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[alsa-devel] [PATCH 1/7] S3C AUDIO: Rename s3c24xx_pcm prefix to s3c_dma
by Jassi Brar 09 Nov '09

09 Nov '09

The s3c24xx_pcm prefix for the soc_platform is inappropriate when some Samsung SoCs have PCM controllers which will eventually have drivers and hence namespace ambiguities. To resolve naming ambiguities in future the following have been renamed in order 1) s3c24xx_pcm_dma_params -> s3c_audio_dma_params 2) s3c24xx_pcm_preallocate_dma_buffer -> s3c_preallocate_dma_buffer 3) s3c24xx_pcm_dmamask -> s3c_dmamask 4) s3c24xx_pcm_XXX -> s3c_dma_XXX Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +- sound/soc/s3c24xx/s3c2412-i2s.c | 4 +- sound/soc/s3c24xx/s3c2443-ac97.c | 10 ++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 10 ++-- sound/soc/s3c24xx/s3c24xx-pcm.c | 76 +++++++++++++++++++------------------- sound/soc/s3c24xx/s3c24xx-pcm.h | 6 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 4 +- 8 files changed, 58 insertions(+), 58 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 28b0ab2..9788e65 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -394,7 +394,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_audio_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index f66854a..02e16db 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -49,8 +49,8 @@ struct s3c_i2sv2_info { unsigned char master; - struct s3c24xx_pcm_dma_params *dma_playback; - struct s3c24xx_pcm_dma_params *dma_capture; + struct s3c_audio_dma_params *dma_playback; + struct s3c_audio_dma_params *dma_capture; u32 suspend_iismod; u32 suspend_iiscon; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ac5e47b..6bc13d5 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -50,14 +50,14 @@ static struct s3c2410_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { +static struct s3c_audio_dma_params s3c2412_i2s_pcm_stereo_out = { .client = &s3c2412_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { +static struct s3c_audio_dma_params s3c2412_i2s_pcm_stereo_in = { .client = &s3c2412_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index b25e9f9..d3bd29a 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -188,21 +188,21 @@ static struct s3c2410_dma_client s3c2443_dma_client_micin = { .name = "AC97 Mic Mono in" }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = { +static struct s3c_audio_dma_params s3c2443_ac97_pcm_stereo_out = { .client = &s3c2443_dma_client_out, .channel = DMACH_PCM_OUT, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = { +static struct s3c_audio_dma_params s3c2443_ac97_pcm_stereo_in = { .client = &s3c2443_dma_client_in, .channel = DMACH_PCM_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { +static struct s3c_audio_dma_params s3c2443_ac97_mic_mono_in = { .client = &s3c2443_dma_client_micin, .channel = DMACH_MIC_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, @@ -290,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_audio_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); @@ -339,7 +339,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_audio_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index c76b8bb..998e752 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -49,14 +49,14 @@ static struct s3c2410_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = { +static struct s3c_audio_dma_params s3c24xx_i2s_pcm_stereo_out = { .client = &s3c24xx_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, .dma_size = 2, }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { +static struct s3c_audio_dma_params s3c24xx_i2s_pcm_stereo_in = { .client = &s3c24xx_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, @@ -258,12 +258,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_audio_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_audio_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; default: @@ -280,7 +280,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_audio_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 27cf097..a0be71f 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -32,7 +32,7 @@ #include "s3c24xx-pcm.h" -static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { +static const struct snd_pcm_hardware s3c_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -62,15 +62,15 @@ struct s3c24xx_runtime_data { dma_addr_t dma_start; dma_addr_t dma_pos; dma_addr_t dma_end; - struct s3c24xx_pcm_dma_params *params; + struct s3c_audio_dma_params *params; }; -/* s3c24xx_pcm_enqueue +/* s3c_dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; @@ -124,19 +124,19 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, spin_lock(&prtd->lock); if (prtd->state & ST_RUNNING) { prtd->dma_loaded--; - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); } spin_unlock(&prtd->lock); } -static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct s3c_audio_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); int ret = 0; @@ -184,7 +184,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; @@ -201,7 +201,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) +static int s3c_dma_prepare(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -234,12 +234,12 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); return ret; } -static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -274,7 +274,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } static snd_pcm_uframes_t -s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) +s3c_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -309,7 +309,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(substream->runtime, res); } -static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) +static int s3c_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; @@ -317,7 +317,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); if (prtd == NULL) @@ -329,7 +329,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) +static int s3c_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -337,14 +337,14 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); if (!prtd) - pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c_dma_close called with prtd == NULL\n"); kfree(prtd); return 0; } -static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, +static int s3c_dma_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -357,23 +357,23 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -static struct snd_pcm_ops s3c24xx_pcm_ops = { - .open = s3c24xx_pcm_open, - .close = s3c24xx_pcm_close, +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c24xx_pcm_hw_params, - .hw_free = s3c24xx_pcm_hw_free, - .prepare = s3c24xx_pcm_prepare, - .trigger = s3c24xx_pcm_trigger, - .pointer = s3c24xx_pcm_pointer, - .mmap = s3c24xx_pcm_mmap, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, }; -static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int s3c_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 = s3c24xx_pcm_hardware.buffer_bytes_max; + size_t size = s3c_dma_hardware.buffer_bytes_max; pr_debug("Entered %s\n", __func__); @@ -388,7 +388,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) return 0; } -static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -411,9 +411,9 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); +static u64 s3c_dmamask = DMA_BIT_MASK(32); -static int s3c24xx_pcm_new(struct snd_card *card, +static int s3c_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -421,19 +421,19 @@ static int s3c24xx_pcm_new(struct snd_card *card, pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c24xx_pcm_dmamask; + card->dev->dma_mask = &s3c_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } if (dai->capture.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) goto out; @@ -444,9 +444,9 @@ static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_platform s3c24xx_soc_platform = { .name = "s3c24xx-audio", - .pcm_ops = &s3c24xx_pcm_ops, - .pcm_new = s3c24xx_pcm_new, - .pcm_free = s3c24xx_pcm_free_dma_buffers, + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); @@ -463,5 +463,5 @@ static void __exit s3c24xx_soc_platform_exit(void) module_exit(s3c24xx_soc_platform_exit); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h index 0088c79..d04ddf8 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h @@ -9,13 +9,13 @@ * ALSA PCM interface for the Samsung S3C24xx CPU */ -#ifndef _S3C24XX_PCM_H -#define _S3C24XX_PCM_H +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H #define ST_RUNNING (1<<0) #define ST_OPENED (1<<1) -struct s3c24xx_pcm_dma_params { +struct s3c_audio_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index b67eed5..03f5a6b 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -46,7 +46,7 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { +static struct s3c_audio_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { [0] = { .channel = DMACH_I2S0_OUT, .client = &s3c64xx_dma_client_out, @@ -61,7 +61,7 @@ static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { }, }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { +static struct s3c_audio_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { [0] = { .channel = DMACH_I2S0_IN, .client = &s3c64xx_dma_client_in, -- 1.6.2.5

4 9

Re: [alsa-devel] Capture not working in MMAP mode with '-v'
by Aggarwal, Anuj 09 Nov '09

09 Nov '09

> -----Original Message----- > From: Aggarwal, Anuj > Sent: Thursday, September 24, 2009 2:28 PM > To: alsa-devel(a)alsa-project.org > Subject: Capture not working in MMAP mode with '-v' > > Hi, > > When I try to capture audio in MMAP mode using the following > command, it works properly: > > arecord -f cd -d 10 -M rec01.wav > > However, if I specify '-v' along with the above command, it fails to > capture anything and prints log and returns immediately: > > Recording WAVE 'rec03.wav' : Signed 16 bit Little Endian, > Rate 44100 Hz, Stereo > Plug PCM: Hardware PCM card 0 'omap3evm' device 0 subdevice 0 > Its setup is: > stream : CAPTURE > access : MMAP_INTERLEAVED > format : S16_LE > subformat : STD > channels : 2 > rate : 44100 > exact rate : 44100 (44100/1) > msbits : 16 > buffer_size : 22052 > period_size : 5513 > period_time : 125011 > tstamp_mode : NONE > period_step : 1 > avail_min : 5513 > period_event : 0 > start_threshold : 1 > stop_threshold : 22052 > silence_threshold: 0 > silence_size : 0 > boundary : 1445199872 > appl_ptr : 0 > hw_ptr : 0 > mmap_area[0] = 0x4030e000,0,32 (16) > mmap_area[1] = 0x4030e000,16,32 (16) > > I have not observed this problem while playing audio, only > while capture this issue crops up. > > Any pointers? Any updates on this? We have observed similar problem on other platforms as well. > > Thanks & Regards, > Anuj Aggarwal >

4 6

[alsa-devel] [PATCH V2] ASoC - Add support for upto 16 channels on OMAP MCBSP
by Liam Girdwood 09 Nov '09

09 Nov '09

From: Graeme Gregory <gg(a)slimlogic.co.uk> This patch increases the number of supported audio channels from 4 to 16 and has been sponsored by Shotspotter Inc. It also fixes a FSYNC rate calculation bug when McBSP is FSYNC master. Signed-off-by: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/omap/omap-mcbsp.c | 59 ++++++++++++++++++++++++------------------- 1 files changed, 33 insertions(+), 26 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3341f49..3ea4a00 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -49,6 +49,8 @@ struct omap_mcbsp_data { */ int active; int configured; + unsigned int in_freq; + int clk_div; }; #define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) @@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; unsigned long port; - unsigned int format; + unsigned int format, div, framesize; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - switch (channels) { - case 2: - if (format == SND_SOC_DAIFMT_I2S) { - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; - /* Set 1 word per (McBSP) frame for phase1 and phase2 */ - wpf--; - regs->rcr2 |= RFRLEN2(wpf - 1); - regs->xcr2 |= XFRLEN2(wpf - 1); - } - case 1: - case 4: - /* Set word per (McBSP) frame for phase1 */ - regs->rcr1 |= RFRLEN1(wpf - 1); - regs->xcr1 |= XFRLEN1(wpf - 1); - break; - default: - /* Unsupported number of channels */ - return -EINVAL; + if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); } + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ @@ -330,15 +323,26 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* In McBSP master modes, FRAME (i.e. sample rate) is generated + * by _counting_ BCLKs. Calculate frame size in BCLKs */ + div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; + framesize = (mcbsp_data->in_freq / div) / params_rate(params); + + if (framesize < wlen * channels) { + printk(KERN_ERR "%s: not enough bandwidth for desired rate and channels\n", + __func__); + return -EINVAL; + } + /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen - 1); + regs->srgr2 |= FPER(framesize - 1); + regs->srgr1 |= FWID((framesize >> 1) - 1); break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID(0); break; } @@ -454,6 +458,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; + mcbsp_data->clk_div = div; regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -554,6 +559,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + mcbsp_data->in_freq = freq; + switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: regs->srgr2 |= CLKSM; @@ -598,13 +605,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .id = (link_id), \ .playback = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ -- 1.6.3.3

4 9

[alsa-devel] [PATCH 1/2] ASoC: sh: fsi: Add runtime PM support
by Kuninori Morimoto 09 Nov '09

09 Nov '09

This patch add support runtime PM. Driver callbacks for Runtime PM are empty because the device registers are always re-initialized after pm_runtime_get_sync(). The Runtime PM functions replaces the clock framework module stop bit handling in this driver. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> Signed-off-by: Magnus Damm <damm(a)opensource.se> --- sound/soc/sh/fsi.c | 39 +++++++++++++++++++++++++-------------- 1 files changed, 25 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e1a3d1a..aaec738 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/list.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +105,6 @@ struct fsi_priv { 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; @@ -559,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -674,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -872,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -903,14 +901,9 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - /* 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_iounmap; - } + pm_suspend_ignore_children(&pdev->dev, true); + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -947,7 +940,7 @@ 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); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -957,9 +950,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, -- 1.6.3.3

1 1

[alsa-devel] [PATCH v4 00/12] introduce skip_spaces(), reducing code size plus some clean-ups
by André Goddard Rosa 08 Nov '09

08 Nov '09

This patch reduces lib.a code size by 173 bytes on my Core 2 with gcc 4.4.1 even considering that it exports a newly defined function skip_spaces() to drivers: text data bss dec hex filename 64867 840 592 66299 102fb (TOTALS-lib.a-before) 64954 584 588 66126 1024e (TOTALS-lib.a-after) and implements some code tidy up. Besides reducing lib.a size, it converts many in-tree drivers to use the newly defined function, which makes another small reduction on kernel size overall when those drivers are used. p.s.: Ingo, I hope to have finally fixed my environment by using send-email now. Please let me know if you have any problem with these and thank you for the good advice. get_maintainer.pl gave lots of people as output, I hope that's ok. Changelog: v4: convert drivers over newly defined skip_spaces() function v3: improved comments on patch 5/7 and factorize a switch statement on 6/7 as per suggestions from Ingo and Frederic (thanks!) v2: addressed feedback from Frederic Weisbecker review (thanks!!) and split into separate patches v1: original submission André Goddard Rosa (12): vsprintf: factorize "(null)" string vsprintf: pre-calculate final string length for later use vsprintf: give it some care to please checkpatch.pl vsprintf: use TOLOWER whenever possible vsprintf: reduce code size by avoiding extra check vsprintf: move local vars to block local vars and remove unneeded ones vsprintf: factor out skip_space code in a separate function vsprintf: reuse almost identical simple_strtoulX() functions ctype: constify read-only _ctype string string: factorize skip_spaces and export it to be generally available string: on strstrip(), first remove leading spaces before running over str tree-wide: convert open calls to remove spaces to skip_spaces() lib function arch/s390/kernel/debug.c | 3 +- arch/um/drivers/mconsole_kern.c | 16 +- arch/x86/kernel/cpu/mtrr/if.c | 11 +- drivers/md/dm-table.c | 6 +- drivers/md/md.c | 4 +- drivers/parisc/pdc_stable.c | 9 +- drivers/platform/x86/thinkpad_acpi.c | 7 +- drivers/pnp/interface.c | 36 +--- drivers/s390/block/dasd_proc.c | 5 +- drivers/video/backlight/lcd.c | 4 +- drivers/video/display/display-sysfs.c | 2 +- fs/cachefiles/daemon.c | 8 +- fs/ext4/super.c | 7 +- include/linux/ctype.h | 2 +- include/linux/string.h | 1 + kernel/params.c | 8 +- lib/argv_split.c | 13 +- lib/ctype.c | 50 +++--- lib/dynamic_debug.c | 4 +- lib/string.c | 19 ++- lib/vsprintf.c | 342 ++++++++++++++++----------------- net/irda/irnet/irnet.h | 1 + net/irda/irnet/irnet_ppp.c | 8 +- net/netfilter/xt_recent.c | 3 +- sound/pci/hda/hda_hwdep.c | 7 +- 25 files changed, 264 insertions(+), 312 deletions(-)

5 17

[alsa-devel] RME HDSPe AIO
by Jonathan Leonard 08 Nov '09

08 Nov '09

Hello, I am writing in regards to the RME HDSPe AIO card and its support in ALSA. The card itself in 64studio is recognized incorrectly as HDSP MADI rev d4. Running lspci -nn shows this line for the device: 02:00.0 Multimedia audio controller [0401]: Xilinx Corporation RME Hammerfall DSP MADI [10ee:3fc6] (rev d4) ALSA in 64studio 3.0 beta 3 loads snd-hdspm incorrectly when it should be loading snd-hdsp. I have reviewed pcitables and pci.ids on this distro, and also the public pci reference and the above device id is currently assigned to: Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d4) The AIO is purported to be very similar to the 9632 and should be using the snd-hdsp module. I would like to use this card in ALSA, and have begun reviewing the guide for making ALSA drivers, but am not sure how exactly to fix the matter since the manufacturer has not incremented the id, or they have not assigned the same id as a 9632. This device is recognized and functioning perfectly on the XP partition. Besides my individual circumstance I wonder if ALSA should not be accomodating this great card somehow. Let me know if I can help by contacting RME developers for info or otherwise. Thank you, jonathan adams leonard

2 2

[alsa-devel] [PATCH]Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK
by Marin Mitov 08 Nov '09

08 Nov '09

Hi all, Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK Signed-off-by: Marin Mitov <mitov(a)issp.bas.bg> ========================================================= --- sound/soc/s6000/s6000-pcm.c.orig 2009-11-08 17:49:09.000000000 +0200 +++ sound/soc/s6000/s6000-pcm.c 2009-11-08 17:50:56.000000000 +0200 @@ -423,7 +423,7 @@ snd_pcm_lib_preallocate_free_for_all(pcm); } -static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; +static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -435,7 +435,7 @@ if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (params->dma_in) { s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in),

1 0

[alsa-devel] [PATCHv2 alsa-lib 0/3] Close-on-exec flag for device nodes
by Rémi Denis-Courmont 08 Nov '09

08 Nov '09

Hello, This is basically the same patch series as yesterday, but addressing Lennart's issue. Comments welcome. configure.in | 2 ++ include/local.h | 17 +++++++++++------ src/control/control_hw.c | 11 ----------- src/hwdep/hwdep_hw.c | 11 ----------- src/pcm/pcm_hw.c | 12 ------------ src/rawmidi/rawmidi_hw.c | 11 ----------- src/seq/seq_hw.c | 11 ----------- src/timer/timer_hw.c | 11 ----------- 8 files changed, 13 insertions(+), 73 deletions(-) -- Rémi Denis-Courmont http://www.remlab.net/

3 10

Re: [alsa-devel] [Intel-gfx] Problems with HDMI audio on Intel DG45FC motherboard
by Wu Fengguang 08 Nov '09

08 Nov '09

On Wed, Oct 21, 2009 at 06:37:58AM +0800, David Härdeman wrote: > On Tue, Oct 20, 2009 at 09:26:00AM +0800, Wu Fengguang wrote: > >On Thu, Oct 15, 2009 at 01:40:53AM +0800, David Härdeman wrote: > >> ii) Is there any documentation somewhere on how this mapping is supposed > >> to be performed in user space? > > > >I think Shane has provided a good example for you in another email :) > >Here are more descriptions for the route plugin: > > So perhaps I should still file a bug with the ALSA bugtracker asking for > the default ALSA userspace to include a channel mapping along Shane's > recommendation? That would be good. I'm not sure if PulseAudio need a separate fix. (Lennart CCed). > >> >Complete silence for how much time? > >> > >> For the entire duration of the particular movie/audio track/video > >> clip/whatever. > >> > > > >I actually have a DG45FC box and have not run into this problem for > >all the HDMI monitors I have. What's your monitor model? > > A Samsung LE-55A956 LCD TV. But I think the problem is not with the > monitor (TV) but rather with the receiver that is between the DG45FC and > the TV. The receiver is a Marantz SR8002 with HDMI support. Got it. If you have a HDMI capable TV or monitor, it would be possible to get rid of the Marantz and check if the problem sticks :) > >Did the kernel emit some error messages? > > Nope OK. You can get more debug info if turning on CONFIG_SND_VERBOSE_PROCFS=y CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_PCM_XRUN_DEBUG=y > >What if you change display > >modes with xrandr? > > Haven't tried that yet...currently running on 1920x1080@60 which should > (I guess) be one of the most common modes with modern LCD TV's. The infoframe would be different for 2-channel pure music and 5.1 channel movie. The video timing might also affect audio stream, because the HDMI audio data is transfered in the gaps of video data. > >HDMI codec/sink seem to take some time to response to the output > >enable and new infoframe, so there are some delay. I've moved the HDMI > >output enable command to module load time, this helps. The infoframe > >is in theory created in run time based on the format of _each_ new > >audio stream, so infoframe transmission has to be started/stopped > >for each track.. > > So my current theory is along the line of HDMI audio infoframe or video > infoframe problems. Do you have any pointers where in the driver(s) > the infoframes (both audio and video) are generated so that I can add > some debugging statements to dump the frames and compare the working > and non-working scenarios? I guess the infoframe itself should be identical for the same clip. Have you tried upgrading to a recent Xorg/intel gfx driver? Anyway, the kernel driver code is linux/sound/pci/hda/patch_intelhdmi.c The relevant function is hdmi_setup_audio_infoframe(). Inside which the hdmi_setup_channel_mapping() is not functioning for G45, hdmi_setup_channel_allocation() is no-op for 2-channel music. hdmi_debug_dip_size() and hdmi_clear_dip_buffers() can also be commented out. And feel free to ask more questions :) Thanks, Fengguang

2 1

[alsa-devel] [PATCH] es18xx: code improvements
by Krzysztof Helt 08 Nov '09

08 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> 1. Set the third argument of the snd_device_new to not NULL, so there is no warning about bug during chip detection. The third argument is not used in this driver. It was changed in my previous patch. 2. Remove the fm_port and mpu_port fields from the snd_es18xx structure. They can be converted to function arguments. 3. Remove the dmaN_size fields from the snd_es18xx structure. These values are used only in pointer functions and can be easily calculated. 4. Remove the ctrl_lock spinlock which is used only in one read function which is called once during chip initialization. There are many writes to the same register and they are not protected on purpose (see the comment ina the snd_es18xx_config_write()). 5. Use the first part of the text5Sources string table as the text4Soruces table (they are the same). 6. Merge the same cases for the ES1887 and ES1888 when setting chip's caps. 7. Move the snd_es18xx_reset() to __devinit section. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/es18xx.c | 91 +++++++++++++++++++++++---------------------------- 1 files changed, 41 insertions(+), 50 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5cf42b4..06e871e 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -102,8 +102,6 @@ struct snd_es18xx { unsigned long port; /* port of ESS chip */ - unsigned long mpu_port; /* MPU-401 port of ESS chip */ - unsigned long fm_port; /* FM port */ unsigned long ctrl_port; /* Control port of ESS chip */ struct resource *res_port; struct resource *res_mpu_port; @@ -116,8 +114,6 @@ struct snd_es18xx { unsigned short audio2_vol; /* volume level of audio2 */ unsigned short active; /* active channel mask */ - unsigned int dma1_size; - unsigned int dma2_size; unsigned int dma1_shift; unsigned int dma2_shift; @@ -135,7 +131,6 @@ struct snd_es18xx { spinlock_t reg_lock; spinlock_t mixer_lock; - spinlock_t ctrl_lock; #ifdef CONFIG_PM unsigned char pm_reg; #endif @@ -354,7 +349,7 @@ static inline int snd_es18xx_mixer_writable(struct snd_es18xx *chip, unsigned ch } -static int snd_es18xx_reset(struct snd_es18xx *chip) +static int __devinit snd_es18xx_reset(struct snd_es18xx *chip) { int i; outb(0x03, chip->port + 0x06); @@ -490,8 +485,6 @@ static int snd_es18xx_playback1_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma2_size = size; - snd_es18xx_rate_set(chip, substream, DAC2); /* Transfer Count Reload */ @@ -591,8 +584,6 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -659,8 +650,6 @@ static int snd_es18xx_playback2_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -821,17 +810,18 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) { if (!(chip->active & DAC2)) return 0; - pos = snd_dma_pointer(chip->dma2, chip->dma2_size); + pos = snd_dma_pointer(chip->dma2, size); return pos >> chip->dma2_shift; } else { if (!(chip->active & DAC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } } @@ -839,11 +829,12 @@ static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *s static snd_pcm_uframes_t snd_es18xx_capture_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (!(chip->active & ADC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } @@ -974,9 +965,6 @@ static int snd_es18xx_capture_close(struct snd_pcm_substream *substream) static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts4Source[4] = { - "Mic", "CD", "Line", "Master" - }; static char *texts5Source[5] = { "Mic", "CD", "Line", "Master", "Mix" }; @@ -994,7 +982,8 @@ static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ele uinfo->value.enumerated.items = 4; if (uinfo->value.enumerated.item > 3) uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts4Source[uinfo->value.enumerated.item]); + strcpy(uinfo->value.enumerated.name, + texts5Source[uinfo->value.enumerated.item]); break; case 0x1887: case 0x1888: @@ -1378,11 +1367,9 @@ ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; - unsigned long flags; - spin_lock_irqsave(&chip->ctrl_lock, flags); + outb(reg, chip->ctrl_port); data = inb(chip->ctrl_port + 1); - spin_unlock_irqrestore(&chip->ctrl_lock, flags); return data; } @@ -1398,7 +1385,9 @@ static void __devinit snd_es18xx_config_write(struct snd_es18xx *chip, #endif } -static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) +static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { int mask = 0; @@ -1412,15 +1401,15 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) if (chip->caps & ES18XX_CONTROL) { /* Hardware volume IRQ */ snd_es18xx_config_write(chip, 0x27, chip->irq); - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { /* FM I/O */ - snd_es18xx_config_write(chip, 0x62, chip->fm_port >> 8); - snd_es18xx_config_write(chip, 0x63, chip->fm_port & 0xff); + snd_es18xx_config_write(chip, 0x62, fm_port >> 8); + snd_es18xx_config_write(chip, 0x63, fm_port & 0xff); } - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU-401 I/O */ - snd_es18xx_config_write(chip, 0x64, chip->mpu_port >> 8); - snd_es18xx_config_write(chip, 0x65, chip->mpu_port & 0xff); + snd_es18xx_config_write(chip, 0x64, mpu_port >> 8); + snd_es18xx_config_write(chip, 0x65, mpu_port & 0xff); /* MPU-401 IRQ */ snd_es18xx_config_write(chip, 0x28, chip->irq); } @@ -1507,11 +1496,12 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) snd_es18xx_mixer_write(chip, 0x7A, 0x68); /* Enable and set hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x06); - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU401 share irq with audio Joystick enabled FM enabled */ - snd_es18xx_mixer_write(chip, 0x40, 0x43 | (chip->mpu_port & 0xf0) >> 1); + snd_es18xx_mixer_write(chip, 0x40, + 0x43 | (mpu_port & 0xf0) >> 1); } snd_es18xx_mixer_write(chip, 0x7f, ((irqmask + 1) << 1) | 0x01); } @@ -1629,7 +1619,9 @@ static int __devinit snd_es18xx_identify(struct snd_es18xx *chip) return 0; } -static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) +static int __devinit snd_es18xx_probe(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { if (snd_es18xx_identify(chip) < 0) { snd_printk(KERN_ERR PFX "[0x%lx] ESS chip not found\n", chip->port); @@ -1650,8 +1642,6 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV; break; case 0x1887: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; - break; case 0x1888: chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; break; @@ -1666,7 +1656,7 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) if (chip->dma1 == chip->dma2) chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME); - return snd_es18xx_initialize(chip); + return snd_es18xx_initialize(chip, mpu_port, fm_port); } static struct snd_pcm_ops snd_es18xx_playback_ops = { @@ -1802,10 +1792,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - spin_lock_init(&chip->ctrl_lock); chip->port = port; - chip->mpu_port = mpu_port; - chip->fm_port = fm_port; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; @@ -1841,11 +1828,11 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, } chip->dma2 = dma2; - if (snd_es18xx_probe(chip) < 0) { + if (snd_es18xx_probe(chip, mpu_port, fm_port) < 0) { snd_es18xx_free(card); - return -ENODEV; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, NULL, &ops); + return -ENODEV; + } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_es18xx_free(card); return err; @@ -1980,7 +1967,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ #ifdef CONFIG_PNP -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; #endif static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */ #ifndef CONFIG_PNP @@ -2160,19 +2147,23 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, chip->fm_port, chip->fm_port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) { - snd_printk(KERN_WARNING PFX "opl3 not detected at 0x%lx\n", chip->fm_port); + if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) { + snd_printk(KERN_WARNING PFX + "opl3 not detected at 0x%lx\n", + fm_port[dev]); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) return err; } } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, - chip->mpu_port, 0, - irq[dev], 0, - &chip->rmidi)) < 0) + err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, + mpu_port[dev], 0, + irq[dev], 0, &chip->rmidi); + if (err < 0) return err; } -- 1.6.4 Tanie rozmowy telefoniczne! 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