December 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Fix reg_cache_size for stac9766
by Axel Lin 05 Dec '11

05 Dec '11

reg_cache_size is supposed to be the number of elements in the register cache, not the size in bytes. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/stac9766.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 5581953..da71615 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -380,7 +380,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, - .reg_cache_size = sizeof(stac9766_reg), + .reg_cache_size = ARRAY_SIZE(stac9766_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = stac9766_reg, -- 1.7.5.4

2 3

Re: [alsa-devel] [PATCH 4/6] compress: add API header and driver header files
by Nallasellan, Singaravelan 05 Dec '11

05 Dec '11

> > > + * @decoded: Frames decoded by DSP > > > + * @rendered: Frames rendered by DSP into a mixer or an audio > > > + output > > > + * @sampling_rate: sampling rate of audio */ struct > > snd_compr_tstamp { > > > + size_t copied_bytes; > > > + size_t copied_total; > > > + size_t decoded; > > > + size_t rendered; > > > + __u32 sampling_rate; > > > > Is it possible to track the rendering sampling rate in the driver > > itself? > > What do you mean? This is a streaming API, the actual rendering rate will depend on > what endpoint you play on. I guess the driver should know the sampling rate of the endpoint. If only the DSP knows about the endpoint sampling rate, this will make sense. > > > > +struct snd_compr_caps { > > > + __u32 num_codecs; > > > + __u32 min_fragment_size; > > > + __u32 max_fragment_size; > > > + __u32 min_fragments; > > > + __u32 max_fragments; > > > + __u32 codecs[MAX_NUM_CODECS]; > > > + __u32 reserved[11]; > > > > Can we add number of compress streams supported in DSP? > > You are confusing number of formats supported with number of streams supported. > The latter is exposed through the actual devices supported. No, DSP may be able to support only so many simultaneous streams at any point of time. Yet, DSP can support any number of formats. Though the device entry may be created for each compressed stream, how will the user space programs get the number of compressed streams supported?

2 1

[alsa-devel] [PATCH] asoc: sst_platform: fix the dsp driver interface
by Vinod Koul 05 Dec '11

05 Dec '11

lower level drivers typically register with upper layers. So fix by exporting symbols from sst_platform driver for dsp driver to register to sst platform driver Now this driver doesnt depend on sst driver, so remove the dependency and the header files Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/soc/mid-x86/Kconfig | 1 - sound/soc/mid-x86/sst_platform.c | 130 ++++++++++++++++++++++++-------------- sound/soc/mid-x86/sst_platform.h | 82 ++++++++++++++++++++++-- 3 files changed, 158 insertions(+), 55 deletions(-) diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index 2935042..61c10bf 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,7 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC - depends on SND_INTEL_SST select SND_SOC_SN95031 select SND_SST_PLATFORM help diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 3e78260..0012f67 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -31,10 +31,51 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include "../../../drivers/staging/intel_sst/intel_sst_ioctl.h" -#include "../../../drivers/staging/intel_sst/intel_sst.h" #include "sst_platform.h" +static struct sst_device *sst; +static DEFINE_MUTEX(sst_lock); + +int sst_register_dsp(struct sst_device *dev) +{ + BUG_ON(!dev); + if (!try_module_get(dev->dev->driver->owner)) + return -ENODEV; + mutex_lock(&sst_lock); + if (sst) { + pr_err("we already have a device %s\n", sst->name); + module_put(dev->dev->driver->owner); + mutex_unlock(&sst_lock); + return -EEXIST; + } + pr_debug("registering device %s\n", dev->name); + sst = dev; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_register_dsp); + +int sst_unregister_dsp(struct sst_device *dev) +{ + BUG_ON(!dev); + if (dev != sst) + return -EINVAL; + + mutex_lock(&sst_lock); + + if (!sst) { + mutex_unlock(&sst_lock); + return -EIO; + } + + module_put(sst->dev->driver->owner); + pr_debug("unreg %s\n", sst->name); + sst = NULL; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_unregister_dsp); + static struct snd_pcm_hardware sst_platform_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_DOUBLE | @@ -134,37 +175,34 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) } static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct snd_sst_stream_params *param) + struct sst_pcm_params *param) { - param->uc.pcm_params.codec = SST_CODEC_TYPE_PCM; - param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; - param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; - param->uc.pcm_params.reserved = 0; - param->uc.pcm_params.sfreq = substream->runtime->rate; - param->uc.pcm_params.ring_buffer_size = - snd_pcm_lib_buffer_bytes(substream); - param->uc.pcm_params.period_count = substream->runtime->period_size; - param->uc.pcm_params.ring_buffer_addr = - virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->uc.pcm_params.period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", - param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz); + param->codec = SST_CODEC_TYPE_PCM; + param->num_chan = (u8) substream->runtime->channels; + param->pcm_wd_sz = substream->runtime->sample_bits; + param->reserved = 0; + param->sfreq = substream->runtime->rate; + param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); + param->period_count = substream->runtime->period_size; + param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); } static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct snd_sst_stream_params param = {{{0,},},}; - struct snd_sst_params str_params = {0}; + struct sst_pcm_params param = {0}; + struct sst_stream_params str_params = {0}; int ret_val; /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, &param); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.uc.pcm_params.codec; + str_params.codec = param.codec; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { str_params.ops = STREAM_OPS_PLAYBACK; str_params.device_type = substream->pcm->device + 1; @@ -176,7 +214,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) pr_debug("Capture stream,Device %d\n", substream->pcm->device); } - ret_val = stream->sstdrv_ops->pcm_control->open(&str_params); + ret_val = stream->ops->open(&str_params); pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); if (ret_val < 0) return ret_val; @@ -215,7 +253,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.mad_substream = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->sstdrv_ops->pcm_control->device_control( + ret_val = stream->ops->device_control( SST_SND_STREAM_INIT, &stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); @@ -228,7 +266,6 @@ static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; struct sst_runtime_stream *stream; - int ret_val = 0; pr_debug("sst_platform_open called\n"); runtime = substream->runtime; @@ -237,27 +274,27 @@ static int sst_platform_open(struct snd_pcm_substream *substream) if (!stream) return -ENOMEM; spin_lock_init(&stream->status_lock); - stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; - /* allocate memory for SST API set */ - stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), - GFP_KERNEL); - if (!stream->sstdrv_ops) { - pr_err("sst: mem allocation for ops fail\n"); + + /* get the sst ops */ + mutex_lock(&sst_lock); + if (!sst) { + pr_err("no device available to run\n"); + mutex_unlock(&sst_lock); kfree(stream); - return -ENOMEM; + return -ENODEV; } - stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; - stream->sstdrv_ops->module_name = SST_CARD_NAMES; - /* registering with SST driver to get access to SST APIs to use */ - ret_val = register_sst_card(stream->sstdrv_ops); - if (ret_val) { - pr_err("sst: sst card registration failed\n"); - kfree(stream->sstdrv_ops); + if (!try_module_get(sst->dev->driver->owner)) { + mutex_unlock(&sst_lock); kfree(stream); - return ret_val; + return -ENODEV; } + stream->ops = sst->ops; + mutex_unlock(&sst_lock); + + stream->stream_info.str_id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ runtime->private_data = stream; return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -272,9 +309,8 @@ static int sst_platform_close(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->sstdrv_ops->pcm_control->close(str_id); - unregister_sst_card(stream->sstdrv_ops); - kfree(stream->sstdrv_ops); + ret_val = stream->ops->close(str_id); + module_put(sst->dev->driver->owner); kfree(stream); return ret_val; } @@ -288,8 +324,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->sstdrv_ops->pcm_control->device_control( - SST_SND_DROP, &str_id); + ret_val = stream->ops->device_control( + SST_SND_DROP, &str_id); return ret_val; } @@ -341,8 +377,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, default: return -EINVAL; } - ret_val = stream->sstdrv_ops->pcm_control->device_control(str_cmd, - &str_id); + ret_val = stream->ops->device_control(str_cmd, &str_id); if (!ret_val) sst_set_stream_status(stream, status); @@ -362,7 +397,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->sstdrv_ops->pcm_control->device_control( + ret_val = stream->ops->device_control( SST_SND_BUFFER_POINTER, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); @@ -469,6 +504,7 @@ static struct platform_driver sst_platform_driver = { static int __init sst_soc_platform_init(void) { pr_debug("sst_soc_platform_init called\n"); + sst = NULL; return platform_driver_register(&sst_platform_driver); } module_init(sst_soc_platform_init); diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index df37028..f04f4f7 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -42,14 +42,14 @@ #define SST_MIN_PERIODS 2 #define SST_MAX_PERIODS (1024*2) #define SST_FIFO_SIZE 0 -#define SST_CARD_NAMES "intel_mid_card" -#define MSIC_VENDOR_ID 3 +#define SST_CODEC_TYPE_PCM 1 -struct sst_runtime_stream { - int stream_status; - struct pcm_stream_info stream_info; - struct intel_sst_card_ops *sstdrv_ops; - spinlock_t status_lock; +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long long buffer_ptr; + int sfreq; }; enum sst_drv_status { @@ -60,4 +60,72 @@ enum sst_drv_status { SST_PLATFORM_DROPPED, }; +enum sst_controls { + SST_SND_ALLOC = 0x00, + SST_SND_PAUSE = 0x01, + SST_SND_RESUME = 0x02, + SST_SND_DROP = 0x03, + SST_SND_FREE = 0x04, + SST_SND_BUFFER_POINTER = 0x05, + SST_SND_STREAM_INIT = 0x06, + SST_SND_START = 0x07, + SST_MAX_CONTROLS = 0x07, +}; + +enum sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, + STREAM_OPS_CAPTURE, +}; + +enum sst_audio_device_type { + SND_SST_DEVICE_HEADSET = 1, + SND_SST_DEVICE_IHF, + SND_SST_DEVICE_VIBRA, + SND_SST_DEVICE_HAPTIC, + SND_SST_DEVICE_CAPTURE, +}; + +/* PCM Parameters */ +struct sst_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u32 ring_buffer_size; + u32 period_count; /* period elapsed in samples*/ + u32 ring_buffer_addr; +}; + +struct sst_stream_params { + u32 result; + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct sst_pcm_params sparams; +}; + +struct sst_ops { + int (*open) (struct sst_stream_params *str_param); + int (*device_control) (int cmd, void *arg); + int (*close) (unsigned int str_id); +}; + +struct sst_runtime_stream { + int stream_status; + struct pcm_stream_info stream_info; + struct sst_ops *ops; + spinlock_t status_lock; +}; + +struct sst_device { + char *name; + struct device *dev; + struct sst_ops *ops; +}; + +int sst_register_dsp(struct sst_device *sst); +int sst_unregister_dsp(struct sst_device *sst); #endif -- 1.7.0.4

2 4

[alsa-devel] [PATCH] ASoC: Convert e740_wm9705 to use gpio_request_one()
by Axel Lin 05 Dec '11

05 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/e740_wm9705.c | 20 ++++++-------------- 1 files changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 35ed7eb8c..818dc57 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -146,29 +146,21 @@ static int __init e740_init(void) if (!machine_is_e740()) return -ENODEV; - ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + /* Disable audio */ + ret = gpio_request_one(GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp"); if (ret) return ret; - ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + ret = gpio_request_one(GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, + "Output amp"); if (ret) goto free_mic_amp_gpio; - ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + ret = gpio_request_one(GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, + "Audio power"); if (ret) goto free_op_amp_gpio; - /* Disable audio */ - ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); - if (ret) - goto free_apwr_gpio; - ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); - if (ret) - goto free_apwr_gpio; - ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); - if (ret) - goto free_apwr_gpio; - e740_snd_device = platform_device_alloc("soc-audio", -1); if (!e740_snd_device) { ret = -ENOMEM; -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: uda1380: Convert to gpio_request_one()
by Axel Lin 05 Dec '11

05 Dec '11

Using gpio_request_one can make the error handling simpler. Also remove a redundant "Failed to issue reset" error message. We already show the error message in uda1380_reset() error path. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/uda1380.c | 28 ++++++++-------------------- 1 files changed, 8 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 3a238cb..c1b05b7 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -732,27 +732,21 @@ static int uda1380_probe(struct snd_soc_codec *codec) return -EINVAL; if (gpio_is_valid(pdata->gpio_reset)) { - ret = gpio_request(pdata->gpio_reset, "uda1380 reset"); + ret = gpio_request_one(pdata->gpio_reset, GPIOF_OUT_INIT_LOW, + "uda1380 reset"); if (ret) goto err_out; - ret = gpio_direction_output(pdata->gpio_reset, 0); - if (ret) - goto err_gpio_reset_conf; } if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request(pdata->gpio_power, "uda1380 power"); - if (ret) - goto err_gpio; - ret = gpio_direction_output(pdata->gpio_power, 0); + ret = gpio_request_one(pdata->gpio_power, GPIOF_OUT_INIT_LOW, + "uda1380 power"); if (ret) - goto err_gpio_power_conf; + goto err_free_gpio; } else { ret = uda1380_reset(codec); - if (ret) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_reset; - } + if (ret) + goto err_free_gpio; } INIT_WORK(&uda1380->work, uda1380_flush_work); @@ -776,13 +770,7 @@ static int uda1380_probe(struct snd_soc_codec *codec) return 0; -err_reset: -err_gpio_power_conf: - if (gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); - -err_gpio_reset_conf: -err_gpio: +err_free_gpio: if (gpio_is_valid(pdata->gpio_reset)) gpio_free(pdata->gpio_reset); err_out: -- 1.7.5.4

2 1

[alsa-devel] [PATCH 01/21] ASoC: Remove unused -codec from Wolfson device driver names
by Mark Brown 05 Dec '11

05 Dec '11

Devices that aren't MFDs don't need to distinguish this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8727.c | 2 +- sound/soc/codecs/wm8900.c | 4 ++-- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- 14 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 3a65571..9e92bbb 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -666,7 +666,7 @@ MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); static struct i2c_driver wm8510_i2c_driver = { .driver = { - .name = "wm8510-codec", + .name = "wm8510", .owner = THIS_MODULE, .of_match_table = wm8510_of_match, }, diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index fad90a3..e817056 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -59,7 +59,7 @@ static int __devexit wm8727_remove(struct platform_device *pdev) static struct platform_driver wm8727_codec_driver = { .driver = { - .name = "wm8727-codec", + .name = "wm8727", .owner = THIS_MODULE, }, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6ac80cf..aa0302b 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1258,7 +1258,7 @@ static int __devexit wm8900_spi_remove(struct spi_device *spi) static struct spi_driver wm8900_spi_driver = { .driver = { - .name = "wm8900-codec", + .name = "wm8900", .owner = THIS_MODULE, }, .probe = wm8900_spi_probe, @@ -1302,7 +1302,7 @@ MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); static struct i2c_driver wm8900_i2c_driver = { .driver = { - .name = "wm8900-codec", + .name = "wm8900", .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index babca49..1149a93 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2564,7 +2564,7 @@ MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); static struct i2c_driver wm8904_i2c_driver = { .driver = { - .name = "wm8904-codec", + .name = "wm8904", .owner = THIS_MODULE, }, .probe = wm8904_i2c_probe, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 9f1cce8..4d2854b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -779,7 +779,7 @@ MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); static struct i2c_driver wm8940_i2c_driver = { .driver = { - .name = "wm8940-codec", + .name = "wm8940", .owner = THIS_MODULE, }, .probe = wm8940_i2c_probe, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ca38722..f02256e 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1037,7 +1037,7 @@ MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); static struct i2c_driver wm8955_i2c_driver = { .driver = { - .name = "wm8955-codec", + .name = "wm8955", .owner = THIS_MODULE, }, .probe = wm8955_i2c_probe, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ed2773f..b04f808 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1030,7 +1030,7 @@ MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); static struct i2c_driver wm8960_i2c_driver = { .driver = { - .name = "wm8960-codec", + .name = "wm8960", .owner = THIS_MODULE, }, .probe = wm8960_i2c_probe, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index c058701..434413b 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1116,7 +1116,7 @@ MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); static struct i2c_driver wm8961_i2c_driver = { .driver = { - .name = "wm8961-codec", + .name = "wm8961", .owner = THIS_MODULE, }, .probe = wm8961_i2c_probe, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b01df56..0606484 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -724,7 +724,7 @@ MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); static struct i2c_driver wm8971_i2c_driver = { .driver = { - .name = "wm8971-codec", + .name = "wm8971", .owner = THIS_MODULE, }, .probe = wm8971_i2c_probe, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index e41f999..b6136bb 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -671,7 +671,7 @@ MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); static struct i2c_driver wm8974_i2c_driver = { .driver = { - .name = "wm8974-codec", + .name = "wm8974", .owner = THIS_MODULE, }, .probe = wm8974_i2c_probe, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 608c672..fbff293 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -822,7 +822,7 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) static struct spi_driver wm8988_spi_driver = { .driver = { - .name = "wm8988-codec", + .name = "wm8988", .owner = THIS_MODULE, }, .probe = wm8988_spi_probe, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 58d7f0b..9a53dc4 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1417,7 +1417,7 @@ MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); static struct i2c_driver wm8990_i2c_driver = { .driver = { - .name = "wm8990-codec", + .name = "wm8990", .owner = THIS_MODULE, }, .probe = wm8990_i2c_probe, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 780c24c..e386127 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1641,7 +1641,7 @@ MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); static struct i2c_driver wm8993_i2c_driver = { .driver = { - .name = "wm8993-codec", + .name = "wm8993", .owner = THIS_MODULE, }, .probe = wm8993_i2c_probe, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index f94c060..4d90c54 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -685,7 +685,7 @@ MODULE_DEVICE_TABLE(i2c, wm9090_id); static struct i2c_driver wm9090_i2c_driver = { .driver = { - .name = "wm9090-codec", + .name = "wm9090", .owner = THIS_MODULE, }, .probe = wm9090_i2c_probe, -- 1.7.7.3

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[alsa-devel] ALSA Underrun / Silence playback
by Irfan shaikh 05 Dec '11

05 Dec '11

Hello devel, I am trying to implement s/w params for ALSA playback device such that there is SILENCE playback for UNDERRUN (-32). To achieve above i used below s/w params API's. But still i observe UNDERRUN in my AUDIO Playback. Please correct me if i am mising somenting. {When i do get boundry, i get some weird value 1572864000. Thus i tried to use alsa buffer size for threshold} /*************************************************************************************/ if (snd_pcm_sw_params_get_boundary (alsaParams->p_swparams, &boundary) < 0) { printf ("Could not gete snd_pcm_sw_params_get_boundary"); return NULL; } // threshold = boundary; threshold = alsaParams->periodsize * alsaParams->no_of_periods; if (snd_pcm_sw_params_set_stop_threshold (alsaParams->pcm_handle, alsaParams->p_swparams, threshold) < 0) { printf("Could not gete snd_pcm_sw_params_get_boundary"); return NULL; } if (snd_pcm_sw_params_set_silence_threshold (alsaParams->pcm_handle, alsaParams->p_swparams, 0) < 0 || snd_pcm_sw_params_set_silence_size (alsaParams->pcm_handle, alsaParams->p_swparams, boundary) < 0) /* play silence on XRUN */ { printf("Could not set silence params"); return NULL; } /*************************************************************************************/ Thanks in advance !! Regards,

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[alsa-devel] ASoC: Dynamic CODEC DAI
by Patrick Lai 05 Dec '11

05 Dec '11

Problem overview: The CODEC which I am working on transports digital audio through SLIMBUS instead of I2S. The concept of CODEC DAI does not apply too well with SLIMBUS architecture as CODEC DAIs are typically defined base on a set of I2S digital audio interface(bit clock, world select, sd lines). On the CODEC, there are 10 digital capture ports. They can be independently configured as mono channels or can be run-time grouped together to function like multi-channel CODEC DAIs. On top of that, only some ports out of 10 ports can accept all analog mic inputs or digital mic inputs. Hence, even though the CODEC satisfies the concurrent use cases it was designed for, software would have to be articulate on grouping the ports in order to utilize all ports. So, I cannot simply code up CODEC DAI definitions in the CODEC driver for a particular machine. For now, I am looking for compile time grouping of these ports as use cases are known at the time machine is designed. Proposal: I want to minimize amount of change to soc framework at the same time meeting my requirement. Here is my proposed design In CODEC driver, I will declare codec dai for each port. Then, I will enhance ASoC framework to take a list of CODEC DAIs and pass list of CODEC DAIs back to callback function such as startup, hw_param, prepare, trigger. I believe this approach will not require massive amount of change to framework. Your feedback is appreciated. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH 1/3] ASoC: uda1380: Return proper error in uda1380_modinit failure path
by Axel Lin 05 Dec '11

05 Dec '11

Return proper error for uda1380_modinit if i2c_add_driver() fails. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/uda1380.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6b933ef..3a238cb 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = { static int __init uda1380_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); #endif - return 0; + return ret; } module_init(uda1380_modinit); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700
by Axel Lin 05 Dec '11

05 Dec '11

SND_SOC_RAUMFELD selects SND_SOC_CS4270 which needs CONFIG_I2C, and also selects SND_SOC_AK4104 which needs SPI_MASTER. Thus make SND_SOC_RAUMFELD depend on I2C && SPI_MASTER. Add depend on SPI_MASTER to fix below build error if CONFIG_SPI_MASTER is not selected. LD .tmp_vmlinux1 sound/built-in.o: In function `ak4104_spi_write': last.c:(.text+0x290cc): undefined reference to `spi_sync' sound/built-in.o: In function `ak4104_probe': last.c:(.text+0x292a0): undefined reference to `spi_write_then_read' sound/built-in.o: In function `ak4104_spi_probe': last.c:(.text+0x29398): undefined reference to `spi_setup' sound/built-in.o: In function `ak4104_init': last.c:(.init.text+0x4ec): undefined reference to `spi_register_driver' make: *** [.tmp_vmlinux1] Error 1 Add depend on I2C to fix below build error if CONFIG_I2C is not selected: CC sound/soc/codecs/cs4270.o sound/soc/codecs/cs4270.c: In function 'cs4270_i2c_probe': sound/soc/codecs/cs4270.c:657: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/cs4270.c: In function 'cs4270_init': sound/soc/codecs/cs4270.c:730: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/cs4270.c: In function 'cs4270_exit': sound/soc/codecs/cs4270.c:736: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/cs4270.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 SND_PXA2XX_SOC_HX4700 selects SND_SOC_AK4641 which needs CONFIG_I2C. Thus make SND_PXA2XX_SOC_HX4700 depend on I2C. Add depend on I2C to fix below build error if CONFIG_I2C is not selected: CC sound/soc/codecs/ak4641.o sound/soc/codecs/ak4641.c: In function 'ak4641_modinit': sound/soc/codecs/ak4641.c:646: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/ak4641.c: In function 'ak4641_exit': sound/soc/codecs/ak4641.c:656: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/ak4641.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/Kconfig | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ffd2242..a0f7d3c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 @@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 + depends on SND_PXA2XX_SOC && MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help -- 1.7.5.4

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