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[alsa-devel] issue with TWL4030
by Grazvydas Ignotas 13 Jul '10

13 Jul '10

Hi, there seems to be a problem with TWL4030 hw_param programming, which is easy to reproduce over OSS emulation with this simple program: int oss_dev = open("/dev/dsp", O_WRONLY); int stereo = 1, bits = 16, rate = 44100; ioctl(oss_dev, SNDCTL_DSP_SETFMT, &bits); ioctl(oss_dev, SNDCTL_DSP_SPEED, &rate); ioctl(oss_dev, SNDCTL_DSP_STEREO, &stereo); write(oss_dev, somebuff, sizeof(somebuff)); The above program ends up playing at 8kHz, instead of expected 44kHz. I did some call tracing and got this: [ 1092.246032] soc_pcm_hw_params, rate 8000 [ 1092.250122] twl4030_hw_params [ 1092.253234] set rate 8000 [ 1092.256103] configured 1 [ 1092.259368] soc_pcm_hw_params, rate 8000 [ 1092.263427] twl4030_hw_params [ 1092.266845] soc_pcm_hw_params, rate 44100 [ 1092.270965] twl4030_hw_params [ 1094.459106] twl4030_shutdown [ 1094.462036] configured 0 So it looks like twl4030->configured is the culprit of this, removing it solves the problem, but I guess that will break something else. Any ideas how to properly fix this? Gražvydas

6 11

[alsa-devel] [PATCH 0/6] ASoC: fsi: modify patch
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 patches for FSI driver almost all are same as v1 Kuninori Morimoto (6): ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode v1 -> v2 o modiry straight start number as 0 o based on bug fix patches Best regards -- Kuninori Morimoto

1 6

[alsa-devel] [PATCH 0/3] further BKL removal
by Arnd Bergmann 12 Jul '10

12 Jul '10

With the ioctl, block, tty, vfs and llseek series on their way into linux-next, these three patches are attacking the hardest remaining issues. If we get these ready for 2.6.36, the kernel should be almost usable with the BKL disabled. In all three cases, I'm cheating a bit, because the BKL still remains lurking in the dark corners of the three subsystems (i810/i830 for drm, OSS for sound and lockd for fs/locks.c). Arnd Cc: Christoph Hellwig <hch(a)lst.de> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Frederic Weisbecker <fweisbec(a)gmail.com> Cc: Ingo Molnar <mingo(a)redhat.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: "J. Bruce Fields" <bfields(a)fieldses.org> Cc: John Kacur <jkacur(a)redhat.com> Cc: Matthew Wilcox <willy(a)linux.intel.com> Cc: Miklos Szeredi <mszeredi(a)suse.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Trond Myklebust <trond.myklebust(a)fys.uio.no> Cc: alsa-devel(a)alsa-project.org Cc: linux-fsdevel(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Arnd Bergmann (2): drm: kill BKL from common code sound: push BKL into open functions Matthew Wilcox (1): Remove BKL from fs/locks.c arch/um/drivers/hostaudio_kern.c | 6 ++ drivers/gpu/drm/drm_drv.c | 4 +- drivers/gpu/drm/drm_fops.c | 23 ++++---- drivers/gpu/drm/i810/i810_dma.c | 44 +++++++++----- drivers/gpu/drm/i810/i810_drv.c | 2 +- drivers/gpu/drm/i810/i810_drv.h | 1 + drivers/gpu/drm/i830/i830_dma.c | 42 +++++++++---- drivers/gpu/drm/i830/i830_drv.c | 2 +- drivers/gpu/drm/i830/i830_drv.h | 1 + fs/Kconfig | 4 + fs/locks.c | 116 ++++++++++++++++++++++-------------- include/drm/drmP.h | 2 +- sound/core/hwdep.c | 14 +++- sound/core/oss/mixer_oss.c | 19 ++++--- sound/oss/au1550_ac97.c | 26 +++++--- sound/oss/dmasound/dmasound_core.c | 28 +++++++-- sound/oss/msnd_pinnacle.c | 10 ++- sound/oss/sh_dac_audio.c | 9 ++- sound/oss/soundcard.c | 20 ++++--- sound/oss/swarm_cs4297a.c | 17 +++++- sound/oss/vwsnd.c | 8 +++ sound/sound_core.c | 6 +-- 22 files changed, 267 insertions(+), 137 deletions(-)

3 7

[alsa-devel] [PATCH 36/36] sound/core: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/core/oss/mixer_oss.c | 22 +++++++++++----------- sound/core/pcm.c | 3 +-- sound/core/timer.c | 4 ++-- 3 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf2..86afb13 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -691,7 +691,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -740,7 +740,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -753,7 +753,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -766,7 +766,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -776,7 +776,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -813,7 +813,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -861,7 +861,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -925,7 +925,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815d..de5a0eb 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -358,8 +358,7 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..5ab1014 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -962,7 +962,7 @@ static int snd_timer_s_start(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long njiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; njiff = (priv->last_jiffies = jiffies); if (priv->correction > timer->sticks - 1) { priv->correction -= timer->sticks - 1; @@ -981,7 +981,7 @@ static int snd_timer_s_stop(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long jiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; del_timer(&priv->tlist); jiff = jiffies; if (time_before(jiff, priv->last_expires)) -- 1.7.1.337.g6068.dirty

1 0

[alsa-devel] [PATCH 35/36] sound/soc/davinci: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/davinci/davinci-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..def454e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -800,7 +800,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; - iram_dma = (struct snd_dma_buffer *)buf->private_data; + iram_dma = buf->private_data; if (iram_dma) { sram_free(iram_dma->area, iram_dma->bytes); kfree(iram_dma); -- 1.7.1.337.g6068.dirty

1 0

[alsa-devel] How does struct snd_pcm_hardware relate to the actual hardware
by Adam Rosenberg 12 Jul '10

12 Jul '10

I am using DMA to transfer data to SPORT1 on my Blackfin BF537. I am using a CS42448 CODEC that has 8 DACs. I am using TDM mode and have set the following: DMA Buffer Size For Each SPORT Data Line (Rx or Tx) = 131072 bytes DMA Periods = 8 static struct snd_pcm_hardware snd_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME), .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, .buffer_bytes_max = 32768, .channels_min = 2, .channels_max = 2, .period_bytes_min = 8, .period_bytes_max = 4096, .periods_min = 8, .periods_max = 8, }; Everytime I receive a DMA interrupt from the SPORT driver I call snd_pcm_period_elapsed for all of the substreams that are currently active. In the PCM Copy callback I calculate the position within the DMA buffer to write to based on the TDM slot for the channel and the position passed into the function. The audio is not being placed into the DMA buffer properly. The audio plays faster than it should and has a constant stutter. I have read the "writing an ALSA driver" guide multiple times but it does not explain the relationship between struct snd_pcm_hardware, snd_pcm_period_elapsed, and the copy callback clearly enough for me to understand what I am doing wrong. Please explain how these interact in more detail to help me fix this problem. I can also send the full source code to the CS42448 driver I am working on if more details are needed. Thank you, Adam

2 2

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Aquila sound driver
by Chanwoo Choi 12 Jul '10

12 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2Sv5 to support sound. As there is currently no driver for the I2Sv5, the code to use I2Sv5 is deleted on this patch. After I2Sv5 driver is posted in a few weeks, I will post incremental patch which some codes for using I2Sv5. The owner of I2Sv5 said that he will post I2Sv5 driver after a few weeks. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++ 3 files changed, 308 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..0c5d8a5 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,11 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.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 @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..706087f --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,298 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(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/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_codec *codec) +{ + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_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 = 24000000; + int ret = 0; + + /* set the cpu 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; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set 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 codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai voice_dai = { + .name = "Voice", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + /* After I2Sv5 driver is posted, cpu_dai will be enabled. */ + /* .cpu_dai = &s5p_i2s_dai[0], */ + .codec_dai = &wm8994_dai[0], + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai = &voice_dai, + .codec_dai = &wm8994_dai[1], + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .platform = &s3c24xx_soc_platform, + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static struct snd_soc_device aquila_snd_devdata = { + .card = &aquila, + .codec_dev = &soc_codec_dev_wm8994, +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&voice_dai); + if (ret) + return ret; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + platform_set_drvdata(aquila_snd_device, &aquila_snd_devdata); + aquila_snd_devdata.dev = &aquila_snd_device->dev; + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] Help required regarding Capture & palyback code for ALSA
by Reena Chauhan 12 Jul '10

12 Jul '10

Hii I m new to ALSA programming .. i have follow the Guide "Introduction to Sound Programming with ALSA" to make code for capture & playback.when i have compiled the code for capture i dont know from where to give the input. in the Guide it is written that "If you have a microphone connected to your sound card, use a mixer program to set the recording source and level. Alternatively, you can run a CD player program and set the recording source to CD. Try running Listing 4 and redirecting the output to a file" i dont know how to use the mixer program .. but I dont know how to do it to capture some input given by me & playback it .. thanx & Regrads -- Reena Chauhan Scientist-C Application & Baseband Platform, SBSG ,DEAL,Raipur Road Dehradun-248001

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Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Raymond Yau 10 Jul '10

10 Jul '10

>> It might be. It is important not to mix VT1708 and VT1708S. For VT1708S, the following DACs are used in code: /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } >> So 0x12 is not used for any of the surround outputs. I think your testing config is not compiled with smart 5.1 requirement Refer to smart51_get() , this function exepct red jack and blue jack are configued to Output But you still plugged in Mic into the pink jack according to your photo. Your speakers have to plugged in both pink and blue jacks

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[alsa-devel] Kirkwood SoC - AVB support questions ..?
by errordeveloper＠gmail.com 10 Jul '10

10 Jul '10

Hello dear ALSA developer list members, I have a question regarding hardware features of Marvell Kirkwood chip. I is known that the NIC core on the chip includes support for Ethernet AV Bridging, which is still a draft standard ..but i suppose that Marvell released this chip with quite functional AVB implementation. grepping the sk98 driver code actually shows numerous lines with AVB as well as PTP, so i suppose they have implemented IOCTL in the sk98 driver ..i'm still doubtfull about Linux kernel PTP implementation ..is there now in some form or not? Did any of you have a look at AVB ? how do you think the driver for it could fit into Linux Netdev and ALSA? How are the CobraNet and EtherSound drivers designed, has anyone used those cards ?? (one was from DigiGram and another from Audio Science IIRC) Regards, -- Ilya Dmitrichenko

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