_dai(&davinci_mcasp_dai[pdev->id]); - dev = davinci_mcasp_dai[pdev->id].private_data; + snd_soc_unregister_dai(&davinci_mcasp_dai[pdata->op_mode]); + dev = davinci_mcasp_dai[pdata->op_mode].private_data; clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 36b7104..554354c 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -50,6 +50,11 @@ struct davinci_audio_dev { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; }; #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 8fd0c3c..1e2dfd0 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -67,6 +67,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; unsigned int data_type; + unsigned short acnt; unsigned int count; period_size = snd_pcm_lib_period_bytes(substream); @@ -91,11 +92,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } + acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); edma_set_src_index(lch, src_bidx, 0); edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index eb4287f..63d9625 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -19,6 +19,7 @@ struct davinci_pcm_dma_params { char *name; /* stream identifier */ int channel; /* sync dma channel ID */ + unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ -- 1.5.6

4 4

[alsa-devel] ASoC: PCM formats for AC97
by Jon Smirl 09 Aug '09

09 Aug '09

Tell me if this is right and I'll do a patch.... ----------------------------------------------------------- The current mpc5200 code has: static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, struct snd_soc_dai psc_ac97_dai[] = { { .playback = { .formats = SNDRV_PCM_FMTBIT_S32_BE, I think I have this backwards, the ac97 driver can take 8/16/24/32 it's the Bestcomm program that doesn't know about anything except 32b. So if we wrote new Bestcomm programs that expanded bytes from memory out into 32b DMA register writes, we could support 8/16b formats. The correct formats would be: static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S32_BE, struct snd_soc_dai psc_ac97_dai[] = { { .playback = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, After implementing new Bestcomm programs.... static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, ----------------------------------------------------------- A related issue in soc-dai.h: #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) shouldn't this be #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16 |\ SNDRV_PCM_FMTBIT_S32) The first form says the AC97 hardware can take both endians, but it doesn't take both endians it takes the endian matching the machine it is attached to. Don't the standard AC97 formats also include 8 and 24b? -- Jon Smirl jonsmirl(a)gmail.com

3 6

[alsa-devel] Testers wanted: New SB X-Fi driver
by Frashman 09 Aug '09

09 Aug '09

Hi Takashi, first of all, thank you for developing this driver. I tested it today. It works, so far. But two annoying things. As you mentioned, I got problems, if the onboard sound modules are loaded. Then i get no sound out auf my xfi. If I disable the loading of snd-hda-intel modules while booting, then it works. The second problem is, that SIDE and SURROUND connectors are interchanged. I'm absolutly sure, it works fine under windows and the connector colors are also correct. If I run speaker-test -c8 the SIDE Speakers sounds come out of my REAR (SURROUND) Speakers, while nothing is coming out of my rear speakers, if i'm using speaker-test -c6 Here lspci: $ lspci 00:00.0 Host bridge: ATI Technologies Inc RD780 Northbridge only dual slot PCI-e_GFX and HT1 K8 part 00:02.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (external gfx0 port A) 00:04.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port A) 00:0a.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port F) 00:12.0 SATA controller: ATI Technologies Inc SB600 Non-Raid-5 SATA 00:13.0 USB Controller: ATI Technologies Inc SB600 USB (OHCI0) 00:13.1 USB Controller: ATI Technologies Inc SB600 USB (OHCI1) 00:13.2 USB Controller: ATI Technologies Inc SB600 USB (OHCI2) 00:13.3 USB Controller: ATI Technologies Inc SB600 USB (OHCI3) 00:13.4 USB Controller: ATI Technologies Inc SB600 USB (OHCI4) 00:13.5 USB Controller: ATI Technologies Inc SB600 USB Controller (EHCI) 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 14) 00:14.1 IDE interface: ATI Technologies Inc SB600 IDE 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 00:14.3 ISA bridge: ATI Technologies Inc SB600 PCI to LPC Bridge 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:00.0 VGA compatible controller: ATI Technologies Inc RV770 [Radeon HD 4850] 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 02:00.0 Audio device: Creative Labs X-Fi Titanium series [EMU20k2] (rev 03) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) 04:06.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 04:06.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 04:0e.0 FireWire (IEEE 1394): Texas Instruments TSB43AB23 IEEE-1394a-2000 Controller (PHY/Link) Thank you so far, Frashman

7 33

[alsa-devel] Auzentech X-Fi Prelude with alsa drivers
by Adrian Patzak 09 Aug '09

09 Aug '09

Dear sirs, I've got a Auzentech X-Fi Prelude and I installed the alsa unstable snapshot drivers. Stereo sound for my headphones and ditital output are working fine but my microphone isn't recording anything. I was checking the mixer settings but i wasn't able to find something wrong. Are the x-fi drivers just not designed to fit with the Auzentech card or do i have a local configuration problem? The output of alsa-info.sh script is attached. Thanking you in anticipation for any answer. upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Fri Aug 7 10:48:57 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!DMI Information !!--------------- Manufacturer: FUJITSU SIEMENS Product Name: AMD690VM-FMH !!Kernel Information !!------------------ Kernel release: 2.6.28-14-generic Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_ctxfi !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [XFi ]: SB-XFi - Creative X-Fi Creative X-Fi 20K1 Unknown !!PCI Soundcards installed in the system !!-------------------------------------- 03:01.0 Multimedia audio controller: Creative Labs SB X-Fi !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 03:01.0 0401: 1102:0005 Subsystem: 415a:0034 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_ctxfi enable : Y,Y,Y,Y,Y,Y,Y,Y id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 multiple : 2 reference_rate : 48000 use_system_timer : N !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Aug 7 08:13 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 24 Aug 7 12:02 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Aug 7 12:02 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 17 Aug 7 08:13 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 18 Aug 7 08:13 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 19 Aug 7 08:13 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 20 Aug 7 08:13 /dev/snd/pcmC0D4p crw-rw----+ 1 root audio 116, 33 Aug 7 08:13 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY E: core-util.c: Home directory /home/adrian not ours. **** List of PLAYBACK Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: ctxfi [Front/WaveIn] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 1: ctxfi [Surround] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 2: ctxfi [Center/LFE] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 3: ctxfi [Side] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 4: ctxfi [IEC958 Non-audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD E: core-util.c: Home directory /home/adrian not ours. **** List of CAPTURE Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: ctxfi [Front/WaveIn] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [XFi] Card hw:0 'XFi'/'Creative X-Fi 20K1 Unknown' Mixer name : '20K1' Components : '' Controls : 30 Simple ctrls : 11 Simple mixer control 'Master',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] Simple mixer control 'PCM',0 Capabilities: pvolume cvolume cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'Front',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Center/LFE',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Side',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Line-in',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'Mic',0 Capabilities: pvolume cvolume cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [on] Simple mixer control 'Digit-IO',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'S/PDIF-in',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'S/PDIF-out',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.XFi { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 256 value.1 256 } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 256 value.1 256 } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Line-in Playback Volume' value.0 256 value.1 256 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Mic Playback Volume' value.0 256 value.1 256 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-in Playback Volume' value.0 256 value.1 256 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-out Playback Volume' value.0 256 value.1 256 } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 256 value.1 256 } control.8 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 256 value.1 256 } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Center/LFE Playback Volume' value.0 256 value.1 256 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 256 value.1 256 } control.11 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Capture Volume' value.0 256 value.1 256 } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'PCM Capture Volume' value.0 256 value.1 256 } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Line-in Capture Volume' value.0 256 value.1 256 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Mic Capture Volume' value.0 256 value.1 256 } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-in Capture Volume' value.0 256 value.1 256 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PCM Capture Switch' value false } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line-in Capture Switch' value false } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Capture Switch' value true } control.19 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-in Capture Switch' value false } control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line-in Playback Switch' value false } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-out Playback Switch' value true } control.22 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-in Playback Switch' value false } control.23 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Front Playback Switch' value true } control.24 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Surround Playback Switch' value false } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Center/LFE Playback Switch' value false } control.26 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Side Playback Switch' value false } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Digit-IO Playback Switch' value true } control.28 { comment.access read comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback Mask' value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.29 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.30 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback PCM Stream' value '0000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } } --endcollapse-- !!All Loaded Modules !!------------------ Module nls_iso8859_1 nls_cp437 vfat fat binfmt_misc ppdev bridge stp bnep video output input_polldev sbp2 lp parport joydev snd_ctxfi snd_pcm snd_timer snd usbhid i2c_piix4 soundcore snd_page_alloc psmouse serio_raw usblp pcspkr nvidia usb_storage ohci1394 ieee1394 r8169 mii fbcon tileblit font bitblit softcursor !!ALSA/HDA dmesg !!------------------

1 0

[alsa-devel] Problems with gain controls
by James Courtier-Dutton 09 Aug '09

09 Aug '09

Hi, Does anyone know how to disable pulse from making any changes to the alsa hardware mixer controls? I would prefer pulse to use software gain controls, and not touch the alsamixer levels. pulse seems to do a very bizarre thing. The single master gain control within pulse changes both the alsamixer PCM and Master controls. Kind Regards James

2 3

[alsa-devel] Typo in soc/atmel/sam9g20_wm8731.c
by Pedro I. Sanchez 08 Aug '09

08 Aug '09

Hello, This is a patch to fix what I believe is a typo in this file. --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 173a239..baa42f3 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -341,7 +341,7 @@ static int __init at91sam9g20ek_init(void) } pllb = clk_get(NULL, "pllb"); - if (IS_ERR(mclk)) { + if (IS_ERR(pllb)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk;

1 0

[alsa-devel] [PATCH 2/2] ASoC: DaVinci: Support Audio on DA830 EVM
by Chaithrika U S 08 Aug '09

08 Aug '09

Add support for audio on DA830 EVM- here McASP1 is interfaced to TLV320AIC3106 codec. 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/Kconfig | 11 +++++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 33 +++++++++++++++++++++++++++++++++ 3 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..677a538 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -41,3 +41,14 @@ config SND_DAVINCI_SOC_SFFSDR help Say Y if you want to add support for SoC audio on Lyrtech SFFSDR board. + +config SND_DA830_SOC_EVM + tristate "SoC Audio support for DA830/OMAPL137 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + + help + Say Y if you want to add support for SoC audio on TI + DA830/OMAPL137 EVM + diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..5e2195f 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-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(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..a5a9212 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,6 +54,9 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_davinci_da830_evm()) + sysclk = 24576000; + else return -EINVAL; @@ -162,6 +165,14 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { @@ -179,6 +190,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +static struct snd_soc_card da830_snd_soc_card = { + .name = "DA830 EVM", + .dai_link = &da830_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +209,11 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +static struct aic3x_setup_data da830_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, @@ -205,6 +228,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { .codec_data = &dm6467_evm_aic3x_setup, }; +/* evm audio subsystem */ +static struct snd_soc_device da830_evm_snd_devdata = { + .card = &da830_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &da830_evm_aic3x_setup, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +252,9 @@ 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_da830_evm()) { + evm_snd_dev_data = &da830_evm_snd_devdata; + index = 1; } else return -EINVAL; -- 1.5.6

2 1

[alsa-devel] [PATCH 2/6] s3c2443-ac97: convert semaphore to mutex
by Uwe Kleine-König 08 Aug '09

08 Aug '09

This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig): CC [M] sound/soc/s3c24xx/s3c2443-ac97.o sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX' sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read': sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function) sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.) sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write': sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function) Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Timur Tabi <timur(a)freescale.com> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 3f03d5d..bf16f20 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97; static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; -static DECLARE_MUTEX(ac97_mutex); +static DEFINE_MUTEX(ac97_mutex); static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, u32 ac_codec_cmd; u32 stat, addr, data; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," " rep addr = %02x\n", reg, addr); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); return (unsigned short)data; } @@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, u32 ac_glbctrl; u32 ac_codec_cmd; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); } -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
by Candelaria Villareal, Jorge 07 Aug '09

07 Aug '09

Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6 line, controlled by register INTBR_PMBR1. Machine driver takes care of enabling gpio line through i2c and codec driver manipulates the line during headset ramp up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c51594d..f7e5b74 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,6 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -39,6 +40,9 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +#define TWL4030_INTBR_PMBR1 0x0D +#define EXTMUTE(value) (value << 2) + static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -280,6 +284,7 @@ static struct snd_soc_card snd_soc_sdp3430 = { static struct twl4030_setup_data twl4030_setup = { .ramp_delay_value = 3, .sysclk = 26000, + .hs_extmute = 1, }; /* Audio subsystem */ @@ -312,6 +317,10 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + /* Set TWL4030 GPIO6 as EXTMUTE signal */ + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), + TWL4030_MODULE_INTBR); + ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; -- 1.6.0.4

3 6

[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 07 Aug '09

07 Aug '09

We've encountered strange behavior in the alsamixer settings using the wm9712 codec. If we unmute the headphone output and then unmute the PCM output, the headphone output gets reset to mute in the hardware register. At this point the hardware register does not match the value in the register cache. I've spent some time debugging this, and the headphone setting is set outside of any code path that would call the ac97_write() routine. As best as I can tell, there is something strange going on in hardware. I've provided this patch that works around the problem. Have any of you seen this before? Is this patch the right approach? - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

5 14

[alsa-devel] Fixed module builds of soc/fsl/*.c
by Takashi Iwai 07 Aug '09

07 Aug '09

JFYI, I merged the patch below now. Takashi --- From afc5e65245255a268ab22a20477ed2c9f2cdfcd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 7 Aug 2009 16:33:53 +0200 Subject: ASoC: Add missing DRV_NAME definitions for fsl/* drivers Module builds are broken due to missing DRV_NAME for efika-audio-fabric and pcm030-audio-fabric. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/fsl/efika-audio-fabric.c | 2 ++ sound/soc/fsl/pcm030-audio-fabric.c | 2 ++ 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 85b0e75..3326e2a 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -30,6 +30,8 @@ #include "mpc5200_psc_ac97.h" #include "../codecs/stac9766.h" +#define DRV_NAME "efika-audio-fabric" + static struct snd_soc_device device; static struct snd_soc_card card; diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8766f7a..b928ef7 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -30,6 +30,8 @@ #include "mpc5200_psc_ac97.h" #include "../codecs/wm9712.h" +#define DRV_NAME "pcm030-audio-fabric" + static struct snd_soc_device device; static struct snd_soc_card card; -- 1.6.3.3

3 2

[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA830 EVM
by Chaithrika U S 07 Aug '09

07 Aug '09

Define resources for McASP1 used on DA830 EVM, add platform device defintion, initialization function. Additionally, this patch also adds version and FIFO related members to platform data structure. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to DaVinci GIT tree arch/arm/mach-davinci/board-da830-evm.c | 22 +++++++++++++ arch/arm/mach-davinci/da830.c | 48 ++++++++++++++++++++++++++-- arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++ arch/arm/mach-davinci/include/mach/da8xx.h | 2 + 4 files changed, 84 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 9025669..4643795 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -23,6 +23,7 @@ #include <mach/irqs.h> #include <mach/cp_intc.h> #include <mach/da8xx.h> +#include <mach/asp.h> #define DA830_EVM_PHY_MASK 0x0 #define DA830_EVM_MDIO_FREQUENCY 2200000 /* PHY bus frequency */ @@ -51,6 +52,25 @@ static struct davinci_uart_config da830_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +static u8 da830_iis_serializer_direction[] = { + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, TX_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da830_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da830_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + static __init void da830_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -93,6 +113,8 @@ static __init void da830_evm_init(void) davinci_serial_init(&da830_evm_uart_config); i2c_register_board_info(1, da830_evm_i2c_devices, ARRAY_SIZE(da830_evm_i2c_devices)); + + da830_init_mcasp1(&da830_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da830.c b/arch/arm/mach-davinci/da830.c index 3a9b634..a160a7f 100644 --- a/arch/arm/mach-davinci/da830.c +++ b/arch/arm/mach-davinci/da830.c @@ -23,6 +23,7 @@ #include <mach/common.h> #include <mach/time.h> #include <mach/da8xx.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -411,9 +412,9 @@ static struct davinci_clk da830_clks[] = { CLK("eqep.0", NULL, &eqep0_clk), CLK("eqep.1", NULL, &eqep1_clk), CLK("da830_lcdc", NULL, &lcdc_clk), - CLK("soc-audio.0", NULL, &mcasp0_clk), - CLK("soc-audio.1", NULL, &mcasp1_clk), - CLK("soc-audio.2", NULL, &mcasp2_clk), + CLK("davinci-mcasp.0", NULL, &mcasp0_clk), + CLK("davinci-mcasp.1", NULL, &mcasp1_clk), + CLK("davinci-mcasp.2", NULL, &mcasp2_clk), CLK("musb_hdrc", NULL, &usb20_clk), CLK(NULL, "aemif", &aemif_clk), CLK(NULL, "aintc", &aintc_clk), @@ -1198,6 +1199,47 @@ static struct davinci_soc_info davinci_soc_info_da830 = { .emac_pdata = &da8xx_emac_pdata, }; +static struct resource da830_mcasp1_resources[] = { + { + .name = "mcasp1", + .start = DAVINCI_DA830_MCASP1_REG_BASE, + .end = DAVINCI_DA830_MCASP1_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .end = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AREVT, + .end = DAVINCI_DA830_DMA_MCASP1_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da830_mcasp1_device = { + .name = "davinci-mcasp", + .id = 1, + .num_resources = ARRAY_SIZE(da830_mcasp1_resources), + .resource = da830_mcasp1_resources, +}; + +void __init da830_init_mcasp1(struct snd_platform_data *pdata) +{ + int ret; + + ret = da8xx_pinmux_setup(da830_mcasp1_pins); + if (ret) + pr_warning("da830_evm_init: mcasp1 mux setup failed: %d\n", + ret); + + da830_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&da830_mcasp1_device); +} + void __init da830_init(void) { davinci_common_init(&davinci_soc_info_da830); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index cdf1f44..4df3839 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da830 register banks */ +#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 + /* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 @@ -26,6 +29,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da830 */ +#define DAVINCI_DA830_DMA_MCASP1_AREVT 2 +#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT @@ -43,6 +50,14 @@ struct snd_platform_data { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + u8 txnumevt; + u8 rxnumevt; +}; + +enum { + MCASP_VERSION_1 = 0, /* DM646x */ + MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; #define INACTIVE_MODE 0 diff --git a/arch/arm/mach-davinci/include/mach/da8xx.h b/arch/arm/mach-davinci/include/mach/da8xx.h index a8cb570..c2aabf1 100644 --- a/arch/arm/mach-davinci/include/mach/da8xx.h +++ b/arch/arm/mach-davinci/include/mach/da8xx.h @@ -15,6 +15,7 @@ #include <mach/edma.h> #include <mach/i2c.h> #include <mach/emac.h> +#include <mach/asp.h> /* * The cp_intc interrupt controller for the da8xx isn't in the same @@ -65,6 +66,7 @@ int da8xx_register_edma(void); int da8xx_register_i2c(int instance, struct davinci_i2c_platform_data *pdata); int da8xx_register_watchdog(void); int da8xx_register_emac(void); +void __init da830_init_mcasp1(struct snd_platform_data *pdata); extern struct platform_device da8xx_serial_device; extern struct emac_platform_data da8xx_emac_pdata; -- 1.5.6

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[alsa-devel] suspicious code in aplay.c
by Dmitri Vorobiev 07 Aug '09

07 Aug '09

Hello, The following code in alsa-utils/aplay/aplay.c looks suspicious to me: 1215 if (!quiet_mode) 1216 fprintf(stderr, _("Suspended. Trying resume. ")); fflush(stderr); Is it really necessary to flush the stderr stream even if there was no call to fprintf()? Perhaps these lines have to be like that: if (!quiet_mode) { fprintf(stderr, _("Suspended. Trying resume. ")); fflush(stderr); } Thanks, Dmitri P.S. I'm not subscribed to the mailing list, so please Cc: me on reply.

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 07 Aug '09

07 Aug '09

The following changes since commit 6d7f68a1eab3d5b3c0a6a5cf9434c77cd3a8c6ac: Janusz Krzysztofik (1): ASoC: add support for Amstrad E3 (Delta) machine 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 There's a merge up of a for-2.6.31 change in here which I've not pushed yet since the change is relatively invasive for so late in the release cycle (the OMAP simultaneous audio change). I didn't redo the branch in case it does get pushed; worst case it's just an extra merge commit which isn't the end of the world. Daniel Mack (1): ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device Janusz Krzysztofik (3): ASoC: CX20442: add some debugging ASoC: CX20442: simplify codec controller usage TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253 Jarkko Nikula (1): ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop John Bonesio (1): ASoC: MPC5200: Support for buffer wrap around Mark Brown (12): ASoC: Begin to factor out register cache I/O functions ASoC: Remove use of hw_read from TLV320AIC3x driver ASoC: Factor out WM8580 register cache code Merge branch 'tlv320aic3x' into reg-cache ASoC: Add I/O control bus information to factored out cache setup ASoC: Factor out I2C 8 bit address 16 bit data I/O ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs ASoC: Factor out 7 bit register 9 bit data SPI write ASoC: Existing S3C24xx AC97 drivers should depend on S3C24xx ASoC: Add WM8776 CODEC driver Merge branch 'for-2.6.31' into for-2.6.32 Merge branch 'reg-cache' into for-2.6.32 Troy Kisky (2): ASoC: DaVinci: i2s: don't bounce through rtd to get dai ASoC: DaVinci: pcm, constrain buffer size to multiple of period arch/arm/plat-omap/include/mach/mcbsp.h | 4 +- arch/arm/plat-omap/mcbsp.c | 50 ++- include/linux/tty.h | 2 +- include/sound/soc.h | 11 +- sound/soc/Makefile | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 27 +- sound/soc/codecs/cx20442.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8510.c | 174 +++----- sound/soc/codecs/wm8523.c | 98 +--- sound/soc/codecs/wm8580.c | 154 ++----- sound/soc/codecs/wm8728.c | 111 +---- sound/soc/codecs/wm8731.c | 141 ++----- sound/soc/codecs/wm8750.c | 154 ++----- sound/soc/codecs/wm8776.c | 781 +++++++++++++++++++++++++++++++ sound/soc/codecs/wm8776.h | 51 ++ sound/soc/codecs/wm8900.c | 319 +++++-------- sound/soc/codecs/wm8903.c | 244 ++++------- sound/soc/codecs/wm8940.c | 143 ++---- sound/soc/codecs/wm8960.c | 200 +++----- sound/soc/codecs/wm8961.c | 231 ++++------ sound/soc/codecs/wm8971.c | 127 ++---- sound/soc/codecs/wm8988.c | 138 ++---- sound/soc/codecs/wm8990.c | 194 +++----- sound/soc/codecs/wm9081.c | 229 ++++------ sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/davinci/davinci-pcm.c | 5 + sound/soc/fsl/mpc5200_dma.c | 17 + sound/soc/omap/ams-delta.c | 6 +- sound/soc/omap/omap-mcbsp.c | 10 +- sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/soc-cache.c | 218 +++++++++ 34 files changed, 2048 insertions(+), 1831 deletions(-) create mode 100644 sound/soc/codecs/wm8776.c create mode 100644 sound/soc/codecs/wm8776.h create mode 100644 sound/soc/soc-cache.c

2 1

[alsa-devel] [PATCH 0/2] Give the N_AMSDELTA line discipline more generic name
by Janusz Krzysztofik 07 Aug '09

07 Aug '09

This series of patches renames the N_AMSDELTA line discipline, used by the Amstrad Delta ASoC machine driver and the CX20442 ASoC codec driver for controlling the CX20442 voice codec over the CX81801 voice modem, giving it a more generic name: N_V253. The name has been chosen after the voice modem standard codename V.253. I think that even if the code of the line discipline is currently implemented inside those two specific drivers, the idea, if not the code, could be reused with other similar hardware configurations, if necessary. As the line discipline is conceived to be registered from a machine driver, there should be no conflicts, even if used by several different drivers that happen to be be built-in side-by-side. [PATCH 1/2] TTY: Rename N_AMSDELTA line discipline to N_V253 [PATCH 2/2] ASoC: AMS_DELTA: Use updated line discipline name Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

2 1

[alsa-devel] Audio support for Simtec Hermes
by Ben Dooks 07 Aug '09

07 Aug '09

Add support for the TLV320AIC33 codec on the Simtec Hermes board, using the core work done for the TLV320AIC23 codecs. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- sound/soc/s3c24xx/Kconfig | 7 + sound/soc/s3c24xx/Makefile | 3 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 151 ++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+) Index: b/sound/soc/s3c24xx/Kconfig =================================================================== --- a/sound/soc/s3c24xx/Kconfig 2009-08-04 12:27:06.000000000 +0100 +++ b/sound/soc/s3c24xx/Kconfig 2009-08-04 12:29:22.000000000 +0100 @@ -127,3 +127,10 @@ config SND_S3C24XX_SOC_SIMTEC_TLV320AIC2 select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC23 select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC Index: b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c 2009-08-04 12:42:00.000000000 +0100 @@ -0,0 +1,151 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP} */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { + .i2c_bus = 0, + .i2c_already = 1, +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); Index: b/sound/soc/s3c24xx/Makefile =================================================================== --- a/sound/soc/s3c24xx/Makefile 2009-08-04 12:27:06.000000000 +0100 +++ b/sound/soc/s3c24xx/Makefile 2009-08-04 12:29:22.000000000 +0100 @@ -24,6 +24,7 @@ snd-soc-ln2440sbc-alc650-objs := ln2440s snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o 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 obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -36,4 +37,6 @@ obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_A obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(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 + -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] PROC_COEF/PROC_INDEX question
by Wu Fengguang 07 Aug '09

07 Aug '09

Hello, I'm not sure of the usage for GET_PROC_COEF/GET_COEF_INDEX, and found that their return value is unstable (changing all the time) for codec: Codec: Realtek ALC888 Address: 2 Function Id: 0x1 Vendor Id: 0x10ec0888 Subsystem Id: 0x80860023 Revision Id: 0x100202 For example: # hda-verb /dev/snd/hwC0D2 0x20 GET_PROC_COEF 0 nid = 0x20, verb = 0xc00, param = 0x0 value = 0x80 # hda-verb /dev/snd/hwC0D2 0x20 GET_COEF_INDEX 0 nid = 0x20, verb = 0xd00, param = 0x0 value = 0xe # hda-verb /dev/snd/hwC0D2 0x20 GET_PROC_COEF 0 nid = 0x20, verb = 0xc00, param = 0x0 value = 0x9c00 # hda-verb /dev/snd/hwC0D2 0x20 GET_COEF_INDEX 0 nid = 0x20, verb = 0xd00, param = 0x0 value = 0xf I wonder if this is the normal case? Thanks, Fengguang

2 4

[alsa-devel] Audio support for TLV320AIC23 on Simtec S3C24XX products
by Ben Dooks 07 Aug '09

07 Aug '09

Add support for the TLV320AIC23 on most of the current Simtec S3C24XX with an initial core set of routines for use with the other S3C24XX designs that do not use the TLV320AIC23. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- arch/arm/common/sbcd/s3c24xx/dev-audio.c | 146 ++++++--- arch/arm/plat-s3c/include/plat/audio-simtec.h | 37 ++ arch/arm/plat-s3c24xx/audio-simtec.c | 97 +----- sound/soc/s3c24xx/Kconfig | 12 sound/soc/s3c24xx/Makefile | 4 sound/soc/s3c24xx/s3c24xx_simtec.c | 391 +++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 + sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 ++++++++ 8 files changed, 728 insertions(+), 118 deletions(-) Index: b/sound/soc/s3c24xx/Kconfig =================================================================== --- a/sound/soc/s3c24xx/Kconfig 2009-08-04 12:24:09.000000000 +0100 +++ b/sound/soc/s3c24xx/Kconfig 2009-08-06 11:53:38.000000000 +0100 @@ -115,3 +115,15 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X select SND_S3C24XX_SOC_I2S select SND_SOC_L3 select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC Index: b/sound/soc/s3c24xx/Makefile =================================================================== --- a/sound/soc/s3c24xx/Makefile 2009-08-04 12:24:09.000000000 +0100 +++ b/sound/soc/s3c24xx/Makefile 2009-08-06 11:53:38.000000000 +0100 @@ -23,6 +23,8 @@ snd-soc-smdk2440-wm8991-objs := smdk2440 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.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 @@ -33,3 +35,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(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_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o Index: b/sound/soc/s3c24xx/s3c24xx_simtec.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c 2009-08-04 12:29:02.000000000 +0100 @@ -0,0 +1,391 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +#define DEBUG + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_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 *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, curently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); Index: b/arch/arm/plat-s3c/include/plat/audio-simtec.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/arch/arm/plat-s3c/include/plat/audio-simtec.h 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,37 @@ +/* arch/arm/plat-s3c/include/plat/audio-simtec.h + * + * Copyright 2008 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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. + * + * Simtec Audio support. +*/ + +/** + * struct s3c24xx_audio_simtec_pdata - platform data for simtec audio + * @use_mpllin: Select codec clock from MPLLin + * @output_cdclk: Need to output CDCLK to the codec + * @have_mic: Set if we have a MIC socket + * @have_lout: Set if we have a LineOut socket + * @amp_gpio: GPIO pin to enable the AMP + * @amp_gain: Option GPIO to control AMP gain + */ +struct s3c24xx_audio_simtec_pdata { + unsigned int use_mpllin:1; + unsigned int output_cdclk:1; + + unsigned int have_mic:1; + unsigned int have_lout:1; + + int amp_gpio; + int amp_gain[2]; + + void (*startup)(void); +}; + +extern int simtec_audio_add(const char *codec_name, + struct s3c24xx_audio_simtec_pdata *pdata); Index: b/arch/arm/plat-s3c24xx/audio-simtec.c =================================================================== --- a/arch/arm/plat-s3c24xx/audio-simtec.c 2009-08-04 12:05:59.000000000 +0100 +++ b/arch/arm/plat-s3c24xx/audio-simtec.c 2009-08-04 12:24:11.000000000 +0100 @@ -45,7 +45,7 @@ #include <plat/audio.h> #include <plat/regs-iis.h> -#include <asm/plat-s3c24xx/audio-simtec.h> +#include <plat/audio-simtec.h> #include <plat/devs.h> #include <plat/pm.h> @@ -56,7 +56,7 @@ static inline int has_lr_routing(void) return machine_is_bast() || machine_is_vr1000(); } -static int simtec_audio_startup(void *pw) +static void simtec_audio_startup(void) { unsigned int tmp; unsigned long flags; @@ -71,94 +71,40 @@ static int simtec_audio_startup(void *pw } local_irq_restore(flags); - - return 0; -} - -static void simtec_audio_shutdown(void *pw) -{ - unsigned int tmp; - unsigned long flags; - - /* switch off LR clock routing */ - - local_irq_save(flags); - - if (has_lr_routing()) { - tmp = __raw_readb(BAST_VA_CTRL1); - tmp &= ~BAST_CPLD_CTRL1_LRMASK; - tmp |= BAST_CPLD_CTRL1_LRCOFF; - __raw_writeb(tmp, BAST_VA_CTRL1); - } - - local_irq_restore(flags); -} - -static int simtec_audio_matchdev(struct device *dev) -{ - printk(KERN_INFO "%s: dev = %p\n", __func__, dev); - - if (dev->parent == NULL || dev->parent->parent == NULL) - return 0; - - printk(KERN_INFO "%s: %s\n", __func__, dev_name(dev->parent)); - if (strncmp(dev_name(dev->parent), "i2c", strlen("i2c")) != 0) - return 0; - - printk(KERN_INFO "%s: %s\n", __func__, dev_name(dev->parent->parent)); - if (strncmp(dev_name(dev->parent->parent), "s3c24", - strlen("s3c24")) == 0) - return 1; - - return 0; } -/* device registration info */ - -static struct s3c24xx_iis_ops simtec_audio_ops = { - .startup = simtec_audio_startup, - .shutdown = simtec_audio_shutdown, +static struct s3c24xx_audio_simtec_pdata simtec_audio_platdata = { }; -static struct s3c24xx_platdata_iis simtec_audio_platdata = { - .ops = &simtec_audio_ops, - .codec_clk = "clkout0", - .match_dev = simtec_audio_matchdev, -}; +static char our_name[32]; static struct platform_device audio_dev = { - .name = "s3c24xx-tlv320aic23", + .name = our_name, + .id = -1, .dev = { .parent = &s3c_device_iis.dev, .platform_data = &simtec_audio_platdata, }, }; -static struct platform_device audio_dev2 = { - .name = "s3c24xx-tlv320aic33", - .dev = { - .parent = &s3c_device_iis.dev, - .platform_data = &simtec_audio_platdata, - }, -}; - -static void __init simtec_audio_cdclk_mpllin(void) +int __init simtec_audio_add(const char *name, + struct s3c24xx_audio_simtec_pdata *spd) { - simtec_audio_platdata.flags = AUDIO_CDCLK_MPLL | AUDIO_CDCLK_OUTPUT; -} + if (!name) + name = "tlv320aic23"; -int __init simtec_audio_add(unsigned int is_aic33, - unsigned int cdclk_master) -{ - if (cdclk_master) - simtec_audio_cdclk_mpllin(); + snprintf(our_name, sizeof(our_name)-1, "s3c24xx-simtec-%s", name); - platform_device_register(&s3c_device_iis); + if (!spd) { + printk(KERN_ERR "%s: no source platadata\n", __func__); + return -EINVAL; + } - if (is_aic33) - platform_device_register(&audio_dev2); - else - platform_device_register(&audio_dev); + /* copy platform data so the source can be __initdata */ + simtec_audio_platdata = *spd; + + platform_device_register(&s3c_device_iis); + platform_device_register(&audio_dev); return 0; } @@ -172,7 +118,8 @@ static int __init simtec_audio_init(void printk(KERN_INFO "Simtec TLV320AIC23 Audio platform driver\n"); - simtec_audio_shutdown(NULL); + if (has_lr_routing()) + simtec_audio_platdata.startup = simtec_audio_startup; platform_device_register(&s3c_device_iis); return platform_device_register(&audio_dev); Index: b/arch/arm/common/sbcd/s3c24xx/dev-audio.c =================================================================== --- a/arch/arm/common/sbcd/s3c24xx/dev-audio.c 2009-08-04 12:05:59.000000000 +0100 +++ b/arch/arm/common/sbcd/s3c24xx/dev-audio.c 2009-08-04 12:24:11.000000000 +0100 @@ -12,6 +12,9 @@ #include <linux/kernel.h> #include <linux/ioport.h> #include <linux/device.h> +#include <linux/ioport.h> +#include <linux/gpio.h> +#include <linux/io.h> #include "s3c24xx.h" #include <asm/mach/sbcd-utils.h> @@ -24,22 +27,69 @@ #include "../devices/platform.h" #include <plat/devs.h> -#include <asm/plat-s3c24xx/audio-simtec.h> +#include <plat/audio-simtec.h> + +enum amp_type { + AMP_VR2K, + AMP_HERMES, /* GAIN0 = GPD9, GAIN1 = GPD10 */ +}; + +static void __iomem *reg; +static enum amp_type type; + +static void amp_gpio_set(struct gpio_chip *chip, + unsigned offset, int value) +{ + u32 to_write; -static unsigned char is_cdclk_master; + if (type == AMP_VR2K) { + to_write = value ? 0x01 : 0x03; + } else { + to_write = value ? 0x03 : 0x02; + } + + writeb(to_write, reg); +} + +static int amp_gpio_output(struct gpio_chip *chip, + unsigned offset, int value) +{ + amp_gpio_set(chip, offset, value); + return 0; +} + +static struct gpio_chip amp_gpio = { + .label = "amp", + .direction_output = amp_gpio_output, + .set = amp_gpio_set, + .ngpio = 1, + .base = -1, +}; + +static struct s3c24xx_audio_simtec_pdata __initdata pdata = { +}; static __init int s3c24xx_drv_bast_audio_attach(struct sbcd_state *state, struct sbcd_item *parent, struct sbcd_item *drv_it) { - int is_tlv320aic33 = sbcd_board_has_driver("tlv320aic33"); + char *name = NULL; /* some vr2000s do not have the necessary setup in their - * configuration data, so force is_cdclk_master here. */ - if (sbcd_board_has_driver("vr2k-amp")) - is_cdclk_master = 1; + * configuration data, so force cdclk_master here. */ + + if (sbcd_board_has_driver("vr2k-amp") || + sbcd_board_has_driver("hermes-amp")) { + pdata.use_mpllin = 1; + pdata.output_cdclk = 1; + } - return simtec_audio_add(is_tlv320aic33, is_cdclk_master); + if (sbcd_board_has_driver("hermes-amp")) + name = "hermes-snd"; + + pdata.amp_gpio = amp_gpio.base; + + return simtec_audio_add(name, &pdata); } @@ -61,8 +111,10 @@ static __init int s3c24xx_drv_audio_atta sbcd_get_optvals(ptr, &optname, &optval); if (strcmp(optname, "cdclk") == 0 && - strcmp(optval, "master") == 0) - is_cdclk_master = 1; + strcmp(optval, "master") == 0) { + pdata.use_mpllin = 1; + pdata.output_cdclk = 1; + } break; } } @@ -76,49 +128,57 @@ static __init int s3c24xx_drv_audio_atta sbcd_drv_decl2(cpu_audio, "cpu-audio", s3c24xx_drv_audio_attach); sbcd_drv_decl2(bast_audio, "bast-audio", s3c24xx_drv_bast_audio_attach); -static struct resource vr2k_amp_resource[] = { - [0] = { - .start = S3C2410_CS1 + 0xc0, - .end = S3C2410_CS1 + 0xc1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct resource hermes_amp_resource[] = { - [0] = { - .start = S3C2410_CS1 + 0x140, - .end = S3C2410_CS1 + 0x141, - .flags = IORESOURCE_MEM, - }, -}; static __init int s3c24xx_drv_amp_attach(struct sbcd_state *state, - struct sbcd_item *parent, - struct sbcd_item *drv_it) + enum amp_type amp_type) { - struct platform_device *pdev; + unsigned long base; int ret; - ret = sbcd_dev_platform_create(state, parent, drv_it, &pdev); - if (ret) { - printk(KERN_ERR "%s: no platform device\n", __func__); - return -ENODEV; + if (state->pass != SBCD_PASS_GPIO) + return 0; + + type = amp_type; + + if (amp_type == AMP_VR2K) { + base = S3C2410_CS1 + 0xc0; + } else { + base = S3C2410_CS1 + 0x140; + pdata.amp_gain[0] = S3C2410_GPD(9); + pdata.amp_gain[1] = S3C2410_GPD(10); } - if (pdev->num_resources == 0) { - if (strcmp(pdev->name, "vr2k-amp") == 0) { - pdev->resource = vr2k_amp_resource; - pdev->num_resources = ARRAY_SIZE(vr2k_amp_resource); - } else if (strcmp(pdev->name, "hermes-amp") == 0) { - pdev->resource = hermes_amp_resource; - pdev->num_resources = ARRAY_SIZE(hermes_amp_resource); - } + /* note, we don't request the region, other gpio blocks may + * be sharing it. */ + + reg = ioremap(base, 2); + if (!reg) { + printk(KERN_ERR "%s: failed to map regs\n", __func__); + return -ENXIO; + } + + ret = gpiochip_add(&amp_gpio); + if (!ret) { + printk(KERN_ERR "%s: failed to add gpio\n", __func__); + return ret; } - pdev->dev.parent = &s3c_device_iis.dev; - platform_device_register(pdev); return 0; } -sbcd_drv_decl2(vr2k_amp, "vr2k-amp", s3c24xx_drv_amp_attach); -sbcd_drv_decl2(hermes_amp, "hermes-amp", s3c24xx_drv_amp_attach); +static __init int s3c24xx_drv_vr2k_amp_attach(struct sbcd_state *state, + struct sbcd_item *parent, + struct sbcd_item *drv_it) +{ + return s3c24xx_drv_amp_attach(state, AMP_VR2K); +} + +static __init int s3c24xx_drv_hermes_amp_attach(struct sbcd_state *state, + struct sbcd_item *parent, + struct sbcd_item *drv_it) +{ + return s3c24xx_drv_amp_attach(state, AMP_HERMES); +} + +sbcd_drv_decl2_gpio(vr2k_amp, "vr2k-amp", s3c24xx_drv_vr2k_amp_attach); +sbcd_drv_decl2_gpio(hermes_amp, "hermes-amp", s3c24xx_drv_hermes_amp_attach); Index: b/sound/soc/s3c24xx/s3c24xx_simtec.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif Index: b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] [PATCH][2.6.31-rc5] ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop
by Jarkko Nikula 07 Aug '09

07 Aug '09

Simultaneous audio playback and capture on OMAP1510 can cause that second stream is stalled if there is enough delay between startup of the audio streams. Current implementation of the omap_mcbsp_start is starting both transmitter and receiver at the same time and it is called only for firstly started audio stream from the OMAP McBSP based ASoC DAI driver. Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is missed if there is no DMA transfer set up at that time when the first word after McBSP startup is transmitted. The problem hasn't noted before since later OMAPs are using level sensitive DMA request lines. Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by allowing to start and stop individually McBSP transmitter and receiver logics. Then call those functions individually for both audio playback and capture streams. This ensures that DMA transfer is setup before transmitter or receiver is started. Thanks to Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> for detailed problem analysis and Peter Ujfalusi <peter.ujfalusi(a)nokia.com> for info about DMA request line behavior differences between the OMAP generations. Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Tony Lindgren <tony(a)atomide.com> --- arch/arm/plat-omap/include/mach/mcbsp.h | 4 +- arch/arm/plat-omap/mcbsp.c | 50 +++++++++++++++++++----------- sound/soc/omap/omap-mcbsp.c | 10 +++--- 3 files changed, 39 insertions(+), 25 deletions(-) diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index bb154ea..57249bb 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -387,8 +387,8 @@ void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config); int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); -void omap_mcbsp_start(unsigned int id); -void omap_mcbsp_stop(unsigned int id); +void omap_mcbsp_start(unsigned int id, int tx, int rx); +void omap_mcbsp_stop(unsigned int id, int tx, int rx); void omap_mcbsp_xmit_word(unsigned int id, u32 word); u32 omap_mcbsp_recv_word(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index efa0e01..a3d2313 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -328,14 +328,15 @@ void omap_mcbsp_free(unsigned int id) EXPORT_SYMBOL(omap_mcbsp_free); /* - * Here we start the McBSP, by enabling the sample - * generator, both transmitter and receivers, - * and the frame sync. + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. */ -void omap_mcbsp_start(unsigned int id) +void omap_mcbsp_start(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -348,32 +349,40 @@ void omap_mcbsp_start(unsigned int id) mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7; mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7; - /* Start the sample generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Start the sample generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + } /* Enable transmitter and receiver */ w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (tx & 1)); w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1)); udelay(100); - /* Start frame sync */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + if (idle) { + /* Start frame sync */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + } /* Dump McBSP Regs */ omap_mcbsp_dump_reg(id); } EXPORT_SYMBOL(omap_mcbsp_start); -void omap_mcbsp_stop(unsigned int id) +void omap_mcbsp_stop(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -386,15 +395,20 @@ void omap_mcbsp_stop(unsigned int id) /* Reset transmitter */ w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(tx & 1)); /* Reset receiver */ w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(rx & 1)); - /* Reset the sample rate generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + } } EXPORT_SYMBOL(omap_mcbsp_stop); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a5d46a7..6a837ff 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -183,21 +183,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); - int err = 0; + int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcbsp_data->active++) - omap_mcbsp_start(mcbsp_data->bus_id); + mcbsp_data->active++; + omap_mcbsp_start(mcbsp_data->bus_id, play, !play); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcbsp_data->active) - omap_mcbsp_stop(mcbsp_data->bus_id); + omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + mcbsp_data->active--; break; default: err = -EINVAL; -- 1.6.3.3

4 6

[alsa-devel] tlv320aic3x: Allow driver to use already registered i2c device
by Ben Dooks 07 Aug '09

07 Aug '09

Add the ability for the driver to use an already registered i2c device instead of creating one of them itself. Add the field i2c_already field to the setup data and not set the i2c bus address the driver will then use the device already setup. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 9 insertions(+), 1 deletion(-) Index: b/sound/soc/codecs/tlv320aic3x.c =================================================================== --- a/sound/soc/codecs/tlv320aic3x.c 2009-08-06 11:53:30.000000000 +0100 +++ b/sound/soc/codecs/tlv320aic3x.c 2009-08-06 15:29:45.000000000 +0100 @@ -1310,6 +1310,7 @@ static int aic3x_i2c_remove(struct i2c_c static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", 0 }, + { "tlv320aic33", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1405,6 +1406,10 @@ static int aic3x_probe(struct platform_d codec->hw_write = (hw_write_t) i2c_master_send; codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = aic3x_add_i2c_device(pdev, setup); + } else if (setup->i2c_already) { + codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = (hw_read_t) aic3x_i2c_read; + i2c_add_driver(&aic3x_i2c_driver); } #else /* Add other interfaces here */ @@ -1421,6 +1426,7 @@ static int aic3x_remove(struct platform_ { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct aic3x_setup_data *setup = socdev->codec_data; /* power down chip */ if (codec->control_data) @@ -1429,7 +1435,8 @@ static int aic3x_remove(struct platform_ snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); + if (setup && !setup->i2c_already) + i2c_unregister_device(codec->control_data); i2c_del_driver(&aic3x_i2c_driver); #endif kfree(codec->private_data); Index: b/sound/soc/codecs/tlv320aic3x.h =================================================================== --- a/sound/soc/codecs/tlv320aic3x.h 2009-08-06 11:53:30.000000000 +0100 +++ b/sound/soc/codecs/tlv320aic3x.h 2009-08-06 12:11:03.000000000 +0100 @@ -283,6 +283,7 @@ int aic3x_button_pressed(struct snd_soc_ struct aic3x_setup_data { int i2c_bus; + unsigned char i2c_already; /* i2c device already registered */ unsigned short i2c_address; unsigned int gpio_func[2]; }; -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] usb audio
by bruce 07 Aug '09

07 Aug '09

Hi I have the TASCAM US-122L USB.I am using ubuntu studio 9.04.The usb audio inputs and outputs are no functional cant fined them anywhere I am using jack connection kit on my studio the midi are recognized but not the audio.I have submitted to launchpad this report.Here are my jack start up log this is with tascam us-122L set as input and output.It will start with just tascam input but there is no audio only midi with that start. jackd 0.116.1 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. server `default' registered 23:53:21.206 JACK was started with PID=8295. registered builtin port type 32 bit float mono audio registered builtin port type 8 bit raw midi clock source = system clock via clock_gettime start poll on 3 fd's loading driver .. new client: alsa_pcm, id = 1 type 1 @ 0x9821d48 fd = -1 apparent rate = 44100 creating alsa driver ... hw:1|hw:1|1024|3|44100|0|0|hwmon|hwmeter|-|32bit control device hw:1 23:53:21.383 ALSA active patchbay scan... 23:53:21.407 jack-imput-output003: 14:0 Midi Through -> 20:0 TASCAM US-122L checked. 23:53:21.416 ALSA connection change. 23:53:21.811 ALSA active patchbay scan... 23:53:21.813 jack-imput-output003: 14:0 Midi Through -> 20:0 TASCAM US-122L checked. ALSA: Cannot open PCM device alsa_pcm for playback. Falling back to capture-only mode cannot load driver module alsa starting server engine shutdown server thread back from poll freeing shared port segments stopping server thread last xrun delay: 0.000 usecs max delay reported by backend: 0.000 usecs freeing engine shared memory max usecs: 0.000, engine deleted cleaning up shared memory cleaning up files unregistering server `default' 23:53:22.586 JACK was stopped successfully. 23:53:22.588 Post-shutdown script... 23:53:22.590 killall jackd jackd: no process killed 23:53:23.015 Post-shutdown script terminated with exit status=256. 23:53:23.727 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. thank you for any help you mite give :-) Bruce McCandless

1 0

[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Daniel Ribeiro 06 Aug '09

06 Aug '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..5b9ed64 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

6 6

[alsa-devel] [PATCH 2/2] ASoC: AMS_DELTA: Use updated line discipline name
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch updates the tty line discipline name used in the Amstrad Delta machine driver. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/ams-delta.c.orig 2009-08-05 21:44:35.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/ams-delta.c 2009-08-05 23:30:53.000000000 +0200 @@ -519,7 +519,7 @@ static int ams_delta_cx20442_init(struct } /* Register optional line discipline for over the modem control */ - ret = tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + ret = tty_register_ldisc(N_V253, &cx81801_ops); if (ret) { dev_warn(card->dev, "Failed to register line discipline, " @@ -644,9 +644,9 @@ static void __exit ams_delta_module_exit } } - if (tty_unregister_ldisc(N_AMSDELTA) != 0) + if (tty_unregister_ldisc(N_V253) != 0) dev_warn(&ams_delta_audio_platform_device->dev, - "failed to unregister AMSDELTA line discipline\n"); + "failed to unregister V253 line discipline\n"); snd_soc_jack_free_gpios(&ams_delta_hook_switch, ARRAY_SIZE(ams_delta_hook_switch_gpios),

1 0

[alsa-devel] [PATCH 1/2] TTY: Rename N_AMSDELTA line discipline to N_V253
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

The patch changes the line discipline name registered in include/linux/tty.h Signed-off-by: Janusz Krzysztofik --- --- linux-2.6.31-rc5/include/linux/tty.h.orig 2009-08-05 21:44:33.000000000 +0200 +++ linux-2.6.31-rc5/include/linux/tty.h 2009-08-05 22:54:12.000000000 +0200 @@ -47,7 +47,7 @@ #define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */ #define N_PPS 18 /* Pulse per Second */ -#define N_AMSDELTA 19 /* codec control over modem, board specific */ +#define N_V253 19 /* Codec control over voice modem */ /* * This character is the same as _POSIX_VDISABLE: it cannot be used as

1 0

[alsa-devel] [PATCH 0/2] ASoC: CX2044: full duplex mode fix
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This series of patches tries to solve the problem of audio mode changes not working correctly while in full duplex mode. [PATCH 1/2] ASoC: CX20442: add some debugging [PATCH 2/2] ASoC: CX20442: simplify codec controller usage Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

2 1

[alsa-devel] [PATCH 2/2] ASoC: CX20442: simplify codec controller usage
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch is a workaround for the problem of several subsequent control statements not being applied correctly to the codec controller (modem). In order to follow the hook switch state change from handset to handsfree while in full duplex mode, two consecutive +VLS control commands were sent to the modem. The first one was M1 (microphone only), the seconds one was M1S1 (both microphone and speaker). As there was no real modem handshaking procedure implemented, neither in the codec nor in the machine driver part of the line discipline, the modem was having the second command missed. Since a possibility to switch to microphone only mode (and speaker only mode as well) seams of no value, I have modified the code to issue single M1S1 command only for any of those cases. Tested on my Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/codecs/cx20442.c.orig 2009-08-04 23:41:03.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/codecs/cx20442.c 2009-08-05 00:56:30.000000000 +0200 @@ -138,9 +138,7 @@ static int cx20442_pm_to_v253_vls(u8 val case 0: return V253_VLS_T; case (1 << CX20442_SPKOUT): - return V253_VLS_S1; case (1 << CX20442_MIC): - return V253_VLS_M1; case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): return V253_VLS_M1S1; case (1 << CX20442_TELOUT):

1 0

[alsa-devel] [PATCH 1/2] ASoC: CX20442: add some debugging
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch adds debugging statement that can help in tracing how the driver is trying to control the codec device. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/codecs/cx20442.c.orig 2009-08-04 23:41:03.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/codecs/cx20442.c 2009-08-05 00:56:30.000000000 +0200 @@ -203,6 +201,7 @@ static int cx20442_write(struct snd_soc_ if (unlikely(len > (ARRAY_SIZE(buf) - 1))) return -ENOMEM; + dev_dbg(codec->dev, "%s: %s\n", __func__, buf); if (codec->hw_write(codec->control_data, buf, len) != len) return -EIO;

1 0

[alsa-devel] [PATCH] ASoC: Add WM8776 CODEC driver
by Mark Brown 06 Aug '09

06 Aug '09

The WM8776 is a high performance, stereo audio CODEC with five channel input selector. The WM8776 is ideal for surround sound processing applications for home hi-fi, DVD-RW and other audio visual equipment. This driver implements support for most WM8776 features - currently the ADC automatic level control/limiter functionality is omitted. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8776.c | 781 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8776.h | 51 +++ 4 files changed, 838 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8776.c create mode 100644 sound/soc/codecs/wm8776.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a..8cd24b0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C @@ -132,6 +133,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8776 + tristate + config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f265380..e80f164 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -21,6 +21,7 @@ snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8776-objs := wm8776.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o @@ -52,6 +53,7 @@ obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c new file mode 100644 index 0000000..ee12bf8 --- /dev/null +++ b/sound/soc/codecs/wm8776.c @@ -0,0 +1,781 @@ +/* + * wm8776.c -- WM8776 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + * + * TODO: Input ALC/limiter support + */ + +#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/spi/spi.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 "wm8776.h" + +static struct snd_soc_codec *wm8776_codec; +struct snd_soc_codec_device soc_codec_dev_wm8776; + +/* codec private data */ +struct wm8776_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8776_CACHEREGNUM]; + int sysclk[2]; +}; + +#ifdef CONFIG_SPI_MASTER +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len); +#endif + +static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { + 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ + 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ + 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ + 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ + 0xa6, 0x01, 0x01 +}; + +/* + * read wm8776 register cache + */ +static inline unsigned int wm8776_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8776_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8776 register cache + */ +static inline void wm8776_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8776_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8776 register space + */ +static int wm8776_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8776_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int wm8776_reset(struct snd_soc_codec *codec) +{ + return wm8776_write(codec, WM8776_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(hp_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new wm8776_snd_controls[] = { +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8776_HPLVOL, WM8776_HPRVOL, + 0, 127, 0, hp_tlv), +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8776_DACLVOL, WM8776_DACRVOL, + 0, 255, 0, dac_tlv), +SOC_SINGLE("Digital Playback ZC Switch", WM8776_DACCTRL1, 0, 1, 0), + +SOC_SINGLE("Deemphasis Switch", WM8776_DACCTRL2, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Volume", WM8776_ADCLVOL, WM8776_ADCRVOL, + 0, 255, 0, adc_tlv), +SOC_DOUBLE("Capture Switch", WM8776_ADCMUX, 7, 6, 1, 1), +SOC_DOUBLE_R("Capture ZC Switch", WM8776_ADCLVOL, WM8776_ADCRVOL, 8, 1, 0), +SOC_SINGLE("Capture HPF Switch", WM8776_ADCIFCTRL, 8, 1, 1), +}; + +static const struct snd_kcontrol_new inmix_controls[] = { +SOC_DAPM_SINGLE("AIN1 Switch", WM8776_ADCMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AIN2 Switch", WM8776_ADCMUX, 1, 1, 0), +SOC_DAPM_SINGLE("AIN3 Switch", WM8776_ADCMUX, 2, 1, 0), +SOC_DAPM_SINGLE("AIN4 Switch", WM8776_ADCMUX, 3, 1, 0), +SOC_DAPM_SINGLE("AIN5 Switch", WM8776_ADCMUX, 4, 1, 0), +}; + +static const struct snd_kcontrol_new outmix_controls[] = { +SOC_DAPM_SINGLE("DAC Switch", WM8776_OUTMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8776_OUTMUX, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_INPUT("AIN1"), +SND_SOC_DAPM_INPUT("AIN2"), +SND_SOC_DAPM_INPUT("AIN3"), +SND_SOC_DAPM_INPUT("AIN4"), +SND_SOC_DAPM_INPUT("AIN5"), + +SND_SOC_DAPM_MIXER("Input Mixer", WM8776_PWRDOWN, 6, 1, + inmix_controls, ARRAY_SIZE(inmix_controls)), + +SND_SOC_DAPM_ADC("ADC", "Capture", WM8776_PWRDOWN, 1, 1), +SND_SOC_DAPM_DAC("DAC", "Playback", WM8776_PWRDOWN, 2, 1), + +SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + outmix_controls, ARRAY_SIZE(outmix_controls)), + +SND_SOC_DAPM_PGA("Headphone PGA", WM8776_PWRDOWN, 3, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("VOUT"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +}; + +static const struct snd_soc_dapm_route routes[] = { + { "Input Mixer", "AIN1 Switch", "AIN1" }, + { "Input Mixer", "AIN2 Switch", "AIN2" }, + { "Input Mixer", "AIN3 Switch", "AIN3" }, + { "Input Mixer", "AIN4 Switch", "AIN4" }, + { "Input Mixer", "AIN5 Switch", "AIN5" }, + + { "ADC", NULL, "Input Mixer" }, + + { "Output Mixer", "DAC Switch", "DAC" }, + { "Output Mixer", "AUX Switch", "AUX" }, + { "Output Mixer", "Bypass Switch", "Input Mixer" }, + + { "VOUT", NULL, "Output Mixer" }, + + { "Headphone PGA", NULL, "Output Mixer" }, + + { "HPOUTL", NULL, "Headphone PGA" }, + { "HPOUTR", NULL, "Headphone PGA" }, +}; + +static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int reg, iface, master; + + switch (dai->id) { + case WM8776_DAI_DAC: + reg = WM8776_DACIFCTRL; + master = 0x80; + break; + case WM8776_DAI_ADC: + reg = WM8776_ADCIFCTRL; + master = 0x100; + break; + default: + return -EINVAL; + } + + iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + /* FIXME: CHECK A/B */ + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0007; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x00c; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x008; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x004; + break; + default: + return -EINVAL; + } + + /* Finally, write out the values */ + snd_soc_update_bits(codec, reg, 0xf, iface); + snd_soc_update_bits(codec, WM8776_MSTRCTRL, 0x180, master); + + return 0; +} + +static int mclk_ratios[] = { + 128, + 192, + 256, + 384, + 512, + 768, +}; + +static int wm8776_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + int iface_reg, iface; + int ratio_shift, master; + int i; + + iface = 0; + + switch (dai->id) { + case WM8776_DAI_DAC: + iface_reg = WM8776_DACIFCTRL; + master = 0x80; + ratio_shift = 4; + break; + case WM8776_DAI_ADC: + iface_reg = WM8776_ADCIFCTRL; + master = 0x100; + ratio_shift = 0; + break; + default: + return -EINVAL; + } + + + /* Set word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x30; + break; + } + + /* Only need to set MCLK/LRCLK ratio if we're master */ + if (snd_soc_read(codec, WM8776_MSTRCTRL) & master) { + for (i = 0; i < ARRAY_SIZE(mclk_ratios); i++) { + if (wm8776->sysclk[dai->id] / params_rate(params) + == mclk_ratios[i]) + break; + } + + if (i == ARRAY_SIZE(mclk_ratios)) { + dev_err(codec->dev, + "Unable to configure MCLK ratio %d/%d\n", + wm8776->sysclk[dai->id], params_rate(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]); + + snd_soc_update_bits(codec, WM8776_MSTRCTRL, + 0x7 << ratio_shift, i << ratio_shift); + } else { + dev_dbg(codec->dev, "DAI in slave mode\n"); + } + + snd_soc_update_bits(codec, iface_reg, 0x30, iface); + + return 0; +} + +static int wm8776_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_write(codec, WM8776_DACMUTE, !!mute); +} + +static int wm8776_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + + BUG_ON(dai->id >= ARRAY_SIZE(wm8776->sysclk)); + + wm8776->sysclk[dai->id] = freq; + + return 0; +} + +static int wm8776_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Disable the global powerdown; DAPM does the rest */ + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); + } + + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 1); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + + +#define WM8776_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 wm8776_dac_ops = { + .digital_mute = wm8776_mute, + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +static struct snd_soc_dai_ops wm8776_adc_ops = { + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +struct snd_soc_dai wm8776_dai[] = { + { + .name = "WM8776 Playback", + .id = WM8776_DAI_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_dac_ops, + }, + { + .name = "WM8776 Capture", + .id = WM8776_DAI_ADC, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_adc_ops, + }, +}; +EXPORT_SYMBOL_GPL(wm8776_dai); + +#ifdef CONFIG_PM +static int wm8776_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; + + wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8776_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8776_suspend NULL +#define wm8776_resume NULL +#endif + +static int wm8776_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8776_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8776_codec; + codec = wm8776_codec; + + /* 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: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8776_snd_controls, + ARRAY_SIZE(wm8776_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets, + ARRAY_SIZE(wm8776_dapm_widgets)); + snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8776_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8776 = { + .probe = wm8776_probe, + .remove = wm8776_remove, + .suspend = wm8776_suspend, + .resume = wm8776_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8776); + +static int wm8776_register(struct wm8776_priv *wm8776) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8776->codec; + + if (wm8776_codec) { + dev_err(codec->dev, "Another WM8776 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8776; + codec->name = "WM8776"; + codec->owner = THIS_MODULE; + codec->read = wm8776_read_reg_cache; + codec->write = wm8776_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8776_set_bias_level; + codec->dai = wm8776_dai; + codec->num_dai = ARRAY_SIZE(wm8776_dai); + codec->reg_cache_size = WM8776_CACHEREGNUM; + codec->reg_cache = &wm8776->reg_cache; + + memcpy(codec->reg_cache, wm8776_reg, sizeof(wm8776_reg)); + + for (i = 0; i < ARRAY_SIZE(wm8776_dai); i++) + wm8776_dai[i].dev = codec->dev; + + ret = wm8776_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err; + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits; right channel only since we always + * update both. */ + snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100); + + wm8776_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8776); + return ret; +} + +static void wm8776_unregister(struct wm8776_priv *wm8776) +{ + wm8776_set_bias_level(&wm8776->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + snd_soc_unregister_codec(&wm8776->codec); + kfree(wm8776); + wm8776_codec = NULL; +} + +#if defined(CONFIG_SPI_MASTER) +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8776_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8776_priv *wm8776; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8776_spi_write; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8776); + + return wm8776_register(wm8776); +} + +static int __devexit wm8776_spi_remove(struct spi_device *spi) +{ + struct wm8776_priv *wm8776 = dev_get_drvdata(&spi->dev); + + wm8776_unregister(wm8776); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8776_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8776_spi_suspend NULL +#define wm8776_spi_resume NULL +#endif + +static struct spi_driver wm8776_spi_driver = { + .driver = { + .name = "wm8776", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8776_spi_probe, + .suspend = wm8776_spi_suspend, + .resume = wm8776_spi_resume, + .remove = __devexit_p(wm8776_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8776_priv *wm8776; + struct snd_soc_codec *codec; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8776); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8776_register(wm8776); +} + +static __devexit int wm8776_i2c_remove(struct i2c_client *client) +{ + struct wm8776_priv *wm8776 = i2c_get_clientdata(client); + wm8776_unregister(wm8776); + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8776_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8776_i2c_suspend NULL +#define wm8776_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8776_i2c_id[] = { + { "wm8776", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); + +static struct i2c_driver wm8776_i2c_driver = { + .driver = { + .name = "wm8776", + .owner = THIS_MODULE, + }, + .probe = wm8776_i2c_probe, + .remove = __devexit_p(wm8776_i2c_remove), + .suspend = wm8776_i2c_suspend, + .resume = wm8776_i2c_resume, + .id_table = wm8776_i2c_id, +}; +#endif + +static int __init wm8776_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8776_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8776_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 SPI driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8776_modinit); + +static void __exit wm8776_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8776_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8776_spi_driver); +#endif +} +module_exit(wm8776_exit); + +MODULE_DESCRIPTION("ASoC WM8776 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8776.h b/sound/soc/codecs/wm8776.h new file mode 100644 index 0000000..6606d25 --- /dev/null +++ b/sound/soc/codecs/wm8776.h @@ -0,0 +1,51 @@ +/* + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + */ + +#ifndef _WM8776_H +#define _WM8776_H + +/* Registers */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + +#define WM8776_CACHEREGNUM 0x17 + +#define WM8776_DAI_DAC 0 +#define WM8776_DAI_ADC 1 + +extern struct snd_soc_dai wm8776_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_wm8776; + +#endif -- 1.6.3.3

1 0

[alsa-devel] [RFC] [PATCH] ASoC: OMAP: full duplex mode fix
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch tries to correct the problem of full duplex mode not working over a single McBSP based CPU DAI. Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-mcbsp.c.orig 2009-08-01 02:40:45.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-mcbsp.c 2009-08-03 03:28:07.000000000 +0200 @@ -183,6 +183,7 @@ static int omap_mcbsp_dai_trigger(struct struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + u16 buf; int err = 0; switch (cmd) { @@ -191,6 +192,14 @@ static int omap_mcbsp_dai_trigger(struct case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!mcbsp_data->active++) omap_mcbsp_start(mcbsp_data->bus_id); + else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* looks like capture already in progress, + * start playback by taking it out of error condition */ + omap_mcbsp_pollwrite(mcbsp_data->bus_id, 0x0); + else + /* looks like playback already in progress, + * start capture by taking it out of error condition */ + omap_mcbsp_pollread(mcbsp_data->bus_id, &buf); break; case SNDRV_PCM_TRIGGER_STOP:

5 15

[alsa-devel] ASoC: Patches for Freescale MXC SoCs (SSI, DMA and machine).
by javier Martin 06 Aug '09

06 Aug '09

Here comes a series of three patches which currently add audio support for MX1x and MX2x platforms. It has been tested in our custom Visstrim_m10 board based on an i.MX27 processor. All patches are based on stable 2.6.30 kernel and each patch applies over the previous one. [alsa-devel] [PATCH 1/3] ASoC: add DMA platform driver for MX1x and MX2x [alsa-devel] [PATCH 2/3] ASoC: add DAI platform ssi driver for MXC [alsa-devel] [PATCH 3/3] ASoC: add machine driver for i.mx27_visstrim_m10 board -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 3

[alsa-devel] [PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 06 Aug '09

06 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 55 ++++++++++++++++++++++----------------------- 1 files changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..69b794b 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -791,6 +791,22 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +/* + * cs4270_i2c_driver - I2C device identification + * + * This structure tells the I2C subsystem how to identify and support a + * given I2C device type. + */ +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, +}; + #ifdef CONFIG_PM /* This suspend/resume implementation can handle both - a simple standby @@ -802,19 +818,18 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); * and all registers are written back to the hardware when resuming. */ -static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; + struct snd_soc_codec *codec = cs4270_codec; int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; return snd_soc_write(codec, CS4270_PWRCTL, reg); } -static int cs4270_i2c_resume(struct i2c_client *client) +static int cs4270_soc_resume(struct platform_device *pdev) { - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -838,29 +853,11 @@ static int cs4270_i2c_resume(struct i2c_client *client) return snd_soc_write(codec, CS4270_PWRCTL, reg); } #else -#define cs4270_i2c_suspend NULL -#define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* - * cs4270_i2c_driver - I2C device identification - * - * This structure tells the I2C subsystem how to identify and support a - * given I2C device type. - */ -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "cs4270", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, - .remove = cs4270_i2c_remove, - .suspend = cs4270_i2c_suspend, - .resume = cs4270_i2c_resume, -}; - -/* * ASoC codec device structure * * Assign this variable to the codec_dev field of the machine driver's @@ -868,7 +865,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

3 5

[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 06 Aug '09

06 Aug '09

Here's another appl_ptr related patch to the alsa soc mpc5200 ac97 driver. This time the patch is to fix a problem with not correctly handling appl_ptr wrapping back around to the beginning of the buffer. I haven't yet seen a symptom related to this. This came up when I was chatting with Jon Smirl about another bug I was examining. Right now there is another problem that is preventing me from playing really long audio streams. Let me know if this is the wrong approach. - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

4 5

[PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 05 Aug '09

05 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model and let the I2C functions only call the SoC core function for resume and suspend, which do nothing currently but will do later. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 27 +++++++++++++++++++++++---- 1 files changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..ca1e24a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -806,15 +806,30 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + return snd_soc_suspend_device(codec->dev); } static int cs4270_i2c_resume(struct i2c_client *client) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + + return snd_soc_resume_device(codec->dev); +} + +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) +{ + struct snd_soc_codec *codec = cs4270_codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -840,6 +855,8 @@ static int cs4270_i2c_resume(struct i2c_client *client) #else #define cs4270_i2c_suspend NULL #define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* @@ -868,7 +885,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

1 0

[alsa-devel] [PATCH - amixer] listen to D-Bus for volume-up, volume-down and mute signals.
by Łukasz Stelmach 05 Aug '09

05 Aug '09

The patch introduces D-Bus support in amixer. With --dbus option given amixer works continiously (like with the --stdin) waiting for org.freedesktop.Hal.Device.Condition with the first argument "ButtonPressed" and the second being one of: "volume-up", "volume-down" and "mute". These are emmited by the HAL input helper. A single amixer process can controll one mixer item. There are options to choose a specific item to controll, an input device to listen and a single adjustment step size. Signed-off-by: Łukasz Stelmach <stlman(a)poczta.fm> --- amixer/Makefile.am | 3 +- amixer/amixer.1 | 18 ++++++ amixer/amixer.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.in | 12 ++++ 4 files changed, 203 insertions(+), 1 deletions(-) diff --git a/amixer/Makefile.am b/amixer/Makefile.am index fcd0e81..fa951a6 100644 --- a/amixer/Makefile.am +++ b/amixer/Makefile.am @@ -1,5 +1,6 @@ INCLUDES = -I$(top_srcdir)/include -LDADD = -lm +LDADD = -lm @DBUS_LIBS@ +CFLAGS += @DBUS_CFLAGS@ # LDFLAGS = -static # CFLAGS += -g -Wall diff --git a/amixer/amixer.1 b/amixer/amixer.1 index b1ac323..bc160be 100644 --- a/amixer/amixer.1 +++ b/amixer/amixer.1 @@ -88,6 +88,24 @@ Select the card number to control. The device name created from this parameter has syntax 'hw:N' where N is specified card number. .TP +\fI\-\-dbus[=\fISCONTROL\fP] + +Connect to system D-Bus and wait for \fIorg.freedesktop.Hal.Device.Condition\fP +signals with ButtonPressed as the first argument and one of: \fIvolume-up\fP, +\fIvolume-down\fP or \fImute\fP as the second one. Control the selected +\fISCONTROL\fP item. The default item is 'Master,0'. + +.TP +\fI\-\-dbus-path=<\fIUDI\fP> + +Choose a particular input device that emits control events. + +.TP +\fI\-\-dbus-step=<\fIN\fP> + +Set an amount by wich to increase or decrease volume upon event. + +.TP \fI\-D\fP device Select the device name to control. The default control name is 'default'. diff --git a/amixer/amixer.c b/amixer/amixer.c index 9620721..31de674 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -18,6 +18,8 @@ * */ +#include "aconfig.h" + #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -31,6 +33,10 @@ #include <sys/poll.h> #include "amixer.h" +#ifdef HAVE_DBUS +#include <dbus/dbus.h> +#endif /* HAVE_DBUS */ + #define LEVEL_BASIC (1<<0) #define LEVEL_INACTIVE (1<<1) #define LEVEL_ID (1<<2) @@ -43,6 +49,12 @@ static int ignore_error = 0; static struct snd_mixer_selem_regopt smixer_options; static char card[64] = "default"; +#ifdef HAVE_DBUS +static char dbus_sctl[64] = "Master,0"; +static char *dbus_path = NULL; +static char dbus_step[8] = "1"; +#endif /* HAVE_DBUS */ + static void error(const char *fmt,...) { va_list va; @@ -60,6 +72,11 @@ static int help(void) printf("\nAvailable options:\n"); printf(" -h,--help this help\n"); printf(" -c,--card N select the card\n"); +#ifdef HAVE_DBUS + printf(" --dbus[=sID] Connect to the system D-Bus and wait for volume messages\n"); + printf(" --dbus-path UDI Choose a particular input device (udi)\n"); + printf(" --dbus-step N Set the step size (default: 1)\n"); +#endif /* HAVE_DBUS */ printf(" -D,--device N select the device, default '%s'\n", card); printf(" -d,--debug debug mode\n"); printf(" -n,--nocheck do not perform range checking\n"); @@ -1887,15 +1904,147 @@ static int exec_stdin(void) return 0; } +#ifdef HAVE_DBUS + +#define OPT_DBUS 0x81 +#define OPT_DBUS_PATH 0x82 +#define OPT_DBUS_STEP 0x83 + +void* get_dbus_arg(DBusMessageIter* args, int type, void** out) { + if (type != dbus_message_iter_get_arg_type(args)) + return NULL; + dbus_message_iter_get_basic(args, out); + dbus_message_iter_next(args); + return *out; +} + +int exec_dbus(void) +{ + char *sargv[MAX_ARGS]; + int sargc; + const char* s; + int stepl; + DBusError error; + DBusMessage* msg; + DBusConnection *conn; + DBusMessageIter args; + + stepl = strlen(dbus_step); + dbus_step[stepl+1] = '\0'; + + dbus_error_init (&error); + conn = dbus_bus_get (DBUS_BUS_SYSTEM, &error); + + if(!conn) { + fprintf(stderr, "dbus:%s: %s\n", + error.name, error.message); + return 1; + } + + dbus_bus_add_match(conn, + "type='signal'," + "interface='org.freedesktop.Hal.Device'," + "member='Condition'", + &error); + /* The first message is a NameOwnerChanged signal, ignore it. */ + msg = dbus_connection_pop_message(conn); + dbus_message_unref(msg); + + if(dbus_error_is_set(&error)) { + fprintf(stderr, "dbus:match error:%s\n", + error.message); + return 1; + } + + while (1) { + sargv[0] = dbus_sctl; sargc = 1; + + if (!(msg = dbus_connection_pop_message(conn))) { + dbus_connection_read_write(conn, -1); + continue; + } + + s = dbus_message_get_interface(msg); + if(strncmp("org.freedesktop.Hal.Device", s, 26)) { + if(debugflag) + fprintf(stderr, "dbus:unsupported interface:%s\n", s); + goto badmessage; + } + + if(dbus_path && + (s=dbus_message_get_path(msg)) && + strncmp(dbus_path, s, strlen(s))) { + if(debugflag) + fprintf(stderr, "dbus:not the selected udi:%s\n", s); + goto badmessage; + } + + if(!dbus_message_iter_init(msg, &args)) { + if(debugflag) + fprintf(stderr, "dbus:message has no arguments\n"); + goto badmessage; + } + + if (strncmp(get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s), "ButtonPressed", 13)) { + if(debugflag) + fprintf(stderr, "dbus:not a ButtonPressed event\n"); + goto badmessage; + } + + if (!get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s)) { + if(debugflag) + fprintf(stderr, "dbus:invalid argument type\n"); + goto badmessage; + } + + /* static int sset(unsigned int argc, char *argv[], int roflag, int keep_handle) */ + if (!strncmp(s, "volume-up", 9)) { + if(debugflag) + fprintf(stderr, "dbus:volume-up\n"); + dbus_step[stepl]='+'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "volume-down", 11)) { + if(debugflag) + fprintf(stderr, "dbus:volume-down\n"); + dbus_step[stepl]='-'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "mute", 4)) { + if(debugflag) + fprintf(stderr, "dbus:mute (toggle)\n"); + sargv[1] = "toggle"; + } else { + if(debugflag) + fprintf(stderr, "dbus:invalid argument value\n"); + goto badmessage; + } + sset(++sargc, sargv, 0, 1); +badmessage: + dbus_message_unref(msg); + } + + if(conn) + dbus_connection_unref(conn); + return 0; +} +#endif /* HAVE_DBUS */ int main(int argc, char *argv[]) { int morehelp, level = 0; int read_stdin = 0; +#ifdef HAVE_DBUS + int read_dbus = 0; + int t; +#endif /* HAVE_DBUS */ static const struct option long_option[] = { {"help", 0, NULL, 'h'}, {"card", 1, NULL, 'c'}, +#ifdef HAVE_DBUS + {"dbus", optional_argument, NULL, OPT_DBUS}, + {"dbus-path", 1, NULL, OPT_DBUS_PATH}, + {"dbus-step", 1, NULL, OPT_DBUS_STEP}, +#endif /* HAVE_DBUS */ {"device", 1, NULL, 'D'}, {"quiet", 0, NULL, 'q'}, {"inactive", 0, NULL, 'i'}, @@ -1964,6 +2113,24 @@ int main(int argc, char *argv[]) case 's': read_stdin = 1; break; +#ifdef HAVE_DBUS + case OPT_DBUS: + if(optarg) { + strncpy(dbus_sctl, optarg, sizeof(dbus_sctl)-1); + dbus_sctl[sizeof(dbus_sctl)-1] = '\0'; + } + read_dbus = 1; + break; + case OPT_DBUS_PATH: + dbus_path=optarg; + break; + case OPT_DBUS_STEP: + t=atoi(optarg); + if(t > 0 && t <= 999999) { + snprintf(dbus_step, sizeof(dbus_step), "%d", atoi(optarg)); + } + break; +#endif /* HAVE_DBUS */ default: fprintf(stderr, "Invalid switch or option needs an argument.\n"); morehelp++; @@ -1978,6 +2145,10 @@ int main(int argc, char *argv[]) if (read_stdin) return exec_stdin(); +#ifdef HAVE_DBUS + if(read_dbus) + return exec_dbus(); +#endif /* HAVE_DBUS */ if (argc - optind <= 0) { return selems(LEVEL_BASIC | level) ? 1 : 0; } diff --git a/configure.in b/configure.in index 1349ff3..717d6c5 100644 --- a/configure.in +++ b/configure.in @@ -7,6 +7,9 @@ AM_INIT_AUTOMAKE(alsa-utils, 1.0.20) AM_GNU_GETTEXT([external]) AM_GNU_GETTEXT_VERSION([0.15]) +dnl required versions of other packages +m4_define([dbus_required_version], [1.2.0]) + dnl Checks for programs. dnl try to gues cross-compiler if not set @@ -74,6 +77,15 @@ AC_ARG_ENABLE(alsamixer, no) alsamixer=false ;; *) AC_MSG_ERROR(bad value ${enableval} for --enable-alsamixer) ;; esac],[alsamixer=true]) +AC_ARG_WITH(dbus, [ --with-dbus Enable D-Bus support in amixer]) + if test "x$with_dbus" != xno; then + PKG_CHECK_MODULES(DBUS, dbus-1 >= dbus_required_version, + have_dbus=yes, + have_dbus=no) + fi + if test "x$have_dbus" = xyes; then + AC_DEFINE(HAVE_DBUS, 1, [Define to 1 if amixer is to support D-Bus]) + fi fi AM_CONDITIONAL(ALSAMIXER, test x$alsamixer = xtrue) -- 1.6.3.3 ---------------------------------------------------------------------- Zostan Dziewczyna lub Chlopakiem lata! 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7 13

[alsa-devel] Question to sophisticated alsa developer
by Andreas Bießmann 05 Aug '09

05 Aug '09

Hi all, I currently test alsa playback on AVR32 board with atmel's abdac driver. My environment is 2.6.30.4 customized kernel (customization for my own board), rootfs and toolchain made of customized buildroot from atmel. I use alsa-lib 1.0.20 with patch [1] applied and alsa-utils with patch [2] applied. The alsa code does compile and work fine (aside from my problem described some lines later). My test case is a sine wave file with 440Hz tone, samplerate is 17089 (cause of samplerate abdac driver offers and to avoid resampling in alsa-lib) playing with aplay. This works fine, I see a 440Hz sine wave on my oscilloscope and there is no effect as described later on. My problem is when I'm enabling dmix layer to run concurrent aplay processes I always get a click at the end of my last played sine wave file. My Oscilloscope shows me that there is sometimes a short part of a single or somtimes multiple sine period(s) moved down (or up) to mid value (I can supply a screenshot of oscilloscope, if this description is not clear enough). Mostly this effect is about 500 us long. The sine wave is missing the information at this position, it is no delay of the analog value at this position. If I overlay a correct sine wave the following part after the problematic position matches exactly the overlayed sine wave. I investigated that this problem always occur somewhere at the end of dmix. When I play a long "null" sample and start repeated play of my short sine wave sample the effect is not there while the null sample is playing. After ending of null sample the described effect appears instantly on each end of my repeated test sample. Here is my asound.conf to enable dmix layer for alsa: ---8<--- pcm.!default { type plug slave.pcm "dmixer" } ctl.mixer0 { type hw card 0 } pcm.!plughw { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0" format S16_BE rate 17089 } bindings { 0 0 1 1 } slowptr false } ctl.dmixer { type hw card 0 } --->8--- Maybe there is already a bloomer in. I have tried to change slowptr stuff in pcm.dmixer without any effect to my problem. I also tried to change buffer sizes in dmixer's slave but these changes had no effect. I also tried some changes in kernel driver for ABDAC. I played a bit with periods_min and period_bytes_min without effect to my problem. The buffer_size in "Direct Stream Mixing PCM" is fixed to the value in "hw:0". But if I double the periods_min in Kernel (default: 6 periods_min, 64 periods_max) for "hw:0" the maximum value for buffer_size in "Direct Stream Mixing PCM" is fixed to 8192. The "Route conversion" layer has always the buffer_size value of the next layer. This is 8192 for doubled periods_min in kernel and dmix enabled and 12288 for doubled periods_min and dmix disabled. Any advice to this "feature"? Here is the output of aplay -v for dmix enabled: ---8<--- # aplay -v sin_440_17089_1sec.wav Playing WAVE 'sin_440_17089_1sec.wav' : Signed 16 bit Little Endian, Rate 17089 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_BE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'Atmel ABDAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1610612736 silence_threshold: 0 silence_size : 1610612736 boundary : 1610612736 appl_ptr : 0 hw_ptr : 0 --->8--- And here the output of aplay -v without dmix: ---8<--- # aplay -v sin_440_17089_1sec.wav Playing WAVE 'sin_440_17089_1sec.wav' : Signed 16 bit Little Endian, Rate 17089 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_BE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 12288 stop_threshold : 12288 silence_threshold: 0 silence_size : 0 boundary : 1610612736 Slave: Hardware PCM card 0 'Atmel ABDAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 12288 stop_threshold : 12288 silence_threshold: 0 silence_size : 0 boundary : 1610612736 appl_ptr : 0 hw_ptr : 0 --->8--- Well, currently I have no more ideas where to dig for my problem. I do need a mixing layer to allow two or three concurrent aplay instances at the same time but the output in this case is (atm) worse. Has anyone some advice where to search for a solution to my problem. Or has anybody some advice to use another feature of alsa to allow concurrent aplay processes? regards Andreas Bießmann [1] http://git.buildroot.net/buildroot/tree/package/multimedia/alsa-lib/alsa-li… [2] http://git.buildroot.net/buildroot/tree/package/multimedia/alsa-utils/alsa-…

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[alsa-devel] pyalsa: synchronizing queue with MIDI clock events
by Christopher Arndt 05 Aug '09

05 Aug '09

Hi everybody, this is my first post on this list, so please excuse me, should it not be the right place to post this. I started exploring programming the ALSA sequencer via the pyalsa Python interface, i.e the alsaseq module. I wrote a small utility that reads MIDI events from one port, filters/processes them and writes it to another. This allows for routing by channel, note range, controller number etc. and to filter out, replace or add MIDI events. Now to synchronize the queue tempo to incoming MIDI clock events, to be able implement synchronized delays or arpeggiators, I use the method in the code shown below. This works ok, but I notice that the measured BPM oscillates +-1 BPM around the value displayed by the clock source (my synth's sequencer) if I measure/average only a few (~10) ticks or do not round the result to an integer value. Is this the right approach? What is a sensible number of ticks to take into account for measurement? Should I use another reference timer than Python's time.time() function? Is pyalsa generally suitable for this kind of application even on older/weaker hardware (e.g. a NSLU2)? Any comments or suggestions for improvement would be very much appreciated! Chris class MidiProcessor(object): def run(self): # This is simplified to only show the general logic while True: events = self.sequencer.receive_events( timeout=RECEIVE_TIMEOUT, maxevents=1) for event in events: if event.type == SEQ_EVENT_CLOCK: self.sync_queue(event) else: # do other MIDI processing / routing def sync_queue(self, event): """Sync queue tempo to incoming MIDI clock events.""" # list to collect the timestamps of the last few ticks lt = self._last_ticks lt.append(time.time()) ltlen = len(lt) if ltlen > 1: # calculate & set bpm: calculate difference between # the times the last few ticks were received and average # all results avg_delta = sum( [y-x for x,y in zip(lt, lt[1:])]) / (ltlen-1) # tick length is a 24th of a quarter note bpm = round(60 / avg_delta / 24) if bpm != self.bpm: self.bpm = bpm self.sequencer.queue_tempo(self.queue, tempo=int(6e7 / self.bpm), ppq=self.ppq) # only remember last 24 received ticks # (length of a quarter note) if ltlen > 24: lt.pop(0)

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[alsa-devel] [PATCH 3/3] ASoC: add machine driver for i.mx27_visstrim_m10 board
by javier Martin 05 Aug '09

05 Aug '09

This adds support for i.mx27_visstrim_sm10 board machine driver which uses an i.mx27 processor plus a wm8974 codec. It has been tested on a visstrim_sm10 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 8 + sound/soc/imx/Makefile | 4 + sound/soc/imx/mx27vis_wm8974.c | 317 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 329 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 886dadd..2c6f568 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -9,5 +9,13 @@ config SND_MX1_MX2_SOC config SND_MXC_SOC_SSI tristate +config SND_SOC_MX27VIS_WM8974 + tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" + depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 + select SND_MXC_SOC_SSI + select SND_SOC_WM8974 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with WM8974. diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 6552cb2..c2ffd2c 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -4,3 +4,7 @@ snd-soc-mxc-ssi-objs := mxc-ssi.o obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o + +# i.MX Machine Support +snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o +obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c new file mode 100644 index 0000000..e4dcb53 --- /dev/null +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -0,0 +1,317 @@ +/* + * mx27vis_wm8974.c -- SoC audio for mx27vis + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin(a)vista-silicon.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/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + + +#include "../codecs/wm8974.h" +#include "mx1_mx2-pcm.h" +#include "mxc-ssi.h" +#include <mach/gpio.h> +#include <mach/iomux.h> + +#define IGNORED_ARG 0 + + +static struct snd_soc_card mx27vis; + +/** + * This function connects SSI1 (HPCR1) as slave to + * SSI1 external signals (PPCR1) + * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from + * port 4 + */ +void audmux_connect_1_4(void) +{ + pr_debug("AUDMUX: normal operation mode\n"); + /* Reset HPCR1 and PPCR1 */ + + DAM_HPCR1 = 0x00000000; + DAM_PPCR1 = 0x00000000; + + /* set to synchronous */ + DAM_HPCR1 |= AUDMUX_HPCR_SYN; + DAM_PPCR1 |= AUDMUX_PPCR_SYN; + + + /* set Rx sources 1 <--> 4 */ + DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */ + DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */ + + /* set Tx frame and Clock direction and source 4 --> 1 output */ + DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR; + DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */ + + return; +} + +static int mx27vis_hifi_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 *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0; + int ret = 0; + + /* + * The WM8974 is better at generating accurate audio clocks than the + * MX27 SSI controller, so we will use it as master when we can. + */ + switch (params_rate(params)) { + case 8000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + mclk = WM8974_MCLKDIV_12; + pll_out = 24576000; + break; + case 16000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + pll_out = 12288000; + break; + case 48000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + pll_out = 12288000; + break; + case 96000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 12288000; + break; + case 11025: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_16; + pll_out = 11289600; + break; + case 22050: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_8; + pll_out = 11289600; + break; + case 44100: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + mclk = WM8974_MCLKDIV_2; + pll_out = 11289600; + break; + case 88200: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = codec_dai->ops->set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from codec DAI configuration\n"); + return ret; + } + + /* set cpu DAI configuration */ + ret = cpu_dai->ops->set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from cpu DAI configuration\n"); + return ret; + } + + /* Put DC field of STCCR to 1 (not zero) */ + ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2); + + /* set the SSI system clock as input */ + ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Error when setting system SSI clk\n"); + return ret; + } + + /* set codec BCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting BCLK division\n"); + return ret; + } + + + /* codec PLL input is 25 MHz */ + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + 25000000, pll_out); + if (ret < 0) { + printk(KERN_ERR "Error when setting PLL input\n"); + return ret; + } + + /*set codec MCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting MCLK division\n"); + return ret; + } + + return 0; +} + +static int mx27vis_hifi_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 codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); +} + +/* + * mx27vis WM8974 HiFi DAI opserations. + */ +static struct snd_soc_ops mx27vis_hifi_ops = { + .hw_params = mx27vis_hifi_hw_params, + .hw_free = mx27vis_hifi_hw_free, +}; + + +static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state) +{ + return 0; +} + +static int mx27vis_resume(struct platform_device *pdev) +{ + return 0; +} + +static int mx27vis_probe(struct platform_device *pdev) +{ + int ret = 0; + + ret = get_ssi_clk(0, &pdev->dev); + + if (ret < 0) { + printk(KERN_ERR "%s: cant get ssi clock\n", __func__); + return ret; + } + + + return 0; +} + +static int mx27vis_remove(struct platform_device *pdev) +{ + put_ssi_clk(0); + return 0; +} + +static struct snd_soc_dai_link mx27vis_dai[] = { +{ /* Hifi Playback*/ + .name = "WM8974", + .stream_name = "WM8974 HiFi", + .cpu_dai = &imx_ssi_pcm_dai[0], + .codec_dai = &wm8974_dai, + .ops = &mx27vis_hifi_ops, +}, +}; + +static struct snd_soc_card mx27vis = { + .name = "mx27vis", + .platform = &mx1_mx2_soc_platform, + .probe = mx27vis_probe, + .remove = mx27vis_remove, + .suspend_pre = mx27vis_suspend, + .resume_post = mx27vis_resume, + .dai_link = mx27vis_dai, + .num_links = ARRAY_SIZE(mx27vis_dai), +}; + +static struct snd_soc_device mx27vis_snd_devdata = { + .card = &mx27vis, + .codec_dev = &soc_codec_dev_wm8974, +}; + +static struct platform_device *mx27vis_snd_device; + +/* Temporal definition of board specific behaviour */ +void gpio_ssi_active(int ssi_num) +{ + int ret = 0; + + unsigned int ssi1_pins[] = { + PC20_PF_SSI1_FS, + PC21_PF_SSI1_RXD, + PC22_PF_SSI1_TXD, + PC23_PF_SSI1_CLK, + }; + unsigned int ssi2_pins[] = { + PC24_PF_SSI2_FS, + PC25_PF_SSI2_RXD, + PC26_PF_SSI2_TXD, + PC27_PF_SSI2_CLK, + }; + if (ssi_num == 0) + ret = mxc_gpio_setup_multiple_pins(ssi1_pins, + ARRAY_SIZE(ssi1_pins), "USB OTG"); + else + ret = mxc_gpio_setup_multiple_pins(ssi2_pins, + ARRAY_SIZE(ssi2_pins), "USB OTG"); + if (ret) + printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num); +} + + +static int __init mx27vis_init(void) +{ + int ret; + + mx27vis_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata); + mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev; + ret = platform_device_add(mx27vis_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_snd_device); + } + + /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */ + gpio_ssi_active(0); + audmux_connect_1_4(); + + return ret; +} + +static void __exit mx27vis_exit(void) +{ + /* We should call some "ssi_gpio_inactive()" properly */ +} + +module_init(mx27vis_init); +module_exit(mx27vis_exit); + + +MODULE_AUTHOR("Javier Martin, javier.martin(a)vista-silicon.com"); +MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis"); +MODULE_LICENSE("GPL"); --- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[alsa-devel] [PATCH 1/6] pulse: replace manual mainloop by pa_mainloop_iterate()
by Lennart Poettering 05 Aug '09

05 Aug '09

The pa_mainloop_prepare()/_poll()/_dispatch() can be simplified by simply calling pa_mainloop_iterate() which does all this in one call. --- pulse/conf_pulse.c | 10 +--------- 1 files changed, 1 insertions(+), 9 deletions(-) diff --git a/pulse/conf_pulse.c b/pulse/conf_pulse.c index 3f9e071..35dd383 100644 --- a/pulse/conf_pulse.c +++ b/pulse/conf_pulse.c @@ -61,15 +61,7 @@ conf_pulse_hook_load_if_running(snd_config_t * root, snd_config_t * config, goto out; do { - err = pa_mainloop_prepare(loop, -1); - if (err < 0) - goto out; - - err = pa_mainloop_poll(loop); - if (err < 0) - goto out; - - err = pa_mainloop_dispatch(loop); + err = pa_mainloop_iterate(loop, 1, NULL); if (err < 0) goto out; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 6/6] pulse: immediately trigger EIO when connection is dropped
by Lennart Poettering 05 Aug '09

05 Aug '09

When the connection is dropped notify the application immediately instead of waiting until the applications calls into us the next time. This makes "aplay" handle connections shutdown similar to hardware unplugs: an immediate EIO is thrown. --- pulse/pulse.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index 9c05171..a94caaf 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -77,10 +77,17 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) static void context_state_cb(pa_context * c, void *userdata) { + pa_context_state_t state; snd_pulse_t *p = userdata; assert(c); - switch (pa_context_get_state(c)) { + state = pa_context_get_state(c); + + /* When we get disconnected, tell the process */x + if (!PA_CONTEXT_IS_GOOD(state)) + pulse_poll_activate(p); + + switch (state) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: case PA_CONTEXT_FAILED: -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 5/6] pulse: rework object destruction paths a bit
by Lennart Poettering 05 Aug '09

05 Aug '09

Make sure we deal better with partially initialized structs. Don't check for pointer state before calling free() since free() does that anyway. --- pulse/ctl_pulse.c | 7 ++----- pulse/pcm_pulse.c | 19 ++++++++++--------- 2 files changed, 12 insertions(+), 14 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 879f260..85863ed 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -606,11 +606,8 @@ static void pulse_close(snd_ctl_ext_t * ext) if (ctl->p) pulse_free(ctl->p); - if (ctl->source) - free(ctl->source); - if (ctl->sink) - free(ctl->sink); - + free(ctl->source); + free(ctl->sink); free(ctl); } diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 98983f8..02a837e 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -819,21 +819,22 @@ static int pulse_close(snd_pcm_ioplug_t * io) assert(pcm); - pa_threaded_mainloop_lock(pcm->p->mainloop); + if (pcm->p && pcm->p->mainloop) { - if (pcm->stream) { - pa_stream_disconnect(pcm->stream); - pa_stream_unref(pcm->stream); - } + pa_threaded_mainloop_lock(pcm->p->mainloop); - pa_threaded_mainloop_unlock(pcm->p->mainloop); + if (pcm->stream) { + pa_stream_disconnect(pcm->stream); + pa_stream_unref(pcm->stream); + } + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + } if (pcm->p) pulse_free(pcm->p); - if (pcm->device) - free(pcm->device); - + free(pcm->device); free(pcm); return 0; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 4/6] pulse: unify stream/context state checks
by Lennart Poettering 05 Aug '09

05 Aug '09

Unify (and simplify) the paths that check for the validity of a stream/context: always call into check_stream()/pulse_check_connection() when applicable instead of rolling our own checks each time. Also check for validity of mainloop before locking it. --- pulse/ctl_pulse.c | 72 ++++++++++++++++++++++++++++++++---------- pulse/pcm_pulse.c | 90 +++++++++++++++++++++++----------------------------- 2 files changed, 95 insertions(+), 67 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 2caa29b..879f260 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -150,9 +150,13 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p) return -EBADFD; + err = pulse_check_connection(ctl->p); + if (err < 0) + return err; + o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); if (o) { @@ -182,17 +186,27 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) static int pulse_elem_count(snd_ctl_ext_t * ext) { snd_ctl_pulse_t *ctl = ext->private_data; - int count = 0; + int count = 0, err; assert(ctl); + if (!ctl->p || !ctl->p->mainloop) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) { + count = err; + goto finish; + } + if (ctl->source) count += 2; if (ctl->sink) count += 2; +finish: pa_threaded_mainloop_unlock(ctl->p->mainloop); return count; @@ -202,16 +216,21 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, snd_ctl_elem_id_t * id) { snd_ctl_pulse_t *ctl = ext->private_data; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + if (ctl->source) { if (offset == 0) snd_ctl_elem_id_set_name(id, SOURCE_VOL_NAME); @@ -220,14 +239,19 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, } else offset += 2; + err = 0; + +finish: pa_threaded_mainloop_unlock(ctl->p->mainloop); - if (offset == 2) - snd_ctl_elem_id_set_name(id, SINK_VOL_NAME); - else if (offset == 3) - snd_ctl_elem_id_set_name(id, SINK_MUTE_NAME); + if (err >= 0) { + if (offset == 2) + snd_ctl_elem_id_set_name(id, SINK_VOL_NAME); + else if (offset == 3) + snd_ctl_elem_id_set_name(id, SINK_MUTE_NAME); + } - return 0; + return err; } static snd_ctl_ext_key_t pulse_find_elem(snd_ctl_ext_t * ext, @@ -266,7 +290,7 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -319,7 +343,7 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -371,7 +395,7 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -476,7 +500,7 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -491,17 +515,23 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, { snd_ctl_pulse_t *ctl = ext->private_data; int offset; - int err = -EAGAIN; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); - if (!ctl->updated || !ctl->subscribed) + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + + if (!ctl->updated || !ctl->subscribed) { + err = -EAGAIN; goto finish; + } if (ctl->source) offset = 2; @@ -540,20 +570,28 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, unsigned short *revents) { snd_ctl_pulse_t *ctl = ext->private_data; - int err = 0; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + if (ctl->updated) *revents = POLLIN; else *revents = 0; + err = 0; + +finish: + pa_threaded_mainloop_unlock(ctl->p->mainloop); return err; diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index a625f55..98983f8 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -135,11 +135,15 @@ static int update_active(snd_pcm_pulse_t *pcm) { if (!pcm->p) return -EBADFD; - ret = check_active(pcm); + ret = check_stream(pcm); if (ret < 0) - return ret; + goto finish; + + ret = check_active(pcm); - if (ret > 0) +finish: + + if (ret != 0) /* On error signal the caller, too */ pulse_poll_activate(pcm->p); else pulse_poll_deactivate(pcm->p); @@ -199,12 +203,12 @@ static int pulse_start(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -244,12 +248,12 @@ static int pulse_stop(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -291,12 +295,12 @@ static int pulse_drain(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -328,7 +332,7 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) @@ -339,12 +343,7 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -379,18 +378,13 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - err = -EBADFD; - goto finish; - } - for (;;) { - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -433,17 +427,12 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -493,17 +482,12 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -624,13 +608,16 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = check_active(pcm); + err = check_stream(pcm); + if (err < 0) + goto finish; + err = check_active(pcm); if (err < 0) goto finish; @@ -655,7 +642,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -757,7 +744,7 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -856,23 +843,26 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) { snd_pcm_pulse_t *pcm = io->private_data; int err = 0; + pa_operation *o; assert (pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (pcm->stream) { - pa_operation *o; - o = pa_stream_cork(pcm->stream, enable, NULL, NULL); - if (o) - pa_operation_unref(o); - else - err = -EIO; - } + err = check_stream(pcm); + if (err < 0) + goto finish; + + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); + if (o) + pa_operation_unref(o); + else + err = -EIO; +finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); return err; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 3/6] pulse: get rid of redundant state variable
by Lennart Poettering 05 Aug '09

05 Aug '09

snd_pulse_t::state was mostly shadowing the state of pa_context_get_state(snd_pulse_t::context), so get rid of it and use the state of the context directly. --- pulse/pulse.c | 14 +++----------- pulse/pulse.h | 6 ------ 2 files changed, 3 insertions(+), 17 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index c313182..9c05171 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -59,12 +59,6 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) assert(p); assert(o); - if (p->state != PULSE_STATE_READY) - return -EBADFD; - - if (!p->mainloop) - return -EBADFD; - for (;;) { int err; @@ -124,8 +118,6 @@ snd_pulse_t *pulse_new(void) if (!p) return NULL; - p->state = PULSE_STATE_INIT; - if (pipe(fd)) { free(p); return NULL; @@ -192,13 +184,15 @@ void pulse_free(snd_pulse_t * p) int pulse_connect(snd_pulse_t * p, const char *server) { int err; + pa_context_state_t state; assert(p); if (!p->context || !p->mainloop) return -EBADFD; - if (p->state != PULSE_STATE_INIT) + state = pa_context_get_state(p->context); + if (state != PA_CONTEXT_UNCONNECTED) return -EBADFD; pa_threaded_mainloop_lock(p->mainloop); @@ -221,8 +215,6 @@ int pulse_connect(snd_pulse_t * p, const char *server) pa_threaded_mainloop_unlock(p->mainloop); - p->state = PULSE_STATE_READY; - return 0; error: diff --git a/pulse/pulse.h b/pulse/pulse.h index 51f9a11..e98124f 100644 --- a/pulse/pulse.h +++ b/pulse/pulse.h @@ -31,12 +31,6 @@ typedef struct snd_pulse { pa_context *context; int thread_fd, main_fd; - - enum { - PULSE_STATE_INIT, - PULSE_STATE_READY, - } state; - } snd_pulse_t; int pulse_check_connection(snd_pulse_t * p); -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 2/6] pulse: move a couple of PCM related functions from pulse.c to pcm_pulse.c
by Lennart Poettering 05 Aug '09

05 Aug '09

A number of functions in pulse.c are only relevant for the PCM handling, so let's move them to pcm_pulse.c. This allows us to simplify their argument lists a bit. --- pulse/pcm_pulse.c | 118 ++++++++++++++++++++++++++++++++++++++++++---------- pulse/pulse.c | 55 ------------------------- pulse/pulse.h | 4 -- 3 files changed, 95 insertions(+), 82 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 24347f9..a625f55 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -49,6 +49,32 @@ typedef struct snd_pcm_pulse { pa_buffer_attr buffer_attr; } snd_pcm_pulse_t; +static int check_stream(snd_pcm_pulse_t *pcm) +{ + int err; + pa_stream_state_t state; + + assert(pcm); + + if (!pcm->p) + return -EBADFD; + + err = pulse_check_connection(pcm->p); + if (err < 0) + return err; + + if (!pcm->stream) + return -EBADFD; + + state = pa_stream_get_state(pcm->stream); + if (!PA_STREAM_IS_GOOD(state)) + return -EIO; + + err = 0; + + return err; +} + static int update_ptr(snd_pcm_pulse_t *pcm) { size_t size; @@ -118,9 +144,53 @@ static int update_active(snd_pcm_pulse_t *pcm) { else pulse_poll_deactivate(pcm->p); + return ret; +} + +static int wait_stream_state(snd_pcm_pulse_t *pcm, pa_stream_state_t target) +{ + pa_stream_state_t state; + + assert(pcm); + + if (!pcm->p) + return -EBADFD; + + for (;;) { + int err; + + err = pulse_check_connection(pcm->p); + if (err < 0) + return err; + + if (!pcm->stream) + return -EBADFD; + + state = pa_stream_get_state(pcm->stream); + if (state == target) + break; + + if (!PA_STREAM_IS_GOOD(state)) + return -EIO; + + pa_threaded_mainloop_wait(pcm->p->mainloop); + } + return 0; } +static void stream_success_cb(pa_stream * p, int success, void *userdata) +{ + snd_pcm_pulse_t *pcm = userdata; + + assert(pcm); + + if (!pcm->p) + return; + + pa_threaded_mainloop_signal(pcm->p->mainloop, 0); +} + static int pulse_start(snd_pcm_ioplug_t * io) { snd_pcm_pulse_t *pcm = io->private_data; @@ -138,18 +208,13 @@ static int pulse_start(snd_pcm_ioplug_t * io) if (err < 0) goto finish; - if (pcm->stream == NULL) - goto finish; - - o = pa_stream_cork(pcm->stream, 0, pulse_stream_success_cb, - pcm->p); + o = pa_stream_cork(pcm->stream, 0, stream_success_cb, pcm); if (!o) { err = -EIO; goto finish; } - u = pa_stream_trigger(pcm->stream, pulse_stream_success_cb, - pcm->p); + u = pa_stream_trigger(pcm->stream, stream_success_cb, pcm); pcm->underrun = 0; err_o = pulse_wait_operation(pcm->p, o); @@ -165,7 +230,6 @@ static int pulse_start(snd_pcm_ioplug_t * io) goto finish; } - finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); @@ -189,18 +253,13 @@ static int pulse_stop(snd_pcm_ioplug_t * io) if (err < 0) goto finish; - if (pcm->stream == NULL) - goto finish; - - o = pa_stream_cork(pcm->stream, 1, pulse_stream_success_cb, - pcm->p); + o = pa_stream_cork(pcm->stream, 1, stream_success_cb, pcm); if (!o) { err = -EIO; goto finish; } - u = pa_stream_flush(pcm->stream, pulse_stream_success_cb, - pcm->p); + u = pa_stream_flush(pcm->stream, stream_success_cb, pcm); if (!u) { pa_operation_unref(o); err = -EIO; @@ -241,7 +300,7 @@ static int pulse_drain(snd_pcm_ioplug_t * io) if (err < 0) goto finish; - o = pa_stream_drain(pcm->stream, pulse_stream_success_cb, pcm->p); + o = pa_stream_drain(pcm->stream, stream_success_cb, pcm); if (!o) { err = -EIO; goto finish; @@ -504,6 +563,23 @@ finish: return ret; } +static void stream_state_cb(pa_stream * p, void *userdata) +{ + snd_pcm_pulse_t *pcm = userdata; + pa_stream_state_t state; + + assert(pcm); + + if (!pcm->p) + return; + + state = pa_stream_get_state(p); + if (!PA_STREAM_IS_GOOD(state)) + pulse_poll_activate(pcm->p); + + pa_threaded_mainloop_signal(pcm->p->mainloop, 0); +} + static void stream_request_cb(pa_stream * p, size_t length, void *userdata) { snd_pcm_pulse_t *pcm = userdata; @@ -586,8 +662,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) if (pcm->stream) { pa_stream_disconnect(pcm->stream); - pulse_wait_stream_state(pcm->p, pcm->stream, - PA_STREAM_TERMINATED); + wait_stream_state(pcm, PA_STREAM_TERMINATED); pa_stream_unref(pcm->stream); pcm->stream = NULL; } @@ -620,9 +695,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } - pa_stream_set_state_callback(pcm->stream, pulse_stream_state_cb, - pcm->p); - + pa_stream_set_state_callback(pcm->stream, stream_state_cb, pcm); pa_stream_set_latency_update_callback(pcm->stream, stream_latency_cb, pcm); if (io->stream == SND_PCM_STREAM_PLAYBACK) { @@ -659,8 +732,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } - err = - pulse_wait_stream_state(pcm->p, pcm->stream, PA_STREAM_READY); + err = wait_stream_state(pcm, PA_STREAM_READY); if (err < 0) { SNDERR("PulseAudio: Unable to create stream: %s\n", pa_strerror(pa_context_errno(pcm->p->context))); pa_stream_unref(pcm->stream); diff --git a/pulse/pulse.c b/pulse/pulse.c index 6f58a7e..c313182 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -44,26 +44,6 @@ int pulse_check_connection(snd_pulse_t * p) return 0; } -void pulse_stream_state_cb(pa_stream * s, void *userdata) -{ - snd_pulse_t *p = userdata; - - assert(s); - assert(p); - - pa_threaded_mainloop_signal(p->mainloop, 0); -} - -void pulse_stream_success_cb(pa_stream * s, int success, void *userdata) -{ - snd_pulse_t *p = userdata; - - assert(s); - assert(p); - - pa_threaded_mainloop_signal(p->mainloop, 0); -} - void pulse_context_success_cb(pa_context * c, int success, void *userdata) { snd_pulse_t *p = userdata; @@ -101,41 +81,6 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) return 0; } -int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, - pa_stream_state_t target) -{ - pa_stream_state_t state; - - assert(p); - assert(stream); - - if (p->state != PULSE_STATE_READY) - return -EBADFD; - - if (!p->mainloop) - return -EBADFD; - - for (;;) { - int err; - - err = pulse_check_connection(p); - if (err < 0) - return err; - - state = pa_stream_get_state(stream); - - if (state == target) - break; - - if (!PA_STREAM_IS_GOOD(state)) - return -EIO; - - pa_threaded_mainloop_wait(p->mainloop); - } - - return 0; -} - static void context_state_cb(pa_context * c, void *userdata) { snd_pulse_t *p = userdata; diff --git a/pulse/pulse.h b/pulse/pulse.h index 7bf1a5b..51f9a11 100644 --- a/pulse/pulse.h +++ b/pulse/pulse.h @@ -41,13 +41,9 @@ typedef struct snd_pulse { int pulse_check_connection(snd_pulse_t * p); -void pulse_stream_state_cb(pa_stream * s, void *userdata); -void pulse_stream_success_cb(pa_stream * s, int success, void *userdata); void pulse_context_success_cb(pa_context * c, int success, void *userdata); int pulse_wait_operation(snd_pulse_t * p, pa_operation * o); -int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, - pa_stream_state_t target); snd_pulse_t *pulse_new(void); void pulse_free(snd_pulse_t * p); -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 2/3] ASoC: add DAI platform ssi driver for MXC
by javier Martin 04 Aug '09

04 Aug '09

This adds support for DAI platform for the SSI present in MXC platforms. It currently does not support i.MX3, the only thing necessary to do this is to export DMA data for i.MX3 interface which I haven't done because I don't have a i.MX3 based board available. It has been tested on i.MX27 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 2 + sound/soc/imx/Makefile | 2 + sound/soc/imx/mxc-ssi.c | 868 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/mxc-ssi.h | 238 +++++++++++++ 4 files changed, 1110 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index a1bf053..886dadd 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -6,6 +6,8 @@ config SND_MX1_MX2_SOC Say Y or M if you want to add support for codecs attached to the MX1 or MX2 SSI interface. +config SND_MXC_SOC_SSI + tristate diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index c390f0f..6552cb2 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,4 +1,6 @@ # i.MX Platform Support snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o +snd-soc-mxc-ssi-objs := mxc-ssi.o obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o +obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c new file mode 100644 index 0000000..3806ff2 --- /dev/null +++ b/sound/soc/imx/mxc-ssi.c @@ -0,0 +1,868 @@ +/* + * mxc-ssi.c -- SSI driver for Freescale IMX + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Based on mxc-alsa-mc13783 (C) 2006 Freescale. + * + * 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. + * + * TODO: + * Need to rework SSI register defs when new defs go into mainline. + * Add support for TDM and FIFO 1. + * Add support for i.mx3x DMA interface. + * + */ + + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <mach/dma-mx1-mx2.h> +#include <asm/mach-types.h> + +#include "mxc-ssi.h" +#include "mx1_mx2-pcm.h" + +#define SSI1_PORT 0 +#define SSI2_PORT 1 + +static int ssi_active[2] = {0, 0}; + +/* DMA information for mx1_mx2 platforms */ +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out0 = { + .name = "SSI1 PCM Stereo out 0", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI1_BASE_ADDR + STX0, + .event_id = DMA_REQ_SSI1_TX0, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out1 = { + .name = "SSI1 PCM Stereo out 1", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI1_BASE_ADDR + STX1, + .event_id = DMA_REQ_SSI1_TX1, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in0 = { + .name = "SSI1 PCM Stereo in 0", + .transfer_type = DMA_MODE_READ, + .per_address = SSI1_BASE_ADDR + SRX0, + .event_id = DMA_REQ_SSI1_RX0, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in1 = { + .name = "SSI1 PCM Stereo in 1", + .transfer_type = DMA_MODE_READ, + .per_address = SSI1_BASE_ADDR + SRX1, + .event_id = DMA_REQ_SSI1_RX1, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out0 = { + .name = "SSI2 PCM Stereo out 0", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI2_BASE_ADDR + STX0, + .event_id = DMA_REQ_SSI2_TX0, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out1 = { + .name = "SSI2 PCM Stereo out 1", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI2_BASE_ADDR + STX1, + .event_id = DMA_REQ_SSI2_TX1, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in0 = { + .name = "SSI2 PCM Stereo in 0", + .transfer_type = DMA_MODE_READ, + .per_address = SSI2_BASE_ADDR + SRX0, + .event_id = DMA_REQ_SSI2_RX0, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in1 = { + .name = "SSI2 PCM Stereo in 1", + .transfer_type = DMA_MODE_READ, + .per_address = SSI2_BASE_ADDR + SRX1, + .event_id = DMA_REQ_SSI2_RX1, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct clk *ssi_clk0, *ssi_clk1; + +int get_ssi_clk(int ssi, struct device *dev) +{ + switch (ssi) { + case 0: + ssi_clk0 = clk_get(dev, "ssi1"); + if (IS_ERR(ssi_clk0)) + return PTR_ERR(ssi_clk0); + return 0; + case 1: + ssi_clk1 = clk_get(dev, "ssi2"); + if (IS_ERR(ssi_clk1)) + return PTR_ERR(ssi_clk1); + return 0; + default: + return -EINVAL; + } +} +EXPORT_SYMBOL(get_ssi_clk); + +void put_ssi_clk(int ssi) +{ + switch (ssi) { + case 0: + clk_put(ssi_clk0); + ssi_clk0 = NULL; + break; + case 1: + clk_put(ssi_clk1); + ssi_clk1 = NULL; + break; + } +} +EXPORT_SYMBOL(put_ssi_clk); + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 scr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + scr = SSI1_SCR; + pr_debug("%s: SCR for SSI1 is %x\n", __func__, scr); + } else { + scr = SSI2_SCR; + pr_debug("%s: SCR for SSI2 is %x\n", __func__, scr); + } + + if (scr & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) { + scr |= SSI_SCR_SYS_CLK_EN; + pr_debug("%s: clk of is output\n", __func__); + } else { + scr &= ~SSI_SCR_SYS_CLK_EN; + pr_debug("%s: clk of is input\n", __func__); + } + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + pr_debug("%s: writeback of SSI1_SCR\n", __func__); + SSI1_SCR = scr; + } else { + pr_debug("%s: writeback of SSI2_SCR\n", __func__); + SSI2_SCR = scr; + } + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + u32 stccr, srccr; + + pr_debug("%s\n", __func__); + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) + return 0; + srccr = SSI1_STCCR; + stccr = SSI1_STCCR; + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) + return 0; + srccr = SSI2_STCCR; + stccr = SSI2_STCCR; + } + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCCR = stccr; + SSI1_SRCCR = srccr; + } else { + SSI2_STCCR = stccr; + SSI2_SRCCR = srccr; + } + return 0; +} + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int mask, int slots) +{ + u32 stmsk, srmsk, stccr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + stccr = SSI1_STCCR; + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + stccr = SSI2_STCCR; + } + + stmsk = srmsk = mask; + stccr &= ~SSI_STCCR_DC_MASK; + stccr |= SSI_STCCR_DC(slots - 1); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STMSK = stmsk; + SSI1_SRMSK = srmsk; + SSI1_SRCCR = SSI1_STCCR = stccr; + } else { + SSI2_STMSK = stmsk; + SSI2_SRMSK = srmsk; + SSI2_SRCCR = SSI2_STCCR = stccr; + } + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + u32 stcr = 0, srcr = 0, scr; + + /* + * This is done to avoid this function to modify + * previous set values in stcr + */ + stcr = SSI1_STCR; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + scr = SSI1_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); + else + scr = SSI2_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); + + if (scr & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + stcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + srcr |= SSI_SRCR_RFSI | SSI_SRCR_REFS | SSI_SRCR_RXBIT0; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + stcr |= SSI_STCR_TXBIT0; + srcr |= SSI_SRCR_RXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + stcr |= SSI_STCR_TFSL; + srcr |= SSI_SRCR_RFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + stcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + srcr |= SSI_SRCR_RFSL | SSI_SRCR_REFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + stcr |= SSI_STCR_TFSI; + stcr &= ~SSI_STCR_TSCKP; + srcr |= SSI_SRCR_RFSI; + srcr &= ~SSI_SRCR_RSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + stcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + srcr &= ~(SSI_SRCR_RSCKP | SSI_SRCR_RFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + stcr &= ~SSI_STCR_TFSI; + stcr |= SSI_STCR_TSCKP; + srcr &= ~SSI_SRCR_RFSI; + srcr |= SSI_SRCR_RSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + stcr |= SSI_STCR_TFDIR; + srcr |= SSI_SRCR_RFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + stcr |= SSI_STCR_TXDIR; + srcr |= SSI_SRCR_RXDIR; + break; + } + + /* sync */ + if (!(fmt & SND_SOC_DAIFMT_ASYNC)) + scr |= SSI_SCR_SYN; + + /* tdm - only for stereo atm */ + if (fmt & SND_SOC_DAIFMT_TDM) + scr |= SSI_SCR_NET; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCR = stcr; + SSI1_SRCR = srcr; + SSI1_SCR = scr; + } else { + SSI2_STCR = stcr; + SSI2_SRCR = srcr; + SSI2_SCR = scr; + } + + return 0; +} + +static int imx_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set up TX DMA params */ + switch (cpu_dai->id) { + case IMX_DAI_SSI0: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out0; + break; + case IMX_DAI_SSI1: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out1; + break; + case IMX_DAI_SSI2: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out0; + break; + case IMX_DAI_SSI3: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out1; + } + pr_debug("%s: (playback)\n", __func__); + } else { + /* set up RX DMA params */ + switch (cpu_dai->id) { + case IMX_DAI_SSI0: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in0; + break; + case IMX_DAI_SSI1: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in1; + break; + case IMX_DAI_SSI2: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in0; + break; + case IMX_DAI_SSI3: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in1; + } + pr_debug("%s: (capture)\n", __func__); + } + + /* + * we cant really change any SSI values after SSI is enabled + * need to fix in software for max flexibility - lrg + */ + if (cpu_dai->active) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + /* reset the SSI port - Sect 45.4.4 */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + + if (!ssi_clk0) + return -EINVAL; + + if (ssi_active[SSI1_PORT]++) { + pr_debug("%s: exit before reset\n", __func__); + return 0; + } + + /* SSI1 Reset */ + SSI1_SCR = 0; + + SSI1_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | + SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | + SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | + SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); + } else { + + if (!ssi_clk1) + return -EINVAL; + + if (ssi_active[SSI2_PORT]++) { + pr_debug("%s: exit before reset\n", __func__); + return 0; + } + + /* SSI2 Reset */ + SSI2_SCR = 0; + + SSI2_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | + SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | + SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | + SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); + } + + return 0; +} + +int imx_ssi_hw_tx_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; + u32 stccr, stcr, sier; + + pr_debug("%s\n", __func__); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + stccr = SSI1_STCCR & ~SSI_STCCR_WL_MASK; + stcr = SSI1_STCR; + sier = SSI1_SIER; + } else { + stccr = SSI2_STCCR & ~SSI_STCCR_WL_MASK; + stcr = SSI2_STCR; + sier = SSI2_SIER; + } + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + stccr |= SSI_STCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + stccr |= SSI_STCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + stccr |= SSI_STCCR_WL(24); + break; + } + + /* enable interrupts */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + stcr |= SSI_STCR_TFEN0; + else + stcr |= SSI_STCR_TFEN1; + sier |= SSI_SIER_TDMAE; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCR = stcr; + SSI1_STCCR = stccr; + SSI1_SIER = sier; + } else { + SSI2_STCR = stcr; + SSI2_STCCR = stccr; + SSI2_SIER = sier; + } + + return 0; +} + +int imx_ssi_hw_rx_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; + u32 srccr, srcr, sier; + + pr_debug("%s\n", __func__); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + srccr = SSI1_SRCCR & ~SSI_SRCCR_WL_MASK; + srcr = SSI1_SRCR; + sier = SSI1_SIER; + } else { + srccr = SSI2_SRCCR & ~SSI_SRCCR_WL_MASK; + srcr = SSI2_SRCR; + sier = SSI2_SIER; + } + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + srccr |= SSI_SRCCR_WL(24); + break; + } + + /* enable interrupts */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + srcr |= SSI_SRCR_RFEN0; + else + srcr |= SSI_SRCR_RFEN1; + sier |= SSI_SIER_RDMAE; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_SRCR = srcr; + SSI1_SRCCR = srccr; + SSI1_SIER = sier; + } else { + SSI2_SRCR = srcr; + SSI2_SRCCR = srccr; + SSI2_SIER = sier; + } + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +int imx_ssi_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_dai *cpu_dai = rtd->dai->cpu_dai; + + int ret; + + /* cant change any parameters when SSI is running */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + } + + /* + * Configure both tx and rx params with the same settings. This is + * really a harware restriction because SSI must be disabled until + * we can change those values. If there is an active audio stream in + * one direction, enabling the other direction with different + * settings would mean disturbing the running one. + */ + ret = imx_ssi_hw_tx_params(substream, params); + if (ret < 0) + return ret; + return imx_ssi_hw_rx_params(substream, params); +} + +int imx_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + pr_debug("%s\n", __func__); + + /* Enable clks here to follow SSI recommended init sequence */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + ret = clk_enable(ssi_clk0); + if (ret < 0) + printk(KERN_ERR "Unable to enable ssi_clk0\n"); + } else { + ret = clk_enable(ssi_clk1); + if (ret < 0) + printk(KERN_ERR "Unable to enable ssi_clk1\n"); + } + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 scr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + scr = SSI1_SCR; + else + scr = SSI2_SCR; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE | SSI_SCR_SSIEN; + else + scr |= SSI_SCR_RE | SSI_SCR_SSIEN; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + SSI1_SCR = scr; + else + SSI2_SCR = scr; + + return 0; +} + +static void imx_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + /* shutdown SSI if neither Tx or Rx is active */ + if (!cpu_dai->active) { + + if (cpu_dai->id == IMX_DAI_SSI0 || + cpu_dai->id == IMX_DAI_SSI2) { + + if (--ssi_active[SSI1_PORT] > 1) + return; + + SSI1_SCR = 0; + clk_disable(ssi_clk0); + } else { + if (--ssi_active[SSI2_PORT]) + return; + SSI2_SCR = 0; + clk_disable(ssi_clk1); + } + } +} + +#ifdef CONFIG_PM +static int imx_ssi_suspend(struct platform_device *dev, + struct snd_soc_dai *dai) +{ + return 0; +} + +static int imx_ssi_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define imx_ssi_suspend NULL +#define imx_ssi_resume NULL +#endif + +#define IMX_SSI_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define IMX_SSI_BITS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .startup = imx_ssi_startup, + .shutdown = imx_ssi_shutdown, + .trigger = imx_ssi_trigger, + .prepare = imx_ssi_prepare, + .hw_params = imx_ssi_hw_params, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_fmt = imx_ssi_set_dai_fmt, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, +}; + +struct snd_soc_dai imx_ssi_pcm_dai[] = { +{ + .name = "imx-i2s-1-0", + .id = IMX_DAI_SSI0, + .suspend = imx_ssi_suspend, + .resume = imx_ssi_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-2-0", + .id = IMX_DAI_SSI1, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-1-1", + .id = IMX_DAI_SSI2, + .suspend = imx_ssi_suspend, + .resume = imx_ssi_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-2-1", + .id = IMX_DAI_SSI3, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +}; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int __init imx_ssi_init(void) +{ + return snd_soc_register_dais(imx_ssi_pcm_dai, + ARRAY_SIZE(imx_ssi_pcm_dai)); +} + +static void __exit imx_ssi_exit(void) +{ + snd_soc_unregister_dais(imx_ssi_pcm_dai, + ARRAY_SIZE(imx_ssi_pcm_dai)); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); +MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com"); +MODULE_DESCRIPTION("i.MX ASoC I2S driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.h b/sound/soc/imx/mxc-ssi.h new file mode 100644 index 0000000..12bbdc9 --- /dev/null +++ b/sound/soc/imx/mxc-ssi.h @@ -0,0 +1,238 @@ +/* + * 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. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#include <mach/hardware.h> + +/* SSI regs definition - MOVE to /arch/arm/plat-mxc/include/mach/ when stable */ +#define SSI1_IO_BASE_ADDR IO_ADDRESS(SSI1_BASE_ADDR) +#define SSI2_IO_BASE_ADDR IO_ADDRESS(SSI2_BASE_ADDR) + +#define STX0 0x00 +#define STX1 0x04 +#define SRX0 0x08 +#define SRX1 0x0c +#define SCR 0x10 +#define SISR 0x14 +#define SIER 0x18 +#define STCR 0x1c +#define SRCR 0x20 +#define STCCR 0x24 +#define SRCCR 0x28 +#define SFCSR 0x2c +#define STR 0x30 +#define SOR 0x34 +#define SACNT 0x38 +#define SACADD 0x3c +#define SACDAT 0x40 +#define SATAG 0x44 +#define STMSK 0x48 +#define SRMSK 0x4c + +#define SSI1_STX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX0))) +#define SSI1_STX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX1))) +#define SSI1_SRX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX0))) +#define SSI1_SRX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX1))) +#define SSI1_SCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SCR))) +#define SSI1_SISR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SISR))) +#define SSI1_SIER (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SIER))) +#define SSI1_STCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCR))) +#define SSI1_SRCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCR))) +#define SSI1_STCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCCR))) +#define SSI1_SRCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCCR))) +#define SSI1_SFCSR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SFCSR))) +#define SSI1_STR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STR))) +#define SSI1_SOR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SOR))) +#define SSI1_SACNT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACNT))) +#define SSI1_SACADD (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACADD))) +#define SSI1_SACDAT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACDAT))) +#define SSI1_SATAG (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SATAG))) +#define SSI1_STMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STMSK))) +#define SSI1_SRMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRMSK))) + + +#define SSI2_STX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX0))) +#define SSI2_STX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX1))) +#define SSI2_SRX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX0))) +#define SSI2_SRX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX1))) +#define SSI2_SCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SCR))) +#define SSI2_SISR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SISR))) +#define SSI2_SIER (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SIER))) +#define SSI2_STCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCR))) +#define SSI2_SRCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCR))) +#define SSI2_STCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCCR))) +#define SSI2_SRCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCCR))) +#define SSI2_SFCSR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SFCSR))) +#define SSI2_STR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STR))) +#define SSI2_SOR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SOR))) +#define SSI2_SACNT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACNT))) +#define SSI2_SACADD (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACADD))) +#define SSI2_SACDAT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACDAT))) +#define SSI2_SATAG (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SATAG))) +#define SSI2_STMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STMSK))) +#define SSI2_SRMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRMSK))) + +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 15) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 15) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + + +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) + +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (x << 4) +#define SSI_SACNT_RD (x << 3) +#define SSI_SACNT_TIF (x << 2) +#define SSI_SACNT_FV (x << 1) +#define SSI_SACNT_AC97EN (x << 0) + +/* Watermarks for FIFO's */ +#define TXFIFO_WATERMARK 0x4 +#define RXFIFO_WATERMARK 0x4 + +/* i.MX DAI SSP ID's */ +#define IMX_DAI_SSI0 0 /* SSI1 FIFO 0 */ +#define IMX_DAI_SSI1 1 /* SSI1 FIFO 1 */ +#define IMX_DAI_SSI2 2 /* SSI2 FIFO 0 */ +#define IMX_DAI_SSI3 3 /* SSI2 FIFO 1 */ + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + + +/* SSI Div 2 */ +#define IMX_SSI_DIV_2_OFF (~SSI_STCCR_DIV2) +#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2 + +extern struct snd_soc_dai imx_ssi_pcm_dai[4]; +extern int get_ssi_clk(int ssi, struct device *dev); +extern void put_ssi_clk(int ssi); +#endif --- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 2

[alsa-devel] [PATCH 1/3] ASoC: add DMA platform driver for MX1x and MX2x
by javier Martin 04 Aug '09

04 Aug '09

This adds support for DMA platform valid for i.MX1 and i.MX2 platforms. This is not valid for i.MX3 since it doesn't share the same DMA interface than i.MX1 and i.MX2. It has been tested on i.MX27 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/Makefile | 1 + sound/soc/imx/Kconfig | 11 + sound/soc/imx/Makefile | 4 + sound/soc/imx/mx1_mx2-pcm.c | 487 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/mx1_mx2-pcm.h | 48 +++++ 5 files changed, 551 insertions(+), 0 deletions(-) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..8120b52 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,3 +11,4 @@ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += imx/ diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig new file mode 100644 index 0000000..a1bf053 --- /dev/null +++ b/sound/soc/imx/Kconfig @@ -0,0 +1,11 @@ +config SND_MX1_MX2_SOC + tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" + depends on (ARCH_MX2 || ARCH_MX1) && SND + select SND_PCM + help + Say Y or M if you want to add support for codecs attached to + the MX1 or MX2 SSI interface. + + + + diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile new file mode 100644 index 0000000..c390f0f --- /dev/null +++ b/sound/soc/imx/Makefile @@ -0,0 +1,4 @@ +# i.MX Platform Support +snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o + +obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c new file mode 100644 index 0000000..94807f8 --- /dev/null +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -0,0 +1,487 @@ +/* + * mx1_mx2-pcm.c -- ALSA SoC interface for Freescale i.MX1x, i.MX2x CPUs + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin(a)vista-silicon.com + * + * Based on mxc-pcm.c by Liam Girdwood. + * + * 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. + * + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma-mx1-mx2.h> + +#include "mx1_mx2-pcm.h" + + +static const struct snd_pcm_hardware mx1_mx2_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = 32 * 1024, + .period_bytes_min = 64, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +struct mx1_mx2_runtime_data { + int dma_ch; + int active; + unsigned int period; + unsigned int periods; + int tx_spin; + spinlock_t dma_lock; + struct mx1_mx2_pcm_dma_params *dma_params; +}; + + +/** + * This function stops the current dma transfer for playback + * and clears the dma pointers. + * + * @param substream pointer to the structure of the current stream. + * + */ +static int audio_stop_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->dma_lock, flags); + + pr_debug("%s\n", __func__); + + prtd->active = 0; + prtd->period = 0; + prtd->periods = 0; + + /* this stops the dma channel and clears the buffer ptrs */ + + imx_dma_disable(prtd->dma_ch); + + spin_unlock_irqrestore(&prtd->dma_lock, flags); + + return 0; +} + +/** + * This function is called whenever a new audio block needs to be + * transferred to the codec. The function receives the address and the size + * of the new block and start a new DMA transfer. + * + * @param substream pointer to the structure of the current stream. + * + */ +static int dma_new_period(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned int dma_size; + unsigned int offset; + int ret = 0; + dma_addr_t mem_addr; + unsigned int dev_addr; + + if (prtd->active) { + dma_size = frames_to_bytes(runtime, runtime->period_size); + offset = dma_size * prtd->period; + + pr_debug("%s: period (%d) out of (%d)\n", __func__, + prtd->period, + runtime->periods); + pr_debug("period_size %d frames\n offset %d bytes\n", + (unsigned int)runtime->period_size, + offset); + pr_debug("dma_size %d bytes\n", dma_size); + + snd_BUG_ON(dma_size > mx1_mx2_pcm_hardware.period_bytes_max); + + mem_addr = (dma_addr_t)(runtime->dma_addr + offset); + dev_addr = prtd->dma_params->per_address; + pr_debug("%s: mem_addr is %x\n dev_addr is %x\n", + __func__, mem_addr, dev_addr); + + ret = imx_dma_setup_single(prtd->dma_ch, mem_addr, + dma_size, dev_addr, + prtd->dma_params->transfer_type); + if (ret < 0) { + printk(KERN_ERR "Error configuring DMA\n"); + return ret; + } + imx_dma_enable(prtd->dma_ch); + + pr_debug("%s: transfer enabled\nmem_addr = %x\n", + __func__, (unsigned int) mem_addr); + pr_debug("dev_addr = %x\ndma_size = %d\n", + (unsigned int) dev_addr, dma_size); + + prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ + prtd->period++; + prtd->period %= runtime->periods; + } + return ret; +} + + +/** + * This is a callback which will be called + * when a TX transfer finishes. The call occurs + * in interrupt context. + * + * @param dat pointer to the structure of the current stream. + * + */ +static void audio_dma_irq(int channel, void *data) +{ + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct mx1_mx2_runtime_data *prtd; + unsigned int dma_size; + unsigned int previous_period; + unsigned int offset; + + substream = data; + runtime = substream->runtime; + prtd = runtime->private_data; + previous_period = prtd->periods; + dma_size = frames_to_bytes(runtime, runtime->period_size); + offset = dma_size * previous_period; + + prtd->tx_spin = 0; + prtd->periods++; + prtd->periods %= runtime->periods; + + pr_debug("%s: irq per %d offset %x\n", __func__, prtd->periods, offset); + + /* + * If we are getting a callback for an active stream then we inform + * the PCM middle layer we've finished a period + */ + if (prtd->active) + snd_pcm_period_elapsed(substream); + + /* + * Trig next DMA transfer + */ + dma_new_period(substream); +} + +/** + * This function configures the hardware to allow audio + * playback operations. It is called by ALSA framework. + * + * @param substream pointer to the structure of the current stream. + * + * @return 0 on success, -1 otherwise. + */ +static int +snd_mx1_mx2_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + prtd->period = 0; + prtd->periods = 0; + + return 0; +} + +static int mx1_mx2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_ERR "%s: failed to malloc pcm pages\n", __func__); + return ret; + } + + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_addr 0x(%x)\n", + __func__, (unsigned int)runtime->dma_addr); + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_area 0x(%x)\n", + __func__, (unsigned int)runtime->dma_area); + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_bytes 0x(%x)\n", + __func__, (unsigned int)runtime->dma_bytes); + + return ret; +} + +static int mx1_mx2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + imx_dma_free(prtd->dma_ch); + + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int mx1_mx2_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct mx1_mx2_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->tx_spin = 0; + /* requested stream startup */ + prtd->active = 1; + pr_debug("%s: starting dma_new_period\n", __func__); + ret = dma_new_period(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* requested stream shutdown */ + pr_debug("%s: stopping dma transfer\n", __func__); + ret = audio_stop_dma(substream); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static snd_pcm_uframes_t +mx1_mx2_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned int offset = 0; + + /* tx_spin value is used here to check if a transfer is active */ + if (prtd->tx_spin) { + offset = (runtime->period_size * (prtd->periods)) + + (runtime->period_size >> 1); + if (offset >= runtime->buffer_size) + offset = runtime->period_size >> 1; + } else { + offset = (runtime->period_size * (prtd->periods)); + if (offset >= runtime->buffer_size) + offset = 0; + } + pr_debug("%s: pointer offset %x\n", __func__, offset); + + return offset; +} + +static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mx1_mx2_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &mx1_mx2_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + prtd = kzalloc(sizeof(struct mx1_mx2_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + + runtime->private_data = prtd; + + if (!dma_data) + return -ENODEV; + + prtd->dma_params = dma_data; + + pr_debug("%s: Requesting dma channel (%s)\n", __func__, + prtd->dma_params->name); + prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, + DMA_PRIO_HIGH); + if (prtd->dma_ch < 0) { + printk(KERN_ERR "Error requesting dma channel\n"); + return ret; + } + imx_dma_config_burstlen(prtd->dma_ch, + prtd->dma_params->watermark_level); + + ret = imx_dma_config_channel(prtd->dma_ch, + prtd->dma_params->per_config, + prtd->dma_params->mem_config, + prtd->dma_params->event_id, 0); + + if (ret) { + pr_debug(KERN_ERR "Error configuring dma channel %d\n", + prtd->dma_ch); + return ret; + } + + pr_debug("%s: Setting tx dma callback function\n", __func__); + ret = imx_dma_setup_handlers(prtd->dma_ch, + audio_dma_irq, NULL, + (void *)substream); + if (ret < 0) { + printk(KERN_ERR "Error setting dma callback function\n"); + return ret; + } + return 0; + + out: + return ret; +} + +static int mx1_mx2_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int mx1_mx2_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops mx1_mx2_pcm_ops = { + .open = mx1_mx2_pcm_open, + .close = mx1_mx2_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mx1_mx2_pcm_hw_params, + .hw_free = mx1_mx2_pcm_hw_free, + .prepare = snd_mx1_mx2_prepare, + .trigger = mx1_mx2_pcm_trigger, + .pointer = mx1_mx2_pcm_pointer, + .mmap = mx1_mx2_pcm_mmap, +}; + +static u64 mx1_mx2_pcm_dmamask = 0xffffffff; + +static int mx1_mx2_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = mx1_mx2_pcm_hardware.buffer_bytes_max; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Reserve uncached-buffered memory area for DMA */ + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + pr_debug("%s: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + __func__, (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void mx1_mx2_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +int mx1_mx2_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &mx1_mx2_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + pr_debug("%s: preallocate playback buffer\n", __func__); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + pr_debug("%s: preallocate capture buffer\n", __func__); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform mx1_mx2_soc_platform = { + .name = "mx1_mx2-audio", + .pcm_ops = &mx1_mx2_pcm_ops, + .pcm_new = mx1_mx2_pcm_new, + .pcm_free = mx1_mx2_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(mx1_mx2_soc_platform); + +static int __init mx1_mx2_soc_platform_init(void) +{ + return snd_soc_register_platform(&mx1_mx2_soc_platform); +} +module_init(mx1_mx2_soc_platform_init); + +static void __exit mx1_mx2_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&mx1_mx2_soc_platform); +} +module_exit(mx1_mx2_soc_platform_exit); + +MODULE_AUTHOR("Javier Martin, javier.martin(a)vista-silicon.com"); +MODULE_DESCRIPTION("Freescale i.MX2x, i.MX1x PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mx1_mx2-pcm.h b/sound/soc/imx/mx1_mx2-pcm.h new file mode 100644 index 0000000..e1e3a3f --- /dev/null +++ b/sound/soc/imx/mx1_mx2-pcm.h @@ -0,0 +1,48 @@ +/* + * mx1_mx2-pcm.h :- ASoC platform header for Freescale i.MX1x, i.MX2x + * + * 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. + */ + +#ifndef _MXC_PCM_H +#define _MXC_PCM_H + +/* AUDMUX register definitions */ +#define AUDMUX_IO_BASE_ADDR IO_ADDRESS(AUDMUX_BASE_ADDR) + +#define DAM_HPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x00))) +#define DAM_HPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x04))) +#define DAM_HPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x08))) +#define DAM_PPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x10))) +#define DAM_PPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x14))) +#define DAM_PPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x1C))) + +#define AUDMUX_HPCR_TFSDIR (1 << 31) +#define AUDMUX_HPCR_TCLKDIR (1 << 30) +#define AUDMUX_HPCR_TFCSEL(x) (((x) & 0xff) << 26) +#define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) +#define AUDMUX_HPCR_SYN (1 << 12) + +#define AUDMUX_PPCR_TFSDIR (1 << 31) +#define AUDMUX_PPCR_TCLKDIR (1 << 30) +#define AUDMUX_PPCR_TFCSEL(x) (((x) & 0xff) << 26) +#define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) +#define AUDMUX_PPCR_SYN (1 << 12) + +/* DMA information for mx1_mx2 platforms */ +struct mx1_mx2_pcm_dma_params { + char *name; /* stream identifier */ + unsigned int transfer_type; /* READ or WRITE DMA transfer */ + dma_addr_t per_address; /* physical address of SSI fifo */ + int event_id; /* fixed DMA number for SSI fifo */ + int watermark_level; /* SSI fifo watermark level */ + int per_config; /* DMA Config flags for peripheral */ + int mem_config; /* DMA Config flags for RAM */ + }; + +/* platform data */ +extern struct snd_soc_platform mx1_mx2_soc_platform; + +#endif -- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 1

[alsa-devel] [PATCH - BT and FM audio for zoom2 1/2] Add clock-only codec to provide McBSP clock source.
by sean.mcneil＠ti.com 04 Aug '09

04 Aug '09

From: Sean McNeil <sean.mcneil(a)ti.com> Signed-off-by: Sean McNeil <sean.mcneil(a)ti.com> diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 818fb37..c343ceb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -56,7 +56,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ 0x01, /* REG_AUDIO_IF (0xE) */ - 0x00, /* REG_VOICE_IF (0xF) */ + 0x04, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ 0x6c, /* REG_ARXR2PGA (0x12) */ @@ -118,25 +118,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; +struct substream_item { + struct list_head started; + struct list_head configured; + struct snd_pcm_substream *substream; + int use256FS; +}; + /* codec private data */ struct twl4030_priv { + struct mutex mutex; + + unsigned int extClock; unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; - - unsigned int configured; - unsigned int rate; - unsigned int sample_bits; - unsigned int channels; + struct list_head started_list; + struct list_head config_list; unsigned int sysclk; /* Headset output state handling */ unsigned int hsl_enabled; unsigned int hsr_enabled; + + struct snd_pcm_hw_params params; }; /* @@ -966,7 +973,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, unsigned short val; unsigned short mask, bitmask; - if (twl4030->configured) { + if (!list_empty(&twl4030->config_list)) { printk(KERN_ERR "twl4030 operation mode cannot be " "changed on-the-fly\n"); return -EBUSY; @@ -1534,34 +1541,68 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } -static void twl4030_constraints(struct twl4030_priv *twl4030, - struct snd_pcm_substream *mst_substream) +static unsigned int twl4030_rate_min(struct substream_item *item, + unsigned int rate) { - struct snd_pcm_substream *slv_substream; - - /* Pick the stream, which need to be constrained */ - if (mst_substream == twl4030->master_substream) - slv_substream = twl4030->slave_substream; - else if (mst_substream == twl4030->slave_substream) - slv_substream = twl4030->master_substream; - else /* This should not happen.. */ - return; + static const unsigned int table[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 96000}; + unsigned int value = rate; + + if (item->use256FS) { + int i; + rate *= 256; + for (i = 0; i < ARRAY_SIZE(table); i++) + if (rate % table[i] == 0) { + value = table[i]; + break; + } + } + return value; +} - /* Set the constraints according to the already configured stream */ - snd_pcm_hw_constraint_minmax(slv_substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - twl4030->rate, - twl4030->rate); - - snd_pcm_hw_constraint_minmax(slv_substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - twl4030->sample_bits, - twl4030->sample_bits); - - snd_pcm_hw_constraint_minmax(slv_substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - twl4030->channels, - twl4030->channels); +static unsigned int twl4030_rate_max(struct substream_item *item, + unsigned int rate) +{ + static const unsigned int table[] = { + 96000, 48000, 44100, 32000, 24000, + 22050, 16000, 12000, 11025, 8000}; + unsigned int value = rate; + + if (item->use256FS) { + int i; + rate *= 256; + for (i = 0; i < ARRAY_SIZE(table); i++) + if (rate % table[i] == 0) { + value = table[i]; + break; + } + } + return value; +} + +static void twl4030_constraints(struct twl4030_priv *twl4030) +{ + struct substream_item *item; + unsigned int value; + + list_for_each_entry(item, &twl4030->started_list, started) { + + /* Set constraints according to the already configured stream */ + value = params_rate(&twl4030->params); + if (value) + snd_pcm_hw_constraint_minmax(item->substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + twl4030_rate_min(item, value), + twl4030_rate_max(item, value)); + + value = hw_param_interval(&twl4030->params, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + if (value && !item->use256FS) + snd_pcm_hw_constraint_minmax(item->substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + value, value); + } } /* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for @@ -1586,6 +1627,53 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, twl4030_write(codec, TWL4030_REG_OPTION, reg); } +static int twl4030_new_substream(struct twl4030_priv *twl4030, + struct snd_pcm_substream *substream, int use256FS) +{ + struct substream_item *item; + + item = kzalloc(sizeof(struct snd_pcm_substream), GFP_KERNEL); + if (!item) + return -ENOMEM; + + item->substream = substream; + item->use256FS = use256FS; + + mutex_lock(&twl4030->mutex); + list_add_tail(&item->started, &twl4030->started_list); + twl4030->extClock += item->use256FS; + mutex_unlock(&twl4030->mutex); + + return 0; +} + +static void twl4030_del_substream(struct twl4030_priv *twl4030, + struct snd_pcm_substream *substream) +{ + struct substream_item *item; + + mutex_lock(&twl4030->mutex); + + list_for_each_entry(item, &twl4030->config_list, configured) { + if (item->substream == substream) { + printk(KERN_ERR "TWL4030 deleted substream still configured!\n"); + list_del(&item->configured); + break; + } + } + + list_for_each_entry(item, &twl4030->started_list, started) { + if (item->substream == substream) { + list_del(&item->started); + twl4030->extClock -= item->use256FS; + kfree(item); + break; + } + } + + mutex_unlock(&twl4030->mutex); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1594,82 +1682,45 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - if (twl4030->master_substream) { - twl4030->slave_substream = substream; - /* The DAI has one configuration for playback and capture, so - * if the DAI has been already configured then constrain this - * substream to match it. */ - if (twl4030->configured) - twl4030_constraints(twl4030, twl4030->master_substream); - } else { - if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & - TWL4030_OPTION_1)) { - /* In option2 4 channel is not supported, set the - * constraint for the first stream for channels, the - * second stream will 'inherit' this cosntraint */ - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 2, 2); - } - twl4030->master_substream = substream; + if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) { + /* In option2 4 channel is not supported, set the + * constraint for the first stream for channels, the + * second stream will 'inherit' this cosntraint */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, 2); } - return 0; + return twl4030_new_substream(twl4030, substream, 0); } static void twl4030_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + 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 twl4030_priv *twl4030 = codec->private_data; - if (twl4030->master_substream == substream) - twl4030->master_substream = twl4030->slave_substream; - - twl4030->slave_substream = NULL; - - /* If all streams are closed, or the remaining stream has not yet - * been configured than set the DAI as not configured. */ - if (!twl4030->master_substream) - twl4030->configured = 0; - else if (!twl4030->master_substream->runtime->channels) - twl4030->configured = 0; + twl4030_del_substream(twl4030, substream); /* If the closing substream had 4 channel, do the necessary cleanup */ if (substream->runtime->channels == 4) twl4030_tdm_enable(codec, substream->stream, 0); } -static int twl4030_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +int twl4030_set_rate(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) { - 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 twl4030_priv *twl4030 = codec->private_data; - u8 mode, old_mode, format, old_format; + u8 mode, old_mode; - /* If the substream has 4 channel, do the necessary setup */ - if (params_channels(params) == 4) { - format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); - - /* Safety check: are we in the correct operating mode and - * the interface is in TDM mode? */ - if ((mode & TWL4030_OPTION_1) && - ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM)) - twl4030_tdm_enable(codec, substream->stream, 1); - else - return -EINVAL; + if (params_rate(&twl4030->params) && + params_rate(&twl4030->params) != params_rate(params)) { + return -EBUSY; } - if (twl4030->configured) - /* Ignoring hw_params for already configured DAI */ - return 0; - /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1712,6 +1763,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + params_rate(&twl4030->params) = params_rate(params); + if (mode != old_mode) { /* change rate and set CODECPDZ */ twl4030_codec_enable(codec, 0); @@ -1719,10 +1772,51 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, twl4030_codec_enable(codec, 1); } + return 0; +} +EXPORT_SYMBOL_GPL(twl4030_set_rate); + +int twl4030_get_clock_divisor(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct twl4030_priv *twl4030 = codec->private_data; + int clock, divisor; + + clock = params_rate(&twl4030->params) * 256; + divisor = clock / params_rate(params); + divisor /= params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + divisor /= 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + divisor /= 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + divisor /= 24; + break; + default: + printk(KERN_ERR "TWL4030 get_clock_divisor: unknown format %d\n", + params_format(params)); + return -EINVAL; + } + + return divisor; +} +EXPORT_SYMBOL_GPL(twl4030_get_clock_divisor); + +static int twl4030_set_format(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct twl4030_priv *twl4030 = codec->private_data; + u8 format, old_format; + /* sample size */ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); - format = old_format; - format &= ~TWL4030_DATA_WIDTH; + format = old_format & ~TWL4030_DATA_WIDTH; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: format |= TWL4030_DATA_WIDTH_16S_16W; @@ -1736,31 +1830,96 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (format != old_format) { + if (format == old_format) + return 0; - /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_codec_enable(codec, 0); + if (params_format(&twl4030->params) && + params_format(&twl4030->params) != params_format(params)) + return -EBUSY; - /* change format */ - twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + *hw_param_mask(&twl4030->params, SNDRV_PCM_HW_PARAM_FORMAT) = + *hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - /* set CODECPDZ afterwards */ - twl4030_codec_enable(codec, 1); + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_codec_enable(codec, 0); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_codec_enable(codec, 1); + + return 0; +} + +static int twl4030_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 twl4030_priv *twl4030 = codec->private_data; + int rval; + + mutex_lock(&twl4030->mutex); + + /* If the substream has 4 channel, do the necessary setup */ + if (params_channels(params) == 4) { + /* Safety check: are we in the correct operating mode? */ + if ((twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) { + twl4030_tdm_enable(codec, substream->stream, 1); + } else { + mutex_unlock(&twl4030->mutex); + return -EINVAL; + } } - /* Store the important parameters for the DAI configuration and set - * the DAI as configured */ - twl4030->configured = 1; - twl4030->rate = params_rate(params); - twl4030->sample_bits = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; - twl4030->channels = params_channels(params); + rval = twl4030_set_rate(codec, params); + if (rval < 0) { + mutex_unlock(&twl4030->mutex); + return rval; + } - /* If both playback and capture streams are open, and one of them + rval = twl4030_set_format(codec, params); + if (rval < 0) { + mutex_unlock(&twl4030->mutex); + return rval; + } + + /* If any other streams are currently open, and one of them * is setting the hw parameters right now (since we are here), set - * constraints to the other stream to match the current one. */ - if (twl4030->slave_substream) - twl4030_constraints(twl4030, substream); + * constraints to the other stream(s) to match the current one. */ + twl4030_constraints(twl4030); + + mutex_unlock(&twl4030->mutex); + + return 0; +} + +static int twl4030_hw_free(struct snd_pcm_substream *substream, + 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 twl4030_priv *twl4030 = codec->private_data; + struct substream_item *item; + + mutex_lock(&twl4030->mutex); + + list_for_each_entry(item, &twl4030->config_list, configured) { + if (item->substream == substream) { + list_del(&item->configured); + break; + } + } + + if (list_empty(&twl4030->config_list)) + memset(&twl4030->params, 0, sizeof(twl4030->params)); + + mutex_unlock(&twl4030->mutex); return 0; } @@ -1797,10 +1956,39 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_set_ext_clock(struct snd_soc_codec *codec, int enable) +{ + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + + if (enable) + format = old_format | TWL4030_CLK256FS_EN; + else + format = old_format & ~TWL4030_CLK256FS_EN; + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_codec_enable(codec, 0); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_codec_enable(codec, 1); + } + + return 0; +} + static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = codec->private_data; + int use256FS = 0; u8 old_format, format; /* get format */ @@ -1811,11 +1999,10 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_AIF_SLAVE_EN); - format &= ~(TWL4030_CLK256FS_EN); break; case SND_SOC_DAIFMT_CBS_CFS: format |= TWL4030_AIF_SLAVE_EN; - format |= TWL4030_CLK256FS_EN; + use256FS = 1; break; default: return -EINVAL; @@ -1846,7 +2033,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, twl4030_codec_enable(codec, 1); } - return 0; + return twl4030_set_ext_clock(codec, use256FS | twl4030->extClock); } static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) @@ -1890,6 +2077,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, 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 twl4030_priv *twl4030 = codec->private_data; u8 infreq; u8 mode; @@ -1917,7 +2105,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, return -EINVAL; } - return 0; + return twl4030_new_substream(twl4030, substream, 1); } static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, @@ -1926,6 +2114,9 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, 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 twl4030_priv *twl4030 = codec->private_data; + + twl4030_del_substream(twl4030, substream); /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -1937,15 +2128,18 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, 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 twl4030_priv *twl4030 = codec->private_data; u8 old_mode, mode; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 1); + mutex_lock(&twl4030->mutex); + /* bit rate */ - old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) - & ~(TWL4030_CODECPDZ); - mode = old_mode; + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode & ~TWL4030_APLL_RATE; switch (params_rate(params)) { case 8000: @@ -1955,6 +2149,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_SEL_16K; break; default: + mutex_unlock(&twl4030->mutex); printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", params_rate(params)); return -EINVAL; @@ -1967,6 +2162,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, twl4030_codec_enable(codec, 1); } + mutex_unlock(&twl4030->mutex); return 0; } @@ -1996,19 +2192,22 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = codec->private_data; + int use256FS = 0; u8 old_format, format; /* get format */ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); - format = old_format; + format = old_format & ~TWL4030_VIF_TRI_EN; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFM: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: format |= TWL4030_VIF_SLAVE_EN; + use256FS = 1; break; default: return -EINVAL; @@ -2033,7 +2232,59 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, twl4030_codec_enable(codec, 1); } - return 0; + return twl4030_set_ext_clock(codec, use256FS | twl4030->extClock); +} + +static int twl4030_clock_startup(struct snd_pcm_substream *substream, + 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 twl4030_priv *twl4030 = codec->private_data; + + return twl4030_new_substream(twl4030, substream, 1); +} + +static int twl4030_clock_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 twl4030_priv *twl4030 = codec->private_data; + int rval; + + mutex_lock(&twl4030->mutex); + + rval = twl4030_set_rate(codec, params); + + /* See if we are a multiple of the current FS. If so, then still OK. */ + if (rval) { + int divisor = twl4030_get_clock_divisor(codec, params); + int clock = params_rate(&twl4030->params) * 256; + int remainder = clock % params_rate(params); + + if (remainder == 0 && divisor <= 256) + rval = 0; + } + + /* If any other streams are currently open, and one of them + * is setting the hw parameters right now (since we are here), set + * constraints to the other stream(s) to match the current one. */ + twl4030_constraints(twl4030); + + mutex_unlock(&twl4030->mutex); + + return rval; +} + +static int twl4030_clock_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + return twl4030_set_ext_clock(codec, 1); } static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) @@ -2056,6 +2307,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .startup = twl4030_startup, .shutdown = twl4030_shutdown, .hw_params = twl4030_hw_params, + .hw_free = twl4030_hw_free, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, .set_tristate = twl4030_set_tristate, @@ -2065,26 +2317,38 @@ static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, + .hw_free = twl4030_hw_free, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, .set_tristate = twl4030_voice_set_tristate, }; +static struct snd_soc_dai_ops twl4030_dai_clock_ops = { + .startup = twl4030_clock_startup, + .shutdown = twl4030_shutdown, + .hw_params = twl4030_clock_hw_params, + .hw_free = twl4030_hw_free, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_clock_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai[] = { { .name = "twl4030", .playback = { .stream_name = "HiFi Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, - .formats = TWL4030_FORMATS,}, + .formats = TWL4030_FORMATS, + }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 4, .rates = TWL4030_RATES, - .formats = TWL4030_FORMATS,}, + .formats = TWL4030_FORMATS, + }, .ops = &twl4030_dai_ops, }, { @@ -2092,17 +2356,37 @@ struct snd_soc_dai twl4030_dai[] = { .playback = { .stream_name = "Voice Playback", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, .ops = &twl4030_dai_voice_ops, }, +{ + .name = "twl4030 Clock", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = SNDRV_PCM_FMTBIT_U8 | TWL4030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = SNDRV_PCM_FMTBIT_U8 | TWL4030_FORMATS, + }, + .ops = &twl4030_dai_clock_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -2220,6 +2504,9 @@ static int twl4030_probe(struct platform_device *pdev) return -ENOMEM; } + mutex_init(&twl4030->mutex); + INIT_LIST_HEAD(&twl4030->started_list); + INIT_LIST_HEAD(&twl4030->config_list); codec->private_data = twl4030; socdev->card->codec = codec; mutex_init(&codec->mutex); @@ -2257,13 +2544,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + return snd_soc_register_dais(twl4030_dai, ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + snd_soc_unregister_dais(twl4030_dai, ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 2b4bfa2..fa23613 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -267,10 +267,15 @@ #define TWL4030_DAI_HIFI 0 #define TWL4030_DAI_VOICE 1 +#define TWL4030_DAI_CLOCK 2 -extern struct snd_soc_dai twl4030_dai[2]; +extern struct snd_soc_dai twl4030_dai[]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; +extern int twl4030_set_rate(struct snd_soc_codec *, struct snd_pcm_hw_params *); +extern int twl4030_get_clock_divisor(struct snd_soc_codec *, + struct snd_pcm_hw_params *); + struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; -- 1.6.0.4

3 3

[alsa-devel] [PATCH 1/8] ASoC: Begin to factor out register cache I/O functions
by Mark Brown 03 Aug '09

03 Aug '09

A lot of CODECs share the same register data formats and therefore replicate the code to manage access to and caching of the register map. In order to reduce code duplication centralised versions of this code will be introduced with drivers able to configure the use of the common code by calling the new snd_soc_codec_set_cache_io() API call during startup. As an initial user the 7 bit address/9 bit data format used by many Wolfson devices is supported for write only CODECs and the drivers with straightforward register cache implementations are converted to use it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 + sound/soc/Makefile | 2 +- sound/soc/codecs/wm8510.c | 141 +++++++++++--------------------- sound/soc/codecs/wm8728.c | 78 +++++------------- sound/soc/codecs/wm8731.c | 108 +++++++------------------ sound/soc/codecs/wm8750.c | 121 ++++++++++------------------ sound/soc/codecs/wm8960.c | 194 +++++++++++++++++---------------------------- sound/soc/codecs/wm8971.c | 121 ++++++++++------------------ sound/soc/codecs/wm8988.c | 105 ++++++++----------------- sound/soc/soc-cache.c | 105 ++++++++++++++++++++++++ 10 files changed, 400 insertions(+), 577 deletions(-) create mode 100644 sound/soc/soc-cache.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 94fcc65..27409dd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,8 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits); #ifdef CONFIG_PM int snd_soc_suspend_device(struct device *dev); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 6f1e28d..4eaf48a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index c8b8dba..7a169bf 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -58,55 +58,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { #define WM8510_POWER1_BIASEN 0x08 #define WM8510_POWER1_BUFIOEN 0x10 -/* - * read wm8510 register cache - */ -static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8510_RESET) - return 0; - if (reg >= WM8510_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8510 register cache - */ -static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8510_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8510 register space - */ -static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8510 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8510_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) +#define wm8510_reset(c) snd_soc_write(c, WM8510_RESET, 0) static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; @@ -327,27 +279,27 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, if (freq_in == 0 || freq_out == 0) { /* Clock CODEC directly from MCLK */ - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); - wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + reg = snd_soc_read(codec, WM8510_CLOCK); + snd_soc_write(codec, WM8510_CLOCK, reg & 0x0ff); /* Turn off PLL */ - reg = wm8510_read_reg_cache(codec, WM8510_POWER1); - wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + reg = snd_soc_read(codec, WM8510_POWER1); + snd_soc_write(codec, WM8510_POWER1, reg & 0x1df); return 0; } pll_factors(freq_out*4, freq_in); - wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); - wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); - wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); - wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); - reg = wm8510_read_reg_cache(codec, WM8510_POWER1); - wm8510_write(codec, WM8510_POWER1, reg | 0x020); + snd_soc_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + snd_soc_write(codec, WM8510_PLLK1, pll_div.k >> 18); + snd_soc_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = snd_soc_read(codec, WM8510_POWER1); + snd_soc_write(codec, WM8510_POWER1, reg | 0x020); /* Run CODEC from PLL instead of MCLK */ - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); - wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + reg = snd_soc_read(codec, WM8510_CLOCK); + snd_soc_write(codec, WM8510_CLOCK, reg | 0x100); return 0; } @@ -363,24 +315,24 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8510_OPCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; - wm8510_write(codec, WM8510_GPIO, reg | div); + reg = snd_soc_read(codec, WM8510_GPIO) & 0x1cf; + snd_soc_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f; - wm8510_write(codec, WM8510_CLOCK, reg | div); + reg = snd_soc_read(codec, WM8510_CLOCK) & 0x11f; + snd_soc_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: - reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; - wm8510_write(codec, WM8510_ADC, reg | div); + reg = snd_soc_read(codec, WM8510_ADC) & 0x1f7; + snd_soc_write(codec, WM8510_ADC, reg | div); break; case WM8510_DACCLK: - reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; - wm8510_write(codec, WM8510_DAC, reg | div); + reg = snd_soc_read(codec, WM8510_DAC) & 0x1f7; + snd_soc_write(codec, WM8510_DAC, reg | div); break; case WM8510_BCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; - wm8510_write(codec, WM8510_CLOCK, reg | div); + reg = snd_soc_read(codec, WM8510_CLOCK) & 0x1e3; + snd_soc_write(codec, WM8510_CLOCK, reg | div); break; default: return -EINVAL; @@ -394,7 +346,7 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; - u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + u16 clk = snd_soc_read(codec, WM8510_CLOCK) & 0x1fe; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -441,8 +393,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8510_write(codec, WM8510_IFACE, iface); - wm8510_write(codec, WM8510_CLOCK, clk); + snd_soc_write(codec, WM8510_IFACE, iface); + snd_soc_write(codec, WM8510_CLOCK, clk); return 0; } @@ -453,8 +405,8 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; - u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; + u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; /* bit size */ switch (params_format(params)) { @@ -493,20 +445,20 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, break; } - wm8510_write(codec, WM8510_IFACE, iface); - wm8510_write(codec, WM8510_ADD, adn); + snd_soc_write(codec, WM8510_IFACE, iface); + snd_soc_write(codec, WM8510_ADD, adn); return 0; } static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + u16 mute_reg = snd_soc_read(codec, WM8510_DAC) & 0xffbf; if (mute) - wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + snd_soc_write(codec, WM8510_DAC, mute_reg | 0x40); else - wm8510_write(codec, WM8510_DAC, mute_reg); + snd_soc_write(codec, WM8510_DAC, mute_reg); return 0; } @@ -514,13 +466,13 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; + u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: power1 |= 0x1; /* VMID 50k */ - wm8510_write(codec, WM8510_POWER1, power1); + snd_soc_write(codec, WM8510_POWER1, power1); break; case SND_SOC_BIAS_STANDBY: @@ -528,18 +480,18 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Initial cap charge at VMID 5k */ - wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); mdelay(100); } power1 |= 0x2; /* VMID 500k */ - wm8510_write(codec, WM8510_POWER1, power1); + snd_soc_write(codec, WM8510_POWER1, power1); break; case SND_SOC_BIAS_OFF: - wm8510_write(codec, WM8510_POWER1, 0); - wm8510_write(codec, WM8510_POWER2, 0); - wm8510_write(codec, WM8510_POWER3, 0); + snd_soc_write(codec, WM8510_POWER1, 0); + snd_soc_write(codec, WM8510_POWER2, 0); + snd_soc_write(codec, WM8510_POWER3, 0); break; } @@ -619,8 +571,6 @@ static int wm8510_init(struct snd_soc_device *socdev) codec->name = "WM8510"; codec->owner = THIS_MODULE; - codec->read = wm8510_read_reg_cache; - codec->write = wm8510_write; codec->set_bias_level = wm8510_set_bias_level; codec->dai = &wm8510_dai; codec->num_dai = 1; @@ -630,13 +580,20 @@ static int wm8510_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", + ret); + goto err; + } + wm8510_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8510: failed to create pcms\n"); - goto pcm_err; + goto err; } /* power on device */ @@ -655,7 +612,7 @@ static int wm8510_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index e7ff212..66da44b 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -43,45 +43,6 @@ static const u16 wm8728_reg_defaults[] = { 0x100, }; -static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); - return cache[reg]; -} - -static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); - cache[reg] = value; -} - -/* - * write to the WM8728 register space - */ -static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8728 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8728_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); static const struct snd_kcontrol_new wm8728_snd_controls[] = { @@ -121,12 +82,12 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec) static int wm8728_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + u16 mute_reg = snd_soc_read(codec, WM8728_DACCTL); if (mute) - wm8728_write(codec, WM8728_DACCTL, mute_reg | 1); + snd_soc_write(codec, WM8728_DACCTL, mute_reg | 1); else - wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1); + snd_soc_write(codec, WM8728_DACCTL, mute_reg & ~1); return 0; } @@ -138,7 +99,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); + u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; @@ -155,7 +116,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8728_write(codec, WM8728_DACCTL, dac); + snd_soc_write(codec, WM8728_DACCTL, dac); return 0; } @@ -164,7 +125,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL); + u16 iface = snd_soc_read(codec, WM8728_IFCTL); /* Currently only I2S is supported by the driver, though the * hardware is more flexible. @@ -204,7 +165,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8728_write(codec, WM8728_IFCTL, iface); + snd_soc_write(codec, WM8728_IFCTL, iface); return 0; } @@ -220,19 +181,19 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Power everything up... */ - reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); - wm8728_write(codec, WM8728_DACCTL, reg & ~0x4); + reg = snd_soc_read(codec, WM8728_DACCTL); + snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - wm8728_write(codec, i, - wm8728_read_reg_cache(codec, i)); + snd_soc_write(codec, i, + snd_soc_read(codec, i)); } break; case SND_SOC_BIAS_OFF: - reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); - wm8728_write(codec, WM8728_DACCTL, reg | 0x4); + reg = snd_soc_read(codec, WM8728_DACCTL); + snd_soc_write(codec, WM8728_DACCTL, reg | 0x4); break; } codec->bias_level = level; @@ -294,8 +255,6 @@ static int wm8728_init(struct snd_soc_device *socdev) codec->name = "WM8728"; codec->owner = THIS_MODULE; - codec->read = wm8728_read_reg_cache; - codec->write = wm8728_write; codec->set_bias_level = wm8728_set_bias_level; codec->dai = &wm8728_dai; codec->num_dai = 1; @@ -307,11 +266,18 @@ static int wm8728_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", + ret); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8728: failed to create pcms\n"); - goto pcm_err; + goto err; } /* power on device */ @@ -331,7 +297,7 @@ static int wm8728_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index dfbc1bb..4eb84ff 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -50,60 +50,12 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); * There is no point in caching the reset register */ static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 + 0x0097, 0x0097, 0x0079, 0x0079, + 0x000a, 0x0008, 0x009f, 0x000a, + 0x0000, 0x0000 }; -/* - * read wm8731 register cache - */ -static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8731_RESET) - return 0; - if (reg >= WM8731_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8731 register cache - */ -static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8731_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8731 register space - */ -static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8731 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8731_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0) +#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -260,12 +212,12 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8731_priv *wm8731 = codec->private_data; - u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3; + u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; int i = get_coeff(wm8731->sysclk, params_rate(params)); u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; - wm8731_write(codec, WM8731_SRATE, srate); + snd_soc_write(codec, WM8731_SRATE, srate); /* bit size */ switch (params_format(params)) { @@ -279,7 +231,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, break; } - wm8731_write(codec, WM8731_IFACE, iface); + snd_soc_write(codec, WM8731_IFACE, iface); return 0; } @@ -291,7 +243,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; /* set active */ - wm8731_write(codec, WM8731_ACTIVE, 0x0001); + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); return 0; } @@ -306,19 +258,19 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - wm8731_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); } } static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8731_APDIGI) & 0xfff7; if (mute) - wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8); + snd_soc_write(codec, WM8731_APDIGI, mute_reg | 0x8); else - wm8731_write(codec, WM8731_APDIGI, mute_reg); + snd_soc_write(codec, WM8731_APDIGI, mute_reg); return 0; } @@ -396,7 +348,7 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8731_write(codec, WM8731_IFACE, iface); + snd_soc_write(codec, WM8731_IFACE, iface); return 0; } @@ -412,12 +364,12 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Clear PWROFF, gate CLKOUT, everything else as-is */ - reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - wm8731_write(codec, WM8731_PWR, reg | 0x0040); + reg = snd_soc_read(codec, WM8731_PWR) & 0xff7f; + snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_write(codec, WM8731_PWR, 0xffff); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_PWR, 0xffff); break; } codec->bias_level = level; @@ -466,7 +418,7 @@ static int wm8731_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; - wm8731_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -556,7 +508,6 @@ static int wm8731_register(struct wm8731_priv *wm8731) { int ret; struct snd_soc_codec *codec = &wm8731->codec; - u16 reg; if (wm8731_codec) { dev_err(codec->dev, "Another WM8731 is registered\n"); @@ -571,8 +522,6 @@ static int wm8731_register(struct wm8731_priv *wm8731) codec->private_data = wm8731; codec->name = "WM8731"; codec->owner = THIS_MODULE; - codec->read = wm8731_read_reg_cache; - codec->write = wm8731_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; @@ -582,6 +531,12 @@ static int wm8731_register(struct wm8731_priv *wm8731) memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8731_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); @@ -593,18 +548,13 @@ static int wm8731_register(struct wm8731_priv *wm8731) wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); + snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); + snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); + snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); + snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - reg = wm8731_read_reg_cache(codec, WM8731_APANA); - wm8731_write(codec, WM8731_APANA, reg & ~0x4); + snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0); wm8731_codec = codec; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b64509b..ed09043 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -55,50 +55,7 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; -/* - * read wm8750 register cache - */ -static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8750_CACHE_REGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8750 register cache - */ -static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8750_CACHE_REGNUM) - return; - cache[reg] = value; -} - -static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8750_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0) +#define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* * WM8750 Controls @@ -594,7 +551,7 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8750_write(codec, WM8750_IFACE, iface); + snd_soc_write(codec, WM8750_IFACE, iface); return 0; } @@ -606,8 +563,8 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8750_priv *wm8750 = codec->private_data; - u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3; - u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0; + u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; int coeff = get_coeff(wm8750->sysclk, params_rate(params)); /* bit size */ @@ -626,9 +583,9 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8750_write(codec, WM8750_IFACE, iface); + snd_soc_write(codec, WM8750_IFACE, iface); if (coeff >= 0) - wm8750_write(codec, WM8750_SRATE, srate | + snd_soc_write(codec, WM8750_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -637,35 +594,35 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8750_ADCDAC) & 0xfff7; if (mute) - wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8750_ADCDAC, mute_reg | 0x8); else - wm8750_write(codec, WM8750_ADCDAC, mute_reg); + snd_soc_write(codec, WM8750_ADCDAC, mute_reg); return 0; } static int wm8750_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; + u16 pwr_reg = snd_soc_read(codec, WM8750_PWR1) & 0xfe3e; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; case SND_SOC_BIAS_OFF: - wm8750_write(codec, WM8750_PWR1, 0x0001); + snd_soc_write(codec, WM8750_PWR1, 0x0001); break; } codec->bias_level = level; @@ -761,8 +718,6 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->name = "WM8750"; codec->owner = THIS_MODULE; - codec->read = wm8750_read_reg_cache; - codec->write = wm8750_write; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; @@ -771,13 +726,23 @@ static int wm8750_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; - wm8750_reset(codec); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); + goto err; + } + + ret = wm8750_reset(codec); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to reset: %d\n", ret); + goto err; + } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto pcm_err; + goto err; } /* charge output caps */ @@ -786,22 +751,22 @@ static int wm8750_init(struct snd_soc_device *socdev) schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ - reg = wm8750_read_reg_cache(codec, WM8750_LDAC); - wm8750_write(codec, WM8750_LDAC, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_RDAC); - wm8750_write(codec, WM8750_RDAC, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V); - wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V); - wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V); - wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V); - wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LINVOL); - wm8750_write(codec, WM8750_LINVOL, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); - wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LDAC); + snd_soc_write(codec, WM8750_LDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_RDAC); + snd_soc_write(codec, WM8750_RDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LOUT1V); + snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_ROUT1V); + snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LOUT2V); + snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_ROUT2V); + snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LINVOL); + snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_RINVOL); + snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); @@ -816,7 +781,7 @@ static int wm8750_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d1769e6..c529ffc 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -69,61 +69,7 @@ struct wm8960_priv { struct snd_soc_codec codec; }; -/* - * read wm8960 register cache - */ -static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8960_RESET) - return 0; - if (reg >= WM8960_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8960 register cache - */ -static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8960_CACHEREGNUM) - return; - cache[reg] = value; -} - -static inline unsigned int wm8960_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return wm8960_read_reg_cache(codec, reg); -} - -/* - * write to the WM8960 register space - */ -static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8960 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8960_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0) +#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) /* enumerated controls */ static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -420,7 +366,7 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8960_write(codec, WM8960_IFACE1, iface); + snd_soc_write(codec, WM8960_IFACE1, iface); return 0; } @@ -431,7 +377,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3; + u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; /* bit size */ switch (params_format(params)) { @@ -446,19 +392,19 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, } /* set iface */ - wm8960_write(codec, WM8960_IFACE1, iface); + snd_soc_write(codec, WM8960_IFACE1, iface); return 0; } static int wm8960_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7; if (mute) - wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8); else - wm8960_write(codec, WM8960_DACCTL1, mute_reg); + snd_soc_write(codec, WM8960_DACCTL1, mute_reg); return 0; } @@ -474,16 +420,16 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg &= ~0x180; reg |= 0x80; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, + snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); @@ -491,43 +437,43 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, reg = WM8960_DISOP; if (pdata) reg |= pdata->dres << 4; - wm8960_write(codec, WM8960_APOP2, reg); + snd_soc_write(codec, WM8960_APOP2, reg); msleep(400); - wm8960_write(codec, WM8960_APOP2, 0); + snd_soc_write(codec, WM8960_APOP2, 0); /* Enable & ramp VMID at 2x50k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg |= 0x80; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); msleep(100); /* Enable VREF */ - wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF); + snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF); /* Disable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN); + snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN); } /* Set VMID to 2x250k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg &= ~0x180; reg |= 0x100; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); break; case SND_SOC_BIAS_OFF: /* Enable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, + snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Disable VMID and VREF, let them discharge */ - wm8960_write(codec, WM8960_POWER1, 0); + snd_soc_write(codec, WM8960_POWER1, 0); msleep(600); - wm8960_write(codec, WM8960_APOP1, 0); + snd_soc_write(codec, WM8960_APOP1, 0); break; } @@ -610,33 +556,33 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, /* Disable the PLL: even if we are changing the frequency the * PLL needs to be disabled while we do so. */ - wm8960_write(codec, WM8960_CLOCK1, - wm8960_read(codec, WM8960_CLOCK1) & ~1); - wm8960_write(codec, WM8960_POWER2, - wm8960_read(codec, WM8960_POWER2) & ~1); + snd_soc_write(codec, WM8960_CLOCK1, + snd_soc_read(codec, WM8960_CLOCK1) & ~1); + snd_soc_write(codec, WM8960_POWER2, + snd_soc_read(codec, WM8960_POWER2) & ~1); if (!freq_in || !freq_out) return 0; - reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f; + reg = snd_soc_read(codec, WM8960_PLL1) & ~0x3f; reg |= pll_div.pre_div << 4; reg |= pll_div.n; if (pll_div.k) { reg |= 0x20; - wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); } - wm8960_write(codec, WM8960_PLL1, reg); + snd_soc_write(codec, WM8960_PLL1, reg); /* Turn it on */ - wm8960_write(codec, WM8960_POWER2, - wm8960_read(codec, WM8960_POWER2) | 1); + snd_soc_write(codec, WM8960_POWER2, + snd_soc_read(codec, WM8960_POWER2) | 1); msleep(250); - wm8960_write(codec, WM8960_CLOCK1, - wm8960_read(codec, WM8960_CLOCK1) | 1); + snd_soc_write(codec, WM8960_CLOCK1, + snd_soc_read(codec, WM8960_CLOCK1) | 1); return 0; } @@ -649,28 +595,28 @@ static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8960_SYSCLKSEL: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1fe; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_SYSCLKDIV: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1f9; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_DACDIV: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1c7; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_OPCLKDIV: - reg = wm8960_read(codec, WM8960_PLL1) & 0x03f; - wm8960_write(codec, WM8960_PLL1, reg | div); + reg = snd_soc_read(codec, WM8960_PLL1) & 0x03f; + snd_soc_write(codec, WM8960_PLL1, reg | div); break; case WM8960_DCLKDIV: - reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f; - wm8960_write(codec, WM8960_CLOCK2, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK2) & 0x03f; + snd_soc_write(codec, WM8960_CLOCK2, reg | div); break; case WM8960_TOCLKSEL: - reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd; - wm8960_write(codec, WM8960_ADDCTL1, reg | div); + reg = snd_soc_read(codec, WM8960_ADDCTL1) & 0x1fd; + snd_soc_write(codec, WM8960_ADDCTL1, reg | div); break; default: return -EINVAL; @@ -830,8 +776,6 @@ static int wm8960_register(struct wm8960_priv *wm8960) codec->private_data = wm8960; codec->name = "WM8960"; codec->owner = THIS_MODULE; - codec->read = wm8960_read_reg_cache; - codec->write = wm8960_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8960_set_bias_level; codec->dai = &wm8960_dai; @@ -841,6 +785,12 @@ static int wm8960_register(struct wm8960_priv *wm8960) memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8960_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); @@ -852,26 +802,26 @@ static int wm8960_register(struct wm8960_priv *wm8960) wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8960_read(codec, WM8960_LINVOL); - wm8960_write(codec, WM8960_LINVOL, reg | 0x100); - reg = wm8960_read(codec, WM8960_RINVOL); - wm8960_write(codec, WM8960_RINVOL, reg | 0x100); - reg = wm8960_read(codec, WM8960_LADC); - wm8960_write(codec, WM8960_LADC, reg | 0x100); - reg = wm8960_read(codec, WM8960_RADC); - wm8960_write(codec, WM8960_RADC, reg | 0x100); - reg = wm8960_read(codec, WM8960_LDAC); - wm8960_write(codec, WM8960_LDAC, reg | 0x100); - reg = wm8960_read(codec, WM8960_RDAC); - wm8960_write(codec, WM8960_RDAC, reg | 0x100); - reg = wm8960_read(codec, WM8960_LOUT1); - wm8960_write(codec, WM8960_LOUT1, reg | 0x100); - reg = wm8960_read(codec, WM8960_ROUT1); - wm8960_write(codec, WM8960_ROUT1, reg | 0x100); - reg = wm8960_read(codec, WM8960_LOUT2); - wm8960_write(codec, WM8960_LOUT2, reg | 0x100); - reg = wm8960_read(codec, WM8960_ROUT2); - wm8960_write(codec, WM8960_ROUT2, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LINVOL); + snd_soc_write(codec, WM8960_LINVOL, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RINVOL); + snd_soc_write(codec, WM8960_RINVOL, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LADC); + snd_soc_write(codec, WM8960_LADC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RADC); + snd_soc_write(codec, WM8960_RADC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LDAC); + snd_soc_write(codec, WM8960_LDAC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RDAC); + snd_soc_write(codec, WM8960_RDAC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LOUT1); + snd_soc_write(codec, WM8960_LOUT1, reg | 0x100); + reg = snd_soc_read(codec, WM8960_ROUT1); + snd_soc_write(codec, WM8960_ROUT1, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LOUT2); + snd_soc_write(codec, WM8960_LOUT2, reg | 0x100); + reg = snd_soc_read(codec, WM8960_ROUT2); + snd_soc_write(codec, WM8960_ROUT2, reg | 0x100); wm8960_codec = codec; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 032dca2..53f3bc9 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -59,44 +59,7 @@ static const u16 wm8971_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; -static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg < WM8971_REG_COUNT) - return cache[reg]; - - return -1; -} - -static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg < WM8971_REG_COUNT) - cache[reg] = value; -} - -static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8971_write_reg_cache (codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0) +#define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; @@ -521,7 +484,7 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8971_write(codec, WM8971_IFACE, iface); + snd_soc_write(codec, WM8971_IFACE, iface); return 0; } @@ -533,8 +496,8 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8971_priv *wm8971 = codec->private_data; - u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3; - u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0; + u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; int coeff = get_coeff(wm8971->sysclk, params_rate(params)); /* bit size */ @@ -553,9 +516,9 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8971_write(codec, WM8971_IFACE, iface); + snd_soc_write(codec, WM8971_IFACE, iface); if (coeff >= 0) - wm8971_write(codec, WM8971_SRATE, srate | + snd_soc_write(codec, WM8971_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -564,33 +527,33 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8971_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8971_ADCDAC) & 0xfff7; if (mute) - wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8971_ADCDAC, mute_reg | 0x8); else - wm8971_write(codec, WM8971_ADCDAC, mute_reg); + snd_soc_write(codec, WM8971_ADCDAC, mute_reg); return 0; } static int wm8971_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; + u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ - wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); break; case SND_SOC_BIAS_OFF: - wm8971_write(codec, WM8971_PWR1, 0x0001); + snd_soc_write(codec, WM8971_PWR1, 0x0001); break; } codec->bias_level = level; @@ -667,8 +630,8 @@ static int wm8971_resume(struct platform_device *pdev) /* charge wm8971 caps */ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { - reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; - wm8971_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->delayed_work, msecs_to_jiffies(1000)); @@ -684,8 +647,6 @@ static int wm8971_init(struct snd_soc_device *socdev) codec->name = "WM8971"; codec->owner = THIS_MODULE; - codec->read = wm8971_read_reg_cache; - codec->write = wm8971_write; codec->set_bias_level = wm8971_set_bias_level; codec->dai = &wm8971_dai; codec->reg_cache_size = ARRAY_SIZE(wm8971_reg); @@ -695,42 +656,48 @@ static int wm8971_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); + goto err; + } + wm8971_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8971: failed to create pcms\n"); - goto pcm_err; + goto err; } /* charge output caps - set vmid to 5k for quick power up */ - reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; - wm8971_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ - reg = wm8971_read_reg_cache(codec, WM8971_LDAC); - wm8971_write(codec, WM8971_LDAC, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_RDAC); - wm8971_write(codec, WM8971_RDAC, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V); - wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V); - wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V); - wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V); - wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LINVOL); - wm8971_write(codec, WM8971_LINVOL, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); - wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_LDAC); + snd_soc_write(codec, WM8971_LDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_RDAC); + snd_soc_write(codec, WM8971_RDAC, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LOUT1V); + snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_ROUT1V); + snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LOUT2V); + snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_ROUT2V); + snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LINVOL); + snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_RINVOL); + snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100); snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); @@ -745,7 +712,7 @@ static int wm8971_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6f15acd..7d5b807 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -57,50 +57,7 @@ struct wm8988_priv { }; -/* - * read wm8988 register cache - */ -static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8988_NUM_REG) - return -1; - return cache[reg]; -} - -/* - * write wm8988 register cache - */ -static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8988_NUM_REG) - return; - cache[reg] = value; -} - -static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8988_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0) +#define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) /* * WM8988 Controls @@ -226,15 +183,15 @@ static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2); + u16 adctl2 = snd_soc_read(codec, WM8988_ADCTL2); /* Use the DAC to gate LRC if active, otherwise use ADC */ - if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180) + if (snd_soc_read(codec, WM8988_PWR2) & 0x180) adctl2 &= ~0x4; else adctl2 |= 0x4; - return wm8988_write(codec, WM8988_ADCTL2, adctl2); + return snd_soc_write(codec, WM8988_ADCTL2, adctl2); } static const char *wm8988_line_texts[] = { @@ -619,7 +576,7 @@ static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8988_write(codec, WM8988_IFACE, iface); + snd_soc_write(codec, WM8988_IFACE, iface); return 0; } @@ -653,8 +610,8 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8988_priv *wm8988 = codec->private_data; - u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3; - u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180; + u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; int coeff; coeff = get_coeff(wm8988->sysclk, params_rate(params)); @@ -685,9 +642,9 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8988_write(codec, WM8988_IFACE, iface); + snd_soc_write(codec, WM8988_IFACE, iface); if (coeff >= 0) - wm8988_write(codec, WM8988_SRATE, srate | + snd_soc_write(codec, WM8988_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -696,19 +653,19 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8988_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8988_ADCDAC) & 0xfff7; if (mute) - wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8988_ADCDAC, mute_reg | 0x8); else - wm8988_write(codec, WM8988_ADCDAC, mute_reg); + snd_soc_write(codec, WM8988_ADCDAC, mute_reg); return 0; } static int wm8988_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1; + u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1; switch (level) { case SND_SOC_BIAS_ON: @@ -716,24 +673,24 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VREF, VMID=2x50k, digital enabled */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* VREF, VMID=2x5k */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); /* Charge caps */ msleep(100); } /* VREF, VMID=2*500k, digital stopped */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x0141); break; case SND_SOC_BIAS_OFF: - wm8988_write(codec, WM8988_PWR1, 0x0000); + snd_soc_write(codec, WM8988_PWR1, 0x0000); break; } codec->bias_level = level; @@ -887,8 +844,6 @@ static int wm8988_register(struct wm8988_priv *wm8988) codec->private_data = wm8988; codec->name = "WM8988"; codec->owner = THIS_MODULE; - codec->read = wm8988_read_reg_cache; - codec->write = wm8988_write; codec->dai = &wm8988_dai; codec->num_dai = 1; codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache); @@ -899,6 +854,12 @@ static int wm8988_register(struct wm8988_priv *wm8988) memcpy(codec->reg_cache, wm8988_reg, sizeof(wm8988_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8988_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); @@ -906,16 +867,16 @@ static int wm8988_register(struct wm8988_priv *wm8988) } /* set the update bits (we always update left then right) */ - reg = wm8988_read_reg_cache(codec, WM8988_RADC); - wm8988_write(codec, WM8988_RADC, reg | 0x100); - reg = wm8988_read_reg_cache(codec, WM8988_RDAC); - wm8988_write(codec, WM8988_RDAC, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V); - wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V); - wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_RINVOL); - wm8988_write(codec, WM8988_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_RADC); + snd_soc_write(codec, WM8988_RADC, reg | 0x100); + reg = snd_soc_read(codec, WM8988_RDAC); + snd_soc_write(codec, WM8988_RDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_ROUT1V); + snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_ROUT2V); + snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_RINVOL); + snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100); wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c new file mode 100644 index 0000000..4eb4333 --- /dev/null +++ b/sound/soc/soc-cache.c @@ -0,0 +1,105 @@ +/* + * soc-cache.c -- ASoC register cache helpers + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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 <sound/soc.h> + +static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + + +static struct { + int addr_bits; + int data_bits; + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + unsigned int (*read)(struct snd_soc_codec *, unsigned int); +} io_types[] = { + { 7, 9, snd_soc_7_9_write, snd_soc_7_9_read }, +}; + +/** + * snd_soc_codec_set_cache_io: Set up standard I/O functions. + * + * @codec: CODEC to configure. + * @type: Type of cache. + * @addr_bits: Number of bits of register address data. + * @data_bits: Number of bits of data per register. + * + * Register formats are frequently shared between many I2C and SPI + * devices. In order to promote code reuse the ASoC core provides + * some standard implementations of CODEC read and write operations + * which can be set up using this function. + * + * The caller is responsible for allocating and initialising the + * actual cache. + * + * Note that at present this code cannot be used by CODECs with + * volatile registers. + */ +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits) +{ + int i; + + /* We don't support volatile registers yet - refactoring of + * the hw_read operation will be required to do so. */ + if (codec->volatile_register) { + printk(KERN_ERR "Volatile registers not yet supported\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(io_types); i++) + if (io_types[i].addr_bits == addr_bits && + io_types[i].data_bits == data_bits) + break; + if (i == ARRAY_SIZE(io_types)) { + printk(KERN_ERR + "No I/O functions for %d bit address %d bit data\n", + addr_bits, data_bits); + return -EINVAL; + } + + codec->write = io_types[i].write; + codec->read = io_types[i].read; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); -- 1.6.3.3

1 7

[alsa-devel] [PATCH 1/4] pulse: get rid of a number of assert()s
by Lennart Poettering 03 Aug '09

03 Aug '09

Instead of hitting an assert when any of the plugin functions is called in an invalid context we should return a clean error to make sure programs are not unnecessarily aborted. This should fix issues such as http://pulseaudio.org/ticket/595 --- pulse/ctl_pulse.c | 35 +++++++++++++++++------ pulse/pcm_pulse.c | 79 +++++++++++++++++++++++++++++++++++++++++------------ pulse/pulse.c | 34 ++++++++++++++++------- 3 files changed, 111 insertions(+), 37 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index c6cf9e2..1b057ef 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -125,8 +125,9 @@ static void event_cb(pa_context * c, pa_subscription_event_type_t t, pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -148,8 +149,9 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -203,6 +205,9 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -260,7 +265,9 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, return -EINVAL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -311,7 +318,9 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -361,7 +370,9 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p && ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -465,6 +476,9 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; + pa_threaded_mainloop_lock(ctl->p->mainloop); ctl->subscribed = !!(subscribe & SND_CTL_EVENT_MASK_VALUE); @@ -481,6 +495,9 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); if (!ctl->updated || !ctl->subscribed) @@ -525,8 +542,8 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, snd_ctl_pulse_t *ctl = ext->private_data; int err = 0; - assert(ctl); - assert(ctl->p); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index c276839..24347f9 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -106,6 +106,9 @@ static int update_active(snd_pcm_pulse_t *pcm) { assert(pcm); + if (!pcm->p) + return -EBADFD; + ret = check_active(pcm); if (ret < 0) return ret; @@ -125,7 +128,9 @@ static int pulse_start(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -174,7 +179,9 @@ static int pulse_stop(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -224,7 +231,9 @@ static int pulse_drain(snd_pcm_ioplug_t * io) int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -259,7 +268,9 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) return -EPIPE; @@ -269,7 +280,10 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -305,11 +319,16 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) pa_usec_t lat = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + err = -EBADFD; + goto finish; + } for (;;) { err = pulse_check_connection(pcm->p); @@ -354,11 +373,16 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -409,11 +433,16 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -480,7 +509,9 @@ static void stream_request_cb(pa_stream * p, size_t length, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; update_active(pcm); } @@ -490,7 +521,9 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pcm->underrun = 1; } @@ -499,7 +532,9 @@ static void stream_latency_cb(pa_stream *p, void *userdata) { snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pa_threaded_mainloop_signal(pcm->p->mainloop, 0); } @@ -512,7 +547,9 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, snd_pcm_pulse_t *pcm = io->private_data; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -541,7 +578,9 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) unsigned c, d; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -645,7 +684,9 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -745,7 +786,9 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) int err = 0; assert (pcm); - assert (pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); diff --git a/pulse/pulse.c b/pulse/pulse.c index 3940238..95d8dde 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -32,8 +32,9 @@ int pulse_check_connection(snd_pulse_t * p) pa_context_state_t state; assert(p); - assert(p->context); - assert(p->mainloop); + + if (!p->context || !p->mainloop) + return -EBADFD; state = pa_context_get_state(p->context); @@ -77,8 +78,12 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) { assert(p); assert(o); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -103,8 +108,12 @@ int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, assert(p); assert(stream); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -197,7 +206,9 @@ snd_pulse_t *pulse_new(void) p->context = pa_context_new(pa_threaded_mainloop_get_api(p->mainloop), buf); - assert(p->context); + + if (!p->context) + goto fail; pa_context_set_state_callback(p->context, context_state_cb, p); @@ -246,9 +257,12 @@ int pulse_connect(snd_pulse_t * p, const char *server) int err; assert(p); - assert(p->context); - assert(p->mainloop); - assert(p->state == PULSE_STATE_INIT); + + if (!p->context || !p->mainloop) + return -EBADFD; + + if (p->state != PULSE_STATE_INIT) + return -EBADFD; pa_threaded_mainloop_lock(p->mainloop); -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] ASoC: DMA and SSI drivers for i.MX27
by javier Martin 03 Aug '09

03 Aug '09

As I described in one of my previous mails, I have been trying to port an old ASoC driver for kernel 2.6.22 to latest stable 2.6.30 kernel. I am currently facing a big problem which make some repeated samples appear with a regular pattern: - frequency of bad samples = 2 * period_size - bad samples in audio stream are taken from "buffer_size" samples back in the original file. You can reproduce this with: aplay -t raw -f S16_LE -r8000 8k16bitpcm.wav (http://en.wikipedia.org/wiki/WAV to get this file) Here I submit two patches: - One which adds ssi and dma drivers plus a custom machine driver (imx27asoc_list.patch). - Another patch which adds wm8974 codec support taken from ASoC wm8974-upstream branch (wm8974_list.patch). Note that the machine driver attached uses wm8974 codec, thus second patch is needed. Please, note also that patches are currently quite dirty, and in an early development state. The purpose of this mail is not to even consider to include them in any branch right now, but to gather some help to solve the problem. As soon as I have it solved I will clean all the stuff and try to prepare it for mainline. -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

1 1

[alsa-devel] ALSA -LADSPA buffer
by arif setiawan 03 Aug '09

03 Aug '09

Hi all, I have some questions about ALSA - LADSPA data exchange in ALSA plugin. As I understand it, LADSPA only use 32 bit floating point, while ALSA can use many types of buffer, S8, S16_LE and so on. My question is how this buffer converted? As I work with OMAP processor I want to use DSP for signal processing. We want the LADSPA plugin connect to the DSP and get plugin advantage. But DSP only work on fixed point arithmatic, so we need to convert the 32 bit float in LADSPA to an integer type buffer. I think I can achieve it by following ALSA-LADSPA buffer conversion. How I can do this? Thank you for your help. Best regards, Arif

1 0

[alsa-devel] omap34xx mixer elements
by Venkam Bala krishna 03 Aug '09

03 Aug '09

Hi all Thanks in advance, I am newbie to alsa Currently working on porting android to omap34xx Right now porting alsa. This is a general question. Please check the below log and specify any suggestions, why is it not listing any of my capture devices. What should be done to recognise my capture devices? Do i have to do it in kernel level or user level?(since my amixer is listing all my elements, given below the log is my amixer output for verification). Any suggestions are welcome thanks once agiain, log ======================== D/AudioHardwareALSA( 799): Mixer: element name: 'Carkit' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Analog Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Digital Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC1 Analog' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC1 Digital Coarse' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC1 Digital Fine' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC2 Analog' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC2 Digital Coarse' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC2 Digital Fine' D/AudioHardwareALSA( 799): Mixer: element name: 'Earpiece' D/AudioHardwareALSA( 799): Mixer: element name: 'Headset' D/AudioHardwareALSA( 799): Mixer: element name: 'Left Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'PreDriv' D/AudioHardwareALSA( 799): Mixer: element name: 'Right Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'Voice Digital Loopback' D/AudioHardwareALSA( 799): Mixer: master 'PCM' not found. D/AudioHardwareALSA( 799): Mixer: route 'Earpiece' found. D/AudioHardwareALSA( 799): Mixer: route 'Speaker' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth' not found. D/AudioHardwareALSA( 799): Mixer: route 'Headphone' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth A2DP' not found. D/AudioHardwareALSA( 799): Mixer: element name: 'Analog' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Analog Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Digital Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'Left Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'Right Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'TX1 Digital' D/AudioHardwareALSA( 799): Mixer: element name: 'TX2 Digital' D/AudioHardwareALSA( 799): Mixer: element name: 'Voice Digital Loopback' D/AudioHardwareALSA( 799): Mixer: master 'Capture' not found. D/AudioHardwareALSA( 799): Mixer: route 'Capture' not found. D/AudioHardwareALSA( 799): Mixer: route '' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth Capture' not found. D/AudioHardwareALSA( 799): Mixer: route 'Capture' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth A2DP Capture' not found. D/AudioHardwareALSA( 799): mixer initialized. =========================== amixer output ================= numid=18,iface=MIXER,name='Analog Capture Volume' numid=23,iface=MIXER,name='Analog Left Capture Route' numid=22,iface=MIXER,name='Analog Right Capture Route' numid=14,iface=MIXER,name='Carkit Playback Volume' numid=34,iface=MIXER,name='CarkitL Mixer AudioL1' numid=35,iface=MIXER,name='CarkitL Mixer AudioL2' numid=33,iface=MIXER,name='CarkitL Mixer Voice' numid=31,iface=MIXER,name='CarkitR Mixer AudioR1' numid=32,iface=MIXER,name='CarkitR Mixer AudioR2' numid=30,iface=MIXER,name='CarkitR Mixer Voice' numid=11,iface=MIXER,name='DAC Voice Analog Downlink Switch' numid=10,iface=MIXER,name='DAC Voice Analog Downlink Volume' numid=9,iface=MIXER,name='DAC Voice Digital Downlink Volume' numid=7,iface=MIXER,name='DAC1 Analog Playback Switch' numid=5,iface=MIXER,name='DAC1 Analog Playback Volume' numid=3,iface=MIXER,name='DAC1 Digital Coarse Playback Volume' numid=1,iface=MIXER,name='DAC1 Digital Fine Playback Volume' numid=8,iface=MIXER,name='DAC2 Analog Playback Switch' numid=6,iface=MIXER,name='DAC2 Analog Playback Volume' numid=4,iface=MIXER,name='DAC2 Digital Coarse Playback Volume' numid=2,iface=MIXER,name='DAC2 Digital Fine Playback Volume' numid=51,iface=MIXER,name='Earpiece Mixer AudioL1' numid=52,iface=MIXER,name='Earpiece Mixer AudioL2' numid=53,iface=MIXER,name='Earpiece Mixer AudioR1' numid=50,iface=MIXER,name='Earpiece Mixer Voice' numid=15,iface=MIXER,name='Earpiece Playback Volume' numid=19,iface=MIXER,name='HS ramp delay' numid=29,iface=MIXER,name='HandsfreeL Mux' numid=28,iface=MIXER,name='HandsfreeR Mux' numid=13,iface=MIXER,name='Headset Playback Volume' numid=40,iface=MIXER,name='HeadsetL Mixer AudioL1' numid=41,iface=MIXER,name='HeadsetL Mixer AudioL2' numid=39,iface=MIXER,name='HeadsetL Mixer Voice' numid=37,iface=MIXER,name='HeadsetR Mixer AudioR1' numid=38,iface=MIXER,name='HeadsetR Mixer AudioR2' numid=36,iface=MIXER,name='HeadsetR Mixer Voice' numid=56,iface=MIXER,name='Left Digital Loopback Volume' numid=60,iface=MIXER,name='Left1 Analog Loopback Switch' numid=58,iface=MIXER,name='Left2 Analog Loopback Switch' numid=12,iface=MIXER,name='PreDriv Playback Volume' numid=47,iface=MIXER,name='PredriveL Mixer AudioL1' numid=48,iface=MIXER,name='PredriveL Mixer AudioL2' numid=49,iface=MIXER,name='PredriveL Mixer AudioR2' numid=46,iface=MIXER,name='PredriveL Mixer Voice' numid=45,iface=MIXER,name='PredriveR Mixer AudioL2' numid=43,iface=MIXER,name='PredriveR Mixer AudioR1' numid=44,iface=MIXER,name='PredriveR Mixer AudioR2' numid=42,iface=MIXER,name='PredriveR Mixer Voice' numid=55,iface=MIXER,name='Right Digital Loopback Volume' numid=61,iface=MIXER,name='Right1 Analog Loopback Switch' numid=59,iface=MIXER,name='Right2 Analog Loopback Switch' numid=25,iface=MIXER,name='TX1 Capture Route' numid=16,iface=MIXER,name='TX1 Digital Capture Volume' numid=24,iface=MIXER,name='TX2 Capture Route' numid=17,iface=MIXER,name='TX2 Digital Capture Volume' numid=21,iface=MIXER,name='Vibra H-bridge direction' numid=20,iface=MIXER,name='Vibra H-bridge mode' numid=27,iface=MIXER,name='Vibra Mux' numid=26,iface=MIXER,name='Vibra Route' numid=57,iface=MIXER,name='Voice Analog Loopback Switch' numid=54,iface=MIXER,name='Voice Digital Loopback Volume' =================== -- Regards Venkam Balakrishna

2 1

[alsa-devel] [PATCH] hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin number
by Wu Fengguang 03 Aug '09

03 Aug '09

The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes. Here we assume only the first ones will be used. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_intelhdmi.c | 104 +++++++++++++++--------------- 1 file changed, 52 insertions(+), 52 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -33,8 +33,8 @@ #include "hda_codec.h" #include "hda_local.h" -#define CVT_NID 0x02 /* audio converter */ -#define PIN_NID 0x03 /* HDMI output pin */ +static hda_nid_t cvt_nid; /* audio converter */ +static hda_nid_t pin_nid; /* HDMI output pin */ #define INTEL_HDMI_EVENT_TAG 0x08 @@ -44,30 +44,6 @@ struct intel_hdmi_spec { struct hdmi_eld sink_eld; }; -static struct hda_verb pinout_enable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {} /* terminator */ -}; - -static struct hda_verb unsolicited_response_verb[] = { - {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | - INTEL_HDMI_EVENT_TAG}, - {} -}; - -static struct hda_verb def_chan_map[] = { - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77}, - {} -}; - - struct hdmi_audio_infoframe { u8 type; /* 0x84 */ u8 ver; /* 0x01 */ @@ -244,11 +220,12 @@ static void hdmi_write_dip_byte(struct h static void hdmi_enable_output(struct hda_codec *codec) { /* Unmute */ - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); /* Enable pin out */ - snd_hda_sequence_write(codec, pinout_enable_verb); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); } /* @@ -256,8 +233,8 @@ static void hdmi_enable_output(struct hd */ static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); } @@ -266,20 +243,20 @@ static void hdmi_start_infoframe_trans(s */ static void hdmi_stop_infoframe_trans(struct hda_codec *codec) { - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) { - return 1 + snd_hda_codec_read(codec, CVT_NID, 0, + return 1 + snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } static void hdmi_set_channel_count(struct hda_codec *codec, int chs) { - snd_hda_codec_write(codec, CVT_NID, 0, + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); if (chs != hdmi_get_channel_count(codec)) @@ -294,7 +271,7 @@ static void hdmi_debug_channel_mapping(s int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, CVT_NID, 0, + slot = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0x7); @@ -307,7 +284,7 @@ static void hdmi_parse_eld(struct hda_co struct intel_hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->sink_eld; - if (!snd_hdmi_get_eld(eld, codec, PIN_NID)) + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld); } @@ -322,11 +299,11 @@ static void hdmi_debug_dip_size(struct h int i; int size; - size = snd_hdmi_get_eld_size(codec, PIN_NID); + size = snd_hdmi_get_eld_size(codec, pin_nid); printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, PIN_NID, 0, + size = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, i); printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); } @@ -340,15 +317,15 @@ static void hdmi_clear_dip_buffers(struc int size; int pi, bi; for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, PIN_NID, 0, + size = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, i); if (size == 0) continue; - hdmi_set_dip_index(codec, PIN_NID, i, 0x0); + hdmi_set_dip_index(codec, pin_nid, i, 0x0); for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, PIN_NID, 0x0); - hdmi_get_dip_index(codec, PIN_NID, &pi, &bi); + hdmi_write_dip_byte(codec, pin_nid, 0x0); + hdmi_get_dip_index(codec, pin_nid, &pi, &bi); if (pi != i) snd_printd(KERN_INFO "dip index %d: %d != %d\n", bi, pi, i); @@ -376,9 +353,9 @@ static void hdmi_fill_audio_infoframe(st sum += params[i]; ai->checksum = - sum; - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < sizeof(ai); i++) - hdmi_write_dip_byte(codec, PIN_NID, params[i]); + hdmi_write_dip_byte(codec, pin_nid, params[i]); } /* @@ -465,6 +442,8 @@ static int hdmi_setup_channel_allocation static void hdmi_setup_channel_mapping(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { + int i; + if (!ai->CA) return; @@ -473,7 +452,11 @@ static void hdmi_setup_channel_mapping(s * ALSA sequence is front/surr/clfe/side? */ - snd_hda_sequence_write(codec, def_chan_map); + for (i = 0; i < 8; i++) + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + (i << 4) | i); + hdmi_debug_channel_mapping(codec); } @@ -597,7 +580,6 @@ static struct hda_pcm_stream intel_hdmi_ .substreams = 1, .channels_min = 2, .channels_max = 8, - .nid = CVT_NID, /* NID to query formats and rates and setup streams */ .ops = { .open = intel_hdmi_playback_pcm_open, .close = intel_hdmi_playback_pcm_close, @@ -613,6 +595,9 @@ static int intel_hdmi_build_pcms(struct codec->num_pcms = 1; codec->pcm_info = info; + /* NID to query formats and rates and setup streams */ + intel_hdmi_pcm_playback.nid = cvt_nid; + info->name = "INTEL HDMI"; info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; @@ -636,8 +621,9 @@ static int intel_hdmi_init(struct hda_co { hdmi_enable_output(codec); - snd_hda_sequence_write(codec, unsolicited_response_verb); - + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | INTEL_HDMI_EVENT_TAG); return 0; } @@ -657,7 +643,7 @@ static struct hda_codec_ops intel_hdmi_p .unsol_event = intel_hdmi_unsol_event, }; -static int patch_intel_hdmi(struct hda_codec *codec) +static int do_patch_intel_hdmi(struct hda_codec *codec) { struct intel_hdmi_spec *spec; @@ -667,7 +653,7 @@ static int patch_intel_hdmi(struct hda_c spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = CVT_NID; + spec->multiout.dig_out_nid = cvt_nid; codec->spec = spec; codec->patch_ops = intel_hdmi_patch_ops; @@ -679,13 +665,27 @@ static int patch_intel_hdmi(struct hda_c return 0; } +static int patch_intel_hdmi(struct hda_codec *codec) +{ + cvt_nid = 0x02; + pin_nid = 0x03; + return do_patch_intel_hdmi(codec); +} + +static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec) +{ + cvt_nid = 0x02; + pin_nid = 0x04; + return do_patch_intel_hdmi(codec); +} + static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi }, { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, - { .id = 0x80860054, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, + { .id = 0x80860054, .name = "P55 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ };

3 6

[alsa-devel] [PATCH] Tell user that stream to be rewound is suspended
by Lubomir Rintel 03 Aug '09

03 Aug '09

Return STRPIPE instead of EBADF when userspace attempts to rewind of forward a stream that was suspended in meanwhile, so that it can be recovered by snd_pcm_recover(). This was causing Pulseaudio to unload the ALSA sink module under a race condition when it attempted to rewind the stream right after resume from suspend, before writing to the stream which would cause it to revive the stream otherwise. Tested to work with Pulseaudio patched to attempt to snd_pcm_recover() upon receiving an error from snd_pcm_rewind(). Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> --- sound/core/pcm_native.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ac2150e..d89c816 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2208,6 +2208,9 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2253,6 +2256,9 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2299,6 +2305,9 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2345,6 +2354,9 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; -- 1.6.2.5

2 2

[alsa-devel] [PATCH] ALSA: Read buffer overflow
by Roel Kluin 03 Aug '09

03 Aug '09

Check whether index is within bounds before testing the element. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c8b273..45e0ee0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -559,7 +559,7 @@ static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, /* Find enumerated value for current pinctl setting */ i = alc_pin_mode_min(dir); - while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir)) + while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl) i++; *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir); return 0;

2 1

[alsa-devel] [PATCH] [ALSA] hda: Correct EAPD for Dell Inspiron 1525
by Chengu Wang 03 Aug '09

03 Aug '09

2 1