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

18 Sep '09

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

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

18 Sep '09

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

4 6

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

18 Sep '09

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

2 9

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

17 Sep '09

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

2 1

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

17 Sep '09

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

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

17 Sep '09

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

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

17 Sep '09

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

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

17 Sep '09

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

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

17 Sep '09

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

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

17 Sep '09

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

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