October 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 05/25] ASoC: Samsung: Rename ASoC DMA driver
by Jassi Brar 19 Oct '10

19 Oct '10

From: Jassi Brar <jassi.brar(a)samsung.com> Rename Samsung ASoC DMA driver s3c-dma.[c/h] -> dma.[c/h] Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Makefile | 2 +- sound/soc/s3c24xx/dma.c | 503 ++++++++++++++++++++++++ sound/soc/s3c24xx/dma.h | 30 ++ sound/soc/s3c24xx/goni_wm8994.c | 2 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/rx1950_uda1380.c | 2 +- sound/soc/s3c24xx/s3c-ac97.c | 2 +- sound/soc/s3c24xx/s3c-dma.c | 503 ------------------------ sound/soc/s3c24xx/s3c-dma.h | 30 -- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c-pcm.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- sound/soc/s3c24xx/smartq_wm8987.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 2 +- sound/soc/s3c24xx/smdk_spdif.c | 2 +- sound/soc/s3c24xx/smdk_wm9713.c | 2 +- sound/soc/s3c24xx/spdif.c | 2 +- 28 files changed, 557 insertions(+), 557 deletions(-) create mode 100644 sound/soc/s3c24xx/dma.c create mode 100644 sound/soc/s3c24xx/dma.h delete mode 100644 sound/soc/s3c24xx/s3c-dma.c delete mode 100644 sound/soc/s3c24xx/s3c-dma.h diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 4e232f1..20aac45 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,5 +1,5 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c-dma.o +snd-soc-s3c24xx-objs := dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o diff --git a/sound/soc/s3c24xx/dma.c b/sound/soc/s3c24xx/dma.c new file mode 100644 index 0000000..b4c7c85 --- /dev/null +++ b/sound/soc/s3c24xx/dma.c @@ -0,0 +1,503 @@ +/* + * dma.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License 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/init.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma.h> + +#include "dma.h" + +static const struct snd_pcm_hardware s3c_dma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, + .fifo_size = 32, +}; + +struct s3c24xx_runtime_data { + spinlock_t lock; + int state; + unsigned int dma_loaded; + unsigned int dma_limit; + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + struct s3c_dma_params *params; +}; + +/* s3c_dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. +*/ +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; + unsigned int limit; + int ret; + + pr_debug("Entered %s\n", __func__); + + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { + unsigned long len = prtd->dma_period; + + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + len) > prtd->dma_end) { + len = prtd->dma_end - pos; + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", + __func__, len); + } + + ret = s3c2410_dma_enqueue(prtd->params->channel, + substream, pos, len); + + if (ret == 0) { + prtd->dma_loaded++; + pos += prtd->dma_period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } else + break; + } + + prtd->dma_pos = pos; +} + +static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, + void *dev_id, int size, + enum s3c2410_dma_buffresult result) +{ + struct snd_pcm_substream *substream = dev_id; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) + return; + + prtd = substream->runtime->private_data; + + if (substream) + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { + prtd->dma_loaded--; + s3c_dma_enqueue(substream); + } + + spin_unlock(&prtd->lock); +} + +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; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + int ret = 0; + + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, client %p, channel %d\n", prtd->params, + prtd->params->client, prtd->params->channel); + + ret = s3c2410_dma_request(prtd->params->channel, + prtd->params->client, NULL); + + if (ret < 0) { + printk(KERN_ERR "failed to get dma channel\n"); + return ret; + } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); + } + + s3c2410_dma_set_buffdone_fn(prtd->params->channel, + s3c24xx_audio_buffdone); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_limit = runtime->hw.periods_min; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + /* TODO - do we need to ensure DMA flushed */ + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + s3c2410_dma_free(prtd->params->channel, prtd->params->client); + prtd->params = NULL; + } + + return 0; +} + +static int s3c_dma_prepare(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, + prtd->params->dma_addr); + } else { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, + prtd->params->dma_addr); + } + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size); + + /* flush the DMA channel */ + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + s3c_dma_enqueue(substream); + + return ret; +} + +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t +s3c_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + unsigned long res; + dma_addr_t src, dst; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + s3c2410_dma_getposition(prtd->params->channel, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - prtd->dma_start; + else + res = src - prtd->dma_start; + + spin_unlock(&prtd->lock); + + pr_debug("Pointer %x %x\n", src, dst); + + /* we seem to be getting the odd error from the pcm library due + * to out-of-bounds pointers. this is maybe due to the dma engine + * not having loaded the new values for the channel before being + * callled... (todo - fix ) + */ + + if (res >= snd_pcm_lib_buffer_bytes(substream)) { + if (res == snd_pcm_lib_buffer_bytes(substream)) + res = 0; + } + + return bytes_to_frames(substream->runtime, res); +} + +static int s3c_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); + + prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + return 0; +} + +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; + + pr_debug("Entered %s\n", __func__); + + if (!prtd) + pr_debug("s3c_dma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static int s3c_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + pr_debug("Entered %s\n", __func__); + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .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 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 = s3c_dma_hardware.buffer_bytes_max; + + pr_debug("Entered %s\n", __func__); + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + pr_debug("Entered %s\n", __func__); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 s3c_dma_mask = DMA_BIT_MASK(32); + +static int s3c_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s3c_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->driver->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->driver->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } +out: + return ret; +} + +static struct snd_soc_platform_driver s3c24xx_soc_platform = { + .ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; + +static int __devinit s3c24xx_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &s3c24xx_soc_platform); +} + +static int __devexit s3c24xx_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver s3c24xx_pcm_driver = { + .driver = { + .name = "samsung-audio", + .owner = THIS_MODULE, + }, + + .probe = s3c24xx_soc_platform_probe, + .remove = __devexit_p(s3c24xx_soc_platform_remove), +}; + +static int __init snd_s3c24xx_pcm_init(void) +{ + return platform_driver_register(&s3c24xx_pcm_driver); +} +module_init(snd_s3c24xx_pcm_init); + +static void __exit snd_s3c24xx_pcm_exit(void) +{ + platform_driver_unregister(&s3c24xx_pcm_driver); +} +module_exit(snd_s3c24xx_pcm_exit); + +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:samsung-audio"); diff --git a/sound/soc/s3c24xx/dma.h b/sound/soc/s3c24xx/dma.h new file mode 100644 index 0000000..f8cd2b4 --- /dev/null +++ b/sound/soc/s3c24xx/dma.h @@ -0,0 +1,30 @@ +/* + * dma.h -- + * + * 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. + * + * ALSA PCM interface for the Samsung S3C24xx CPU + */ + +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +struct s3c_dma_params { + struct s3c2410_dma_client *client; /* stream identifier */ + int channel; /* Channel ID */ + dma_addr_t dma_addr; + int dma_size; /* Size of the DMA transfer */ +}; + +#define S3C24XX_DAI_I2S 0 + +/* platform data */ +extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; + +#endif diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c index c568f69..201056c 100644 --- a/sound/soc/s3c24xx/goni_wm8994.c +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -25,7 +25,7 @@ #include <linux/mfd/wm8994/core.h> #include <linux/mfd/wm8994/registers.h> #include "../codecs/wm8994.h" -#include "s3c-dma.h" +#include "dma.h" #include "s3c64xx-i2s.h" #define MACHINE_NAME 0 diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 7a6b0fa..4e1b8ac 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -25,7 +25,7 @@ #include <asm/mach-types.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c2412-i2s.h" #include "../codecs/wm8750.h" diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 6b7bb38..36e7e85 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -23,7 +23,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c-ac97.h" static struct snd_soc_card ln2440sbc; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index a9a4bbb..c4b2013 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <mach/gta02.h> #include "../codecs/wm8753.h" -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973_gta02; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index d5e4148..96dda57 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -36,7 +36,7 @@ #include "../codecs/wm8753.h" #include "lm4857.h" -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" /* define the scenarios */ diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c index 99bb86e..07197ee 100644 --- a/sound/soc/s3c24xx/rx1950_uda1380.c +++ b/sound/soc/s3c24xx/rx1950_uda1380.c @@ -35,7 +35,7 @@ #include <asm/mach-types.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" #include "../codecs/uda1380.h" diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c index f891eb7..408f9c9 100644 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -24,7 +24,7 @@ #include <mach/dma.h> #include <plat/audio.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c-ac97.h" #define AC_CMD_ADDR(x) (x << 16) diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c deleted file mode 100644 index 0c1cd6c..0000000 --- a/sound/soc/s3c24xx/s3c-dma.c +++ /dev/null @@ -1,503 +0,0 @@ -/* - * s3c-dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License 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/init.h> -#include <linux/io.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "s3c-dma.h" - -static const struct snd_pcm_hardware s3c_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct s3c24xx_runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_limit; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -/* s3c_dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. -*/ -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; - unsigned int limit; - int ret; - - pr_debug("Entered %s\n", __func__); - - if (s3c_dma_has_circular()) - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - else - limit = prtd->dma_limit; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - while (prtd->dma_loaded < limit) { - unsigned long len = prtd->dma_period; - - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + len) > prtd->dma_end) { - len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); - } - - ret = s3c2410_dma_enqueue(prtd->params->channel, - substream, pos, len); - - if (ret == 0) { - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } else - break; - } - - prtd->dma_pos = pos; -} - -static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, - void *dev_id, int size, - enum s3c2410_dma_buffresult result) -{ - struct snd_pcm_substream *substream = dev_id; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) - return; - - prtd = substream->runtime->private_data; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { - prtd->dma_loaded--; - s3c_dma_enqueue(substream); - } - - spin_unlock(&prtd->lock); -} - -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; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - int ret = 0; - - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - ret = s3c2410_dma_request(prtd->params->channel, - prtd->params->client, NULL); - - if (ret < 0) { - printk(KERN_ERR "failed to get dma channel\n"); - return ret; - } - - /* use the circular buffering if we have it available. */ - if (s3c_dma_has_circular()) - s3c2410_dma_setflags(prtd->params->channel, - S3C2410_DMAF_CIRCULAR); - } - - s3c2410_dma_set_buffdone_fn(prtd->params->channel, - s3c24xx_audio_buffdone); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_limit = runtime->hw.periods_min; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int s3c_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - /* TODO - do we need to ensure DMA flushed */ - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - s3c2410_dma_free(prtd->params->channel, prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int s3c_dma_prepare(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, - prtd->params->dma_addr); - } else { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, - prtd->params->dma_addr); - } - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size); - - /* flush the DMA channel */ - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - s3c_dma_enqueue(substream); - - return ret; -} - -static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -s3c_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - unsigned long res; - dma_addr_t src, dst; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - s3c2410_dma_getposition(prtd->params->channel, &src, &dst); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - res = dst - prtd->dma_start; - else - res = src - prtd->dma_start; - - spin_unlock(&prtd->lock); - - pr_debug("Pointer %x %x\n", src, dst); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * callled... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int s3c_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); - - prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -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; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("s3c_dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int s3c_dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops s3c_dma_ops = { - .open = s3c_dma_open, - .close = s3c_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .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 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 = s3c_dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 s3c_dma_mask = DMA_BIT_MASK(32); - -static int s3c_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->driver->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->driver->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static struct snd_soc_platform_driver s3c24xx_soc_platform = { - .ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; - -static int __devinit s3c24xx_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &s3c24xx_soc_platform); -} - -static int __devexit s3c24xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver s3c24xx_pcm_driver = { - .driver = { - .name = "samsung-audio", - .owner = THIS_MODULE, - }, - - .probe = s3c24xx_soc_platform_probe, - .remove = __devexit_p(s3c24xx_soc_platform_remove), -}; - -static int __init snd_s3c24xx_pcm_init(void) -{ - return platform_driver_register(&s3c24xx_pcm_driver); -} -module_init(snd_s3c24xx_pcm_init); - -static void __exit snd_s3c24xx_pcm_exit(void) -{ - platform_driver_unregister(&s3c24xx_pcm_driver); -} -module_exit(snd_s3c24xx_pcm_exit); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:samsung-audio"); diff --git a/sound/soc/s3c24xx/s3c-dma.h b/sound/soc/s3c24xx/s3c-dma.h deleted file mode 100644 index 748c07d..0000000 --- a/sound/soc/s3c24xx/s3c-dma.h +++ /dev/null @@ -1,30 +0,0 @@ -/* - * s3c-dma.h -- - * - * 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. - * - * ALSA PCM interface for the Samsung S3C24xx CPU - */ - -#ifndef _S3C_AUDIO_H -#define _S3C_AUDIO_H - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ - int channel; /* Channel ID */ - dma_addr_t dma_addr; - int dma_size; /* Size of the DMA transfer */ -}; - -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index b3866d5..c471431 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -28,7 +28,7 @@ #include "regs-i2s-v2.h" #include "s3c-i2s-v2.h" -#include "s3c-dma.h" +#include "dma.h" #undef S3C_IIS_V2_SUPPORTED diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index 2e020e1..e111d23 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -29,7 +29,7 @@ #include <plat/audio.h> #include <plat/dma.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c-pcm.h" static struct s3c2410_dma_client s3c_pcm_dma_client_out = { diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 4a861cf..d953ff4 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -35,7 +35,7 @@ #include <mach/regs-gpio.h> #include <mach/dma.h> -#include "s3c-dma.h" +#include "dma.h" #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index e060daa..13e41ed 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -38,7 +38,7 @@ #include <plat/regs-iis.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" static struct s3c2410_dma_client s3c24xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index c4c1114..3f052a5 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -21,7 +21,7 @@ #include <plat/audio-simtec.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index 5180c2a..8b246ab 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index 7a7bb53..a922e1e 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index 50d44fa..87eeb46 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -24,7 +24,7 @@ #include <plat/regs-iis.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c24xx-i2s.h" #include "../codecs/uda134x.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c index a962847..46b65d7 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c @@ -21,7 +21,7 @@ #include <mach/map.h> #include <mach/dma.h> -#include "s3c-dma.h" +#include "dma.h" #include "regs-i2s-v2.h" #include "s3c64xx-i2s.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ae7acb6..0288d4e 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -25,7 +25,7 @@ #include <mach/map.h> #include <mach/dma.h> -#include "s3c-dma.h" +#include "dma.h" #include "regs-i2s-v2.h" #include "s3c64xx-i2s.h" diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c index 863631a..1d55312 100644 --- a/sound/soc/s3c24xx/smartq_wm8987.c +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -24,7 +24,7 @@ #include <asm/mach-types.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c64xx-i2s.h" #include "../codecs/wm8750.h" diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 911bb60..c50d19c 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -19,7 +19,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c-ac97.h" static struct snd_soc_card smdk2443; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index 7e75c8d..9ddc964 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -19,7 +19,7 @@ #include <sound/soc-dapm.h> #include "../codecs/wm8580.h" -#include "s3c-dma.h" +#include "dma.h" #include "s3c64xx-i2s.h" /* diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c index 082b88d..4fc6a9f 100644 --- a/sound/soc/s3c24xx/smdk_spdif.c +++ b/sound/soc/s3c24xx/smdk_spdif.c @@ -18,7 +18,7 @@ #include <sound/soc.h> -#include "s3c-dma.h" +#include "dma.h" #include "spdif.h" /* Audio clock settings are belonged to board specific part. Every diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index ea96a51..80f2aef 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -15,7 +15,7 @@ #include <linux/device.h> #include <sound/soc.h> -#include "s3c-dma.h" +#include "dma.h" #include "s3c-ac97.h" static struct snd_soc_card smdk; diff --git a/sound/soc/s3c24xx/spdif.c b/sound/soc/s3c24xx/spdif.c index ce554e9..dc85df3 100644 --- a/sound/soc/s3c24xx/spdif.c +++ b/sound/soc/s3c24xx/spdif.c @@ -20,7 +20,7 @@ #include <plat/audio.h> #include <mach/dma.h> -#include "s3c-dma.h" +#include "dma.h" #include "spdif.h" /* Registers */ -- 1.6.2.5

1 0

[alsa-devel] [PATCH 03/25] ASoC: Samsung: Rename DMA device
by Jassi Brar 19 Oct '10

19 Oct '10

From: Jassi Brar <jassi.brar(a)samsung.com> Some Samsung SoCs have a PCM(DSP) controller. So the name s3c24xx-pcm-audio for DMA driver is not very appropraite. This patch moves :- s3c24xx-pcm-audio -> samsung-audio Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/mach-s3c64xx/dev-audio.c | 2 +- arch/arm/plat-s3c24xx/devs.c | 2 +- sound/soc/s3c24xx/goni_wm8994.c | 4 ++-- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 4 ++-- sound/soc/s3c24xx/neo1973_wm8753.c | 4 ++-- sound/soc/s3c24xx/rx1950_uda1380.c | 2 +- sound/soc/s3c24xx/s3c-dma.c | 4 ++-- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/smartq_wm8987.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 4 ++-- sound/soc/s3c24xx/smdk_spdif.c | 2 +- sound/soc/s3c24xx/smdk_wm9713.c | 2 +- 17 files changed, 22 insertions(+), 22 deletions(-) diff --git a/arch/arm/mach-s3c64xx/dev-audio.c b/arch/arm/mach-s3c64xx/dev-audio.c index 3838335..c45cc37 100644 --- a/arch/arm/mach-s3c64xx/dev-audio.c +++ b/arch/arm/mach-s3c64xx/dev-audio.c @@ -340,7 +340,7 @@ void __init s3c64xx_ac97_setup_gpio(int num) static u64 s3c_device_audio_dmamask = 0xffffffffUL; struct platform_device s3c_device_pcm = { - .name = "s3c24xx-pcm-audio", + .name = "samsung-audio", .id = -1, .dev = { .dma_mask = &s3c_device_audio_dmamask, diff --git a/arch/arm/plat-s3c24xx/devs.c b/arch/arm/plat-s3c24xx/devs.c index 2f91057..4bf0b39 100644 --- a/arch/arm/plat-s3c24xx/devs.c +++ b/arch/arm/plat-s3c24xx/devs.c @@ -264,7 +264,7 @@ EXPORT_SYMBOL(s3c_device_iis); static u64 s3c_device_audio_dmamask = 0xffffffffUL; struct platform_device s3c_device_pcm = { - .name = "s3c24xx-pcm-audio", + .name = "samsung-audio", .id = -1, .dev = { .dma_mask = &s3c_device_audio_dmamask, diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c index ef22f14..c568f69 100644 --- a/sound/soc/s3c24xx/goni_wm8994.c +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -251,7 +251,7 @@ static struct snd_soc_dai_link goni_dai[] = { .stream_name = "WM8994 HiFi", .cpu_dai_name = "s3c64xx-i2s-v4", .codec_dai_name = "wm8994-hifi", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8994-codec.0-0x1a", .init = goni_wm8994_init, .ops = &goni_hifi_ops, @@ -260,7 +260,7 @@ static struct snd_soc_dai_link goni_dai[] = { .stream_name = "Voice", .cpu_dai_name = "goni-voice-dai", .codec_dai_name = "wm8994-voice", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8994-codec.0-0x1a", .ops = &goni_voice_ops, }, diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 49605cd..7a6b0fa 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -141,7 +141,7 @@ static struct snd_soc_dai_link jive_dai = { .stream_name = "WM8750", .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8750-codec.0-0x1a", .init = jive_wm8750_init, .ops = &jive_ops, diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index abe64ab..6b7bb38 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { .cpu_dai_name = "s3c-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index c457bfd..a9a4bbb 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -400,7 +400,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8753-codec.0-0x1a", .ops = &neo1973_gta02_hifi_ops, }, @@ -411,7 +411,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, .codec_name = "wm8753-codec.0-0x1a", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index d7a39a0..d5e4148 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -556,7 +556,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", .stream_name = "WM8753 HiFi", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", .codec_name = "wm8753-codec.0-0x1a", @@ -566,7 +566,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753-codec.0-0x1a", diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c index ffd5cf2..99bb86e 100644 --- a/sound/soc/s3c24xx/rx1950_uda1380.c +++ b/sound/soc/s3c24xx/rx1950_uda1380.c @@ -96,7 +96,7 @@ static struct snd_soc_dai_link rx1950_uda1380_dai[] = { .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "uda1380-hifi", .init = rx1950_uda1380_init, - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "uda1380-codec.0-001a", .ops = &rx1950_ops, }, diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 54bff83..0c1cd6c 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -477,7 +477,7 @@ static int __devexit s3c24xx_soc_platform_remove(struct platform_device *pdev) static struct platform_driver s3c24xx_pcm_driver = { .driver = { - .name = "s3c24xx-pcm-audio", + .name = "samsung-audio", .owner = THIS_MODULE, }, @@ -500,4 +500,4 @@ module_exit(snd_s3c24xx_pcm_exit); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:s3c24xx-pcm-audio"); +MODULE_ALIAS("platform:samsung-audio"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index f884537..5180c2a 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -99,7 +99,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = { .codec_name = "tlv320aic3x-codec.0-0x1a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .init = simtec_hermes_init, }; diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index c096759..7a7bb53 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -88,7 +88,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = { .codec_name = "tlv320aic3x-codec.0-0x1a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .init = simtec_tlv320aic23_init, }; diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index bd48ffb..50d44fa 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -231,7 +231,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-i2s", .ops = &s3c24xx_uda134x_ops, - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", }; static struct snd_soc_card snd_soc_s3c24xx_uda134x = { diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c index dd20ca7..863631a 100644 --- a/sound/soc/s3c24xx/smartq_wm8987.c +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -213,7 +213,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .stream_name = "SmartQ Hi-Fi", .cpu_dai_name = "s3c64xx-i2s.0", .codec_dai_name = "wm8750-hifi", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8750-codec.0-0x1a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 4613288..911bb60 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -31,7 +31,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { .cpu_dai_name = "s3c-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index 052e499..7e75c8d 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -224,7 +224,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { .stream_name = "Playback", .cpu_dai_name = "s3c64xx-iis-v4", .codec_dai_name = "wm8580-hifi-playback", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8580-codec.0-001b", .init = smdk64xx_wm8580_init_paifrx, .ops = &smdk64xx_ops, @@ -234,7 +234,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { .stream_name = "Capture", .cpu_dai_name = "s3c64xx-iis-v4", .codec_dai_name = "wm8580-hifi-capture", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .codec_name = "wm8580-codec.0-001b", .init = smdk64xx_wm8580_init_paiftx, .ops = &smdk64xx_ops, diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c index f31d22a..082b88d 100644 --- a/sound/soc/s3c24xx/smdk_spdif.c +++ b/sound/soc/s3c24xx/smdk_spdif.c @@ -157,7 +157,7 @@ static struct snd_soc_card smdk; static struct snd_soc_dai_link smdk_dai = { .name = "S/PDIF", .stream_name = "S/PDIF PCM Playback", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .cpu_dai_name = "samsung-spdif", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 33ba8fd..ea96a51 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -45,7 +45,7 @@ static struct snd_soc_card smdk; static struct snd_soc_dai_link smdk_dai = { .name = "AC97", .stream_name = "AC97 PCM", - .platform_name = "s3c24xx-pcm-audio", + .platform_name = "samsung-audio", .cpu_dai_name = "s3c-ac97", .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", -- 1.6.2.5

1 0

[alsa-devel] [PATCH 02/25] ASoC: Samsung: Remove redundant AQUILA driver
by Jassi Brar 19 Oct '10

19 Oct '10

From: Jassi Brar <jassi.brar(a)samsung.com> AQUILA and GONI are essentially the same h/w w.r.t ASoC. They only differ by the fact that GONI has stereo speaker-out whereas AQUILA has mono. The difference can easily be handled in the same MACHINE driver by making machine-specific runtime changes. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 17 +-- sound/soc/s3c24xx/Makefile | 4 +- sound/soc/s3c24xx/aquila_wm8994.c | 295 ------------------------------------- sound/soc/s3c24xx/goni_wm8994.c | 19 +++- 4 files changed, 23 insertions(+), 312 deletions(-) delete mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8a6b53c..6efdf65 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -144,22 +144,13 @@ config SND_S3C64XX_SOC_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 +config SND_SOC_GONI_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" + depends on SND_S3C24XX_SOC && (MACH_GONI || MACH_AQUILA) select SND_S3C64XX_SOC_I2S_V4 select SND_SOC_WM8994 help - Say Y if you want to add support for SoC audio on aquila - with the WM8994. - -config SND_S5PV210_SOC_GONI_WM8994 - tristate "SoC I2S Audio support for GONI - WM8994" - depends on SND_S3C24XX_SOC && MACH_GONI - select SND_S3C64XX_SOC_I2S_V4 - select SND_SOC_WM8994 - help - Say Y if you want to add support for SoC audio on goni + Say Y if you want to add support for SoC audio on goni or aquila with the WM8994. config SND_SOC_SMDK_SPDIF diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index ee8f41d..4e232f1 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -33,7 +33,6 @@ snd-soc-rx1950-uda1380-objs := rx1950_uda1380.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 snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o @@ -50,6 +49,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(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-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o -obj-$(CONFIG_SND_S5PV210_SOC_GONI_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_SPDIF) += snd-soc-smdk-spdif.o +obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c deleted file mode 100644 index 235d197..0000000 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ /dev/null @@ -1,295 +0,0 @@ -/* - * 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 "s3c64xx-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_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->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->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->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, S3C64XX_CLKSRC_PCLK, - 0, SND_SOC_CLOCK_IN); - 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->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_driver voice_dai = { - .name = "aquila-voice-dai", - .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", - .cpu_dai_name = "s3c64xx-i2s-v4", - .codec_dai_name = "wm8994-hifi", - .platform_name = "s3c24xx-pcm-audio", - .codec_name = "wm8994-codec.0-0x1a", - .init = aquila_wm8994_init, - .ops = &aquila_hifi_ops, -}, { - .name = "WM8994 Voice", - .stream_name = "Voice", - .cpu_dai_name = "aquila-voice-dai", - .codec_dai_name = "wm8994-voice", - .platform_name = "s3c24xx-pcm-audio", - .codec_name = "wm8994-codec.0-0x1a", - .ops = &aquila_voice_ops, -}, -}; - -static struct snd_soc_card aquila = { - .name = "aquila", - .dai_link = aquila_dai, - .num_links = ARRAY_SIZE(aquila_dai), -}; - -static int __init aquila_init(void) -{ - int ret; - - if (!machine_is_aquila()) - return -ENODEV; - - aquila_snd_device = platform_device_alloc("soc-audio", -1); - if (!aquila_snd_device) - return -ENOMEM; - - /* register voice DAI here */ - ret = snd_soc_register_dai(&aquila_snd_device->dev, &voice_dai); - if (ret) - return ret; - - platform_set_drvdata(aquila_snd_device, &aquila); - 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"); diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c index 694f702..ef22f14 100644 --- a/sound/soc/s3c24xx/goni_wm8994.c +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -28,6 +28,14 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" +#define MACHINE_NAME 0 +#define CPU_VOICE_DAI 1 + +static const char *aquila_str[] = { + [MACHINE_NAME] = "aquila", + [CPU_VOICE_DAI] = "aquila-voice-dai", +}; + static struct snd_soc_card goni; static struct platform_device *goni_snd_device; @@ -115,6 +123,11 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + if (machine_is_aquila()) { + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + } + snd_soc_dapm_sync(codec); /* Headset jack detection */ @@ -263,7 +276,11 @@ static int __init goni_init(void) { int ret; - if (!machine_is_goni()) + if (machine_is_aquila()) { + voice_dai.name = aquila_str[CPU_VOICE_DAI]; + goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI]; + goni.name = aquila_str[MACHINE_NAME]; + } else if (!machine_is_goni()) return -ENODEV; goni_snd_device = platform_device_alloc("soc-audio", -1); -- 1.6.2.5

1 0

[alsa-devel] emachines em350 quirk
by Eliot Blennerhassett 19 Oct '10

19 Oct '10

Greetings, I have this emachines em350 notebook, which is equipped with ALC272X codec (a personal project this time...) By default the internal mic does not work. (See also https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/639846) Using hda_analyzer, The internal mic appears to be a digital mic on NID 12 This change enables it: Diff for codec 0/0 (0x10ec0272): --- +++ @@ -312,12 +312,12 @@ 0x0c 0x0d* 0x0e Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] Connection: 10 0x18 0x19 0x1a 0x1b 0x1d 0x14 0x15 0x16 0x0b 0x12 Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1d 0x14 0x15 0x16 0x0b 0x13 According to the ALC272 datasheet, there is no feedthrough path from DMIC to analog out. Also no Some of the other controls for this netbook are also superfluous. It only has (as far as I can make out) Internal speaker on NID 0x14 (LOUT1 PORT-D) Internal mic: DMIC on NID 0x12 (DMIC-1/2) External headphone on NID 0x21 (HP-OUT PORT-I) External mic on NID 0x18 (MIC1 PORT-B) So, I'm looking for either an existing model=X to use with this machine, or how to create a new one? regards Eliot

1 0

Re: [alsa-devel] [PATCH] sst: Intel SST audio driver
by Mark Brown 19 Oct '10

19 Oct '10

On Sun, Oct 03, 2010 at 11:22:44AM +0100, Alan Cox wrote: > > Having things as a series does make things easier for review, reducing > > reviewer fatigue if nothing else. I've given this a bit of a once > > over below but not a deep dive - splitting the series would really > > help with reviewability. > It really wouldn't help in this case for most people I think. You end up > with what there is in the original which is a series of patches for old > code all quite large and just a big one chopped into chunks, followed > by a series of updates that fix half the bugs you found reviewing the > first big splat. IIRC at least one version had a split where the ALSA integration stuff was separated out from the underlying DSP interface code - that was pretty helpful since it helps focus on the ALSA specifics. > I did think about it, but what history we have is fairly basic because > it was basically written then fired in my direction to help sort out Yeah, I wasn't thinking about history so much as about helping break down the areas covered for comprehensibility. > > To be honest all this stuff looks sufficiently generic that it might > > be worth considering factoring out an abstraction which can be used by > > other offload engines - having a standard kernel API for them would > > be a real win. That's just a nice to have, though. > Agreed, although gstreamer is pretty good at that it would save work if > it can be partly generic. It's not trivial however because the offload Plus the fact that not everyone is using gstreamer at the application level :/ > interface with suitable firmware loaded does things other than PCM and > you have very firmware specific interfaces for configuring those. We ought to be able to come up with something for the core streaming stuff, though. Like I say, it's just a nice to have though. > So are you happy for it to go into staging. I'm hoping that way at > least all the changes will be visible and trackable. I'll start by > sending GregKH a follow up patch which is a TODO list summary based on > your feedback Happy is a strong term but this looks like exactly the sort of stuff staging is there for so yes, it should go in. Probably best to look for feedback from at least Takashi and/or Jaroslav as well, but obviously we can update the TODO later. I do have some nervousness about the concept of staging for embedded stuff since I worry that inclusion in staging can send the wrong message to vendors but that's a completely separate issue to this driver.

5 17

[alsa-devel] [PATCH v3 1/9] davinci: EMAC support for Omapl138-Hawkboard
by vm.rod25＠gmail.com 19 Oct '10

19 Oct '10

From: Victor Rodriguez <vm.rod25(a)gmail.com> This patch adds EMAC support for the Hawkboard-L138 system Signed-off-by: Victor Rodriguez <victor.rodriguez(a)sasken.com> --- arch/arm/mach-davinci/board-omapl138-hawk.c | 49 +++++++++++++++++++++++++++ 1 files changed, 49 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/board-omapl138-hawk.c b/arch/arm/mach-davinci/board-omapl138-hawk.c index c472dd8..3ae5178 100644 --- a/arch/arm/mach-davinci/board-omapl138-hawk.c +++ b/arch/arm/mach-davinci/board-omapl138-hawk.c @@ -19,6 +19,53 @@ #include <mach/cp_intc.h> #include <mach/da8xx.h> +#include <mach/mux.h> + +#define HAWKBOARD_PHY_ID "0:07" + +static short omapl138_hawk_mii_pins[] __initdata = { + DA850_MII_TXEN, DA850_MII_TXCLK, DA850_MII_COL, DA850_MII_TXD_3, + DA850_MII_TXD_2, DA850_MII_TXD_1, DA850_MII_TXD_0, DA850_MII_RXER, + DA850_MII_CRS, DA850_MII_RXCLK, DA850_MII_RXDV, DA850_MII_RXD_3, + DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, DA850_MDIO_CLK, + DA850_MDIO_D, + -1 +}; + +static int __init omapl138_hawk_config_emac(void) +{ + void __iomem *cfgchip3; + int ret; + u32 val; + struct davinci_soc_info *soc_info = &davinci_soc_info; + + if (!machine_is_omapl138_hawkboard()) + return 0; + + cfgchip3 = DA8XX_SYSCFG0_VIRT(DA8XX_CFGCHIP3_REG); + + val = __raw_readl(cfgchip3); + + val &= ~BIT(8); + ret = davinci_cfg_reg_list(omapl138_hawk_mii_pins); + pr_info("EMAC: MII PHY configured\n"); + + if (ret) + pr_warning("%s: " + "cpgmac/mii mux setup failed: %d\n", __func__, ret); + + /* configure the CFGCHIP3 register for MII */ + __raw_writel(val, cfgchip3); + + soc_info->emac_pdata->phy_id = HAWKBOARD_PHY_ID; + + ret = da8xx_register_emac(); + if (ret) + pr_warning("%s: " + "emac registration failed: %d\n", __func__, ret); + return 0; +} + static struct davinci_uart_config omapl138_hawk_uart_config __initdata = { .enabled_uarts = 0x7, @@ -30,6 +77,8 @@ static __init void omapl138_hawk_init(void) davinci_serial_init(&omapl138_hawk_uart_config); + ret = omapl138_hawk_config_emac(); + ret = da8xx_register_watchdog(); if (ret) pr_warning("omapl138_hawk_init: " -- 1.6.0.5

3 4

[alsa-devel] [PATCH 1/2] sound/soc/davinci/davinci-mcasp.c: Return error code in failure
by Julia Lawall 19 Oct '10

19 Oct '10

In this code, 0 is returned on failure, even though other failures return -ENOMEM or other similar values. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @a@ identifier alloc; identifier ret; constant C; expression x; @@ x = alloc(...); if (x == NULL) { <+... $ret = -C; \| return -C; $ ...+> } @@ identifier f, a.alloc; expression ret; expression x,e1,e2,e3; @@ ret = 0 ... when != ret = e1 *x = alloc(...) ... when != ret = e2 if (x == NULL) { ... when != ret = e3 return ret; } // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- Another call to platform_get_resource in the same function returns -ENODEV on error, so I have used that value. sound/soc/davinci/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c8e97dc..86918ee 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -898,6 +898,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENODEV; goto err_release_region; } @@ -912,6 +913,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENODEV; goto err_release_region; }

3 2

[alsa-devel] alsa-utils cross-compilation problem
by Reena Chauhan 19 Oct '10

19 Oct '10

hi I had tried to cross compile the Alsa-utils 1.0.23 & m getting following errors: checking for libasound headers version >= 1.0.16... not present. configure: error: Sufficiently new version of libasound not found In my host PC alsa-lib 1.0.22 is installed -- Reena Chauhan Scientist-C Application & Baseband Platform, SBSG ,DEAL,Raipur Road Dehradun-248001

2 1

[alsa-devel] [PATCH] Per Application Volume Control plugin for ALSA
by James Huk 18 Oct '10

18 Oct '10

Hello everybody. This is my first "post" here, so please be "gentle". I have created PAVC plugin for ALSA, it is based on COPY plugin, and use SOFTVOL plugin for sound output. As far as I can tell from current tests - It works perfectly, and it doesn't break compatibility with any apps I use (and PulseAudio does). So I was wondering - is there any chance this plugin would be added to the next official ALSA release? Here is the patch that enables it on current alsa-lib-1.0.23 (hopefully this mailing list is a good place to "post" this): --- a/configure.in 2010-09-23 17:03:00.000000000 +0200 +++ b/configure.in 2010-09-23 17:09:11.029900400 +0200 @@ -445,7 +445,7 @@ pcm_plugins="" fi -PCM_PLUGIN_LIST="copy linear route mulaw alaw adpcm rate plug multi shm file null empty share meter hooks lfloat ladspa dmix dshare dsnoop asym iec958 softvol extplug ioplug mmap_emul" +PCM_PLUGIN_LIST="copy linear route mulaw alaw adpcm rate plug multi shm file null empty share meter hooks lfloat ladspa dmix dshare dsnoop asym iec958 softvol extplug ioplug mmap_emul pavc" build_pcm_plugin="no" for t in $PCM_PLUGIN_LIST; do @@ -516,6 +516,7 @@ AM_CONDITIONAL(BUILD_PCM_PLUGIN_EXTPLUG, test x$build_pcm_extplug = xyes) AM_CONDITIONAL(BUILD_PCM_PLUGIN_IOPLUG, test x$build_pcm_ioplug = xyes) AM_CONDITIONAL(BUILD_PCM_PLUGIN_MMAP_EMUL, test x$build_pcm_mmap_emul = xyes) +AM_CONDITIONAL(BUILD_PCM_PLUGIN_PAVC, test x$build_pcm_pavc = xyes) dnl Defines for plug plugin if test "$build_pcm_rate" = "yes"; then --- a/src/pcm/Makefile.am 2010-04-16 13:11:05.000000000 +0200 +++ b/src/pcm/Makefile.am 2010-09-23 17:20:10.601899695 +0200 @@ -102,6 +102,9 @@ if BUILD_PCM_PLUGIN_MMAP_EMUL libpcm_la_SOURCES += pcm_mmap_emul.c endif +if BUILD_PCM_PLUGIN_PAVC +libpcm_la_SOURCES += pcm_pavc.c +endif EXTRA_DIST = pcm_dmix_i386.c pcm_dmix_x86_64.c pcm_dmix_generic.c --- a/src/pcm/pcm.c 2010-04-16 13:11:05.000000000 +0200 +++ b/src/pcm/pcm.c 2010-09-23 17:10:43.813900406 +0200 @@ -2047,7 +2047,7 @@ static const char *const build_in_pcms[] = { "adpcm", "alaw", "copy", "dmix", "file", "hooks", "hw", "ladspa", "lfloat", "linear", "meter", "mulaw", "multi", "null", "empty", "plug", "rate", "route", "share", - "shm", "dsnoop", "dshare", "asym", "iec958", "softvol", "mmap_emul", + "shm", "dsnoop", "dshare", "asym", "iec958", "softvol", "mmap_emul","pavc", NULL }; --- a/include/pcm.h 2010-04-16 13:11:05.000000000 +0200 +++ b/include/pcm.h 2010-09-23 17:11:58.389899158 +0200 @@ -376,7 +376,9 @@ SND_PCM_TYPE_EXTPLUG, /** Mmap-emulation plugin */ SND_PCM_TYPE_MMAP_EMUL, - SND_PCM_TYPE_LAST = SND_PCM_TYPE_MMAP_EMUL + /** PAVC plugin */ + SND_PCM_TYPE_PAVC, + SND_PCM_TYPE_LAST = SND_PCM_TYPE_PAVC }; /** PCM type */ --- a/src/pcm/pcm_pavc.c +++ b/src/pcm/pcm_pavc.c @@ -0,0 +1,846 @@ +/* + * PCM - PAVC plugin + * Based on PCM - Copy conversion plugin by Abramo Bagnara <abramo(a)alsa-project.org> + * + * Copyright (c) 2010 by Adrian Kobyliński <huk256(a)gmail.com> + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <byteswap.h> +#include "pcm_local.h" +#include "pcm_plugin.h" +#include <stdio.h> +#include <string.h> +#include <unistd.h> +#include <sys/wait.h> + + +#ifndef PIC +/* entry for static linking */ +const char *_snd_module_pcm_pavc = ""; +#endif + +#ifndef DOC_HIDDEN +typedef struct { + /* This field need to be the first */ + snd_pcm_plugin_t plug; +} snd_pcm_pavc_t; +#endif + +static int snd_pcm_pavc_hw_refine_cprepare(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params) +{ + int err; + snd_pcm_access_mask_t access_mask = { SND_PCM_ACCBIT_SHM }; + err = _snd_pcm_hw_param_set_mask(params, SND_PCM_HW_PARAM_ACCESS, + &access_mask); + if (err < 0) + return err; + params->info &= ~(SND_PCM_INFO_MMAP | SND_PCM_INFO_MMAP_VALID); + return 0; +} + +static int snd_pcm_pavc_hw_refine_sprepare(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *sparams) +{ + snd_pcm_access_mask_t saccess_mask = { SND_PCM_ACCBIT_MMAP }; + _snd_pcm_hw_params_any(sparams); + _snd_pcm_hw_param_set_mask(sparams, SND_PCM_HW_PARAM_ACCESS, + &saccess_mask); + return 0; +} + +static int snd_pcm_pavc_hw_refine_schange(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params, + snd_pcm_hw_params_t *sparams) +{ + int err; + unsigned int links = ~SND_PCM_HW_PARBIT_ACCESS; + err = _snd_pcm_hw_params_refine(sparams, links, params); + if (err < 0) + return err; + return 0; +} + +static int snd_pcm_pavc_hw_refine_cchange(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params, + snd_pcm_hw_params_t *sparams) +{ + int err; + unsigned int links = ~SND_PCM_HW_PARBIT_ACCESS; + err = _snd_pcm_hw_params_refine(params, links, sparams); + if (err < 0) + return err; + return 0; +} + +static int snd_pcm_pavc_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) +{ + return snd_pcm_hw_refine_slave(pcm, params, + snd_pcm_pavc_hw_refine_cprepare, + snd_pcm_pavc_hw_refine_cchange, + snd_pcm_pavc_hw_refine_sprepare, + snd_pcm_pavc_hw_refine_schange, + snd_pcm_generic_hw_refine); +} + +static int snd_pcm_pavc_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) +{ + return snd_pcm_hw_params_slave(pcm, params, + snd_pcm_pavc_hw_refine_cchange, + snd_pcm_pavc_hw_refine_sprepare, + snd_pcm_pavc_hw_refine_schange, + snd_pcm_generic_hw_params); +} + +static snd_pcm_uframes_t + snd_pcm_pavc_write_areas(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t size, + const snd_pcm_channel_area_t *slave_areas, + snd_pcm_uframes_t slave_offset, + snd_pcm_uframes_t *slave_sizep) +{ + if (size > *slave_sizep) + size = *slave_sizep; + snd_pcm_areas_copy(slave_areas, slave_offset, + areas, offset, + pcm->channels, size, pcm->format); + *slave_sizep = size; + return size; +} + +static snd_pcm_uframes_t + snd_pcm_pavc_read_areas(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t size, + const snd_pcm_channel_area_t *slave_areas, + snd_pcm_uframes_t slave_offset, + snd_pcm_uframes_t *slave_sizep) +{ + if (size > *slave_sizep) + size = *slave_sizep; + snd_pcm_areas_copy(areas, offset, + slave_areas, slave_offset, + pcm->channels, size, pcm->format); + *slave_sizep = size; + return size; +} + +static void snd_pcm_pavc_dump(snd_pcm_t *pcm, snd_output_t *out) +{ + snd_pcm_pavc_t *pavc = pcm->private_data; + snd_output_printf(out, "Copy conversion PCM\n"); + if (pcm->setup) { + snd_output_printf(out, "Its setup is:\n"); + snd_pcm_dump_setup(pcm, out); + } + snd_output_printf(out, "Slave: "); + snd_pcm_dump(pavc->plug.gen.slave, out); +} + +static const snd_pcm_ops_t snd_pcm_pavc_ops = { + .close = snd_pcm_generic_close, + .info = snd_pcm_generic_info, + .hw_refine = snd_pcm_pavc_hw_refine, + .hw_params = snd_pcm_pavc_hw_params, + .hw_free = snd_pcm_generic_hw_free, + .sw_params = snd_pcm_generic_sw_params, + .channel_info = snd_pcm_generic_channel_info, + .dump = snd_pcm_pavc_dump, + .nonblock = snd_pcm_generic_nonblock, + .async = snd_pcm_generic_async, + .mmap = snd_pcm_generic_mmap, + .munmap = snd_pcm_generic_munmap, +}; + +/** + * \brief Creates a new copy PCM + * \param pcmp Returns created PCM handle + * \param name Name of PCM + * \param slave Slave PCM handle + * \param close_slave When set, the slave PCM handle is closed with copy PCM + * \retval zero on success otherwise a negative error code + * \warning Using of this function might be dangerous in the sense + * of compatibility reasons. The prototype might be freely + * changed in future. + */ +int snd_pcm_pavc_open(snd_pcm_t **pcmp, const char *name, snd_pcm_t *slave, int close_slave) +{ + snd_pcm_t *pcm; + snd_pcm_pavc_t *pavc; + int err; + assert(pcmp && slave); + pavc = calloc(1, sizeof(snd_pcm_pavc_t)); + if (!pavc) { + return -ENOMEM; + } + snd_pcm_plugin_init(&pavc->plug); + pavc->plug.read = snd_pcm_pavc_read_areas; + pavc->plug.write = snd_pcm_pavc_write_areas; + pavc->plug.undo_read = snd_pcm_plugin_undo_read_generic; + pavc->plug.undo_write = snd_pcm_plugin_undo_write_generic; + pavc->plug.gen.slave = slave; + pavc->plug.gen.close_slave = close_slave; + + err = snd_pcm_new(&pcm, SND_PCM_TYPE_PAVC, name, slave->stream, slave->mode); + if (err < 0) { + free(pavc); + return err; + } + pcm->ops = &snd_pcm_pavc_ops; + pcm->fast_ops = &snd_pcm_plugin_fast_ops; + pcm->private_data = pavc; + pcm->poll_fd = slave->poll_fd; + pcm->poll_events = slave->poll_events; + pcm->monotonic = slave->monotonic; + snd_pcm_set_hw_ptr(pcm, &pavc->plug.hw_ptr, -1, 0); + snd_pcm_set_appl_ptr(pcm, &pavc->plug.appl_ptr, -1, 0); + *pcmp = pcm; + + return 0; +} + +int countString(char * x,char x2,int count) +{ + int i; + int z=0; + for(i=0;i<count;++i) + { + if(x[i]==x2) + { + ++z; + } + } + return z; +} + +char *deleteChar(char *x, char x2, int count) +{ + char *ret; + ret=(char*) malloc(count+1); + int i; + for(i=0;i<count;++i) + { + ret[i]=0; + } + for(i=0;i<count;++i) + { + if(x[i]!=x2) + { + ret[i]=x[i]; + } + else + { + ret[i]=' '; + } + } + return ret; +} + +char *getProceses() +{ + FILE *pipe; + if ( !(pipe = (FILE*)popen("lsof -F p /dev/snd/pcmC0D0p","r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + return NULL; + } + + int i=0,j=0; + int c; + + do { + c = fgetc (pipe); + ++i; + } while (c != EOF); + fclose (pipe); + + char *buf=(char*)malloc(sizeof(char)*i); + + for(j=0;j<i;++j) + { + buf[j]=0; + } + + i=0; + + if ( !(pipe = (FILE*)popen("lsof -F p /dev/snd/pcmC0D0p","r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + return NULL; + } + + do { + c = fgetc (pipe); + if(c!=EOF && c!='p' && c!='\n') + { + buf[i]=c; + } + else if(c=='p') + { + buf[i]=' '; + } + else if(c=='\n') + { + buf[i]=','; + } + ++i; + } while (c != EOF); + fclose (pipe); + + return buf; +} + +char *loadPid(char *name) +{ + FILE *file; + int c,i=0,j=0; + + file=fopen(name,"r"); + if(file!=NULL) + { + do{ + c=fgetc(file); + ++i; + }while(c!=EOF); + fclose(file); + + char *buf=malloc((sizeof(char))*i); + for(j=0;j<i;++j) + { + buf[j]=0; + } + + i=0; + + file=fopen(name,"r"); + if(file!=NULL) + { + do{ + c=fgetc(file); + if(c!=EOF && c!='\n') + buf[i]=c; + ++i; + }while(c!=EOF); + fclose(file); + + return buf; + } + else + { + return NULL; + } + + } + else + { + return NULL; + } +} + +void savePid(char *pid,char *file) +{ + int i; + char command3[256]; + + for(i=0;i<256;++i) + { + command3[i]=0; + } + + strcat(command3,"echo "); + strcat(command3,pid); + strcat(command3," > "); + strcat(command3,file); + + for(i=5;i<256;++i) + { + if(command3[i]!=EOF && command3[i]!='\n' && command3[i]!='*') + { + command3[i]=command3[i]; + } + else + { + command3[i]=' '; + } + } + + system(command3); +} + +int checkDir(char *name) +{ + char command[128]; + memset(command,0,128); + + strcat(command,"ls -aF "); + strcat(command,getenv("HOME")); + strcat(command," | grep \\./$ | grep -w "); + strcat(command,name); + + FILE *pipe; + if ( !(pipe = (FILE*)popen(command,"r")) ) + { + perror("Problems with pipe"); + return -1; + } + + int i=0; + int c; + + do { + c = fgetc (pipe); + if(c!=EOF && c!='\n') + ++i; + } while (c != EOF); + fclose (pipe); + + if(i>0) + { + return 1; + } + else + { + return 0; + } +} + +int createDir(char *name) +{ + pid_t PID; + int status; + + PID=fork(); + + if(PID>0) + { + char command[128]; + memset(command,0,128); + + strcat(command,getenv("HOME")); + strcat(command,"/"); + strcat(command,".softvol"); + + execlp("mkdir","mkdir",command,NULL); + } + else + { + pid_t wpid = waitpid(PID, &status, WUNTRACED); + } + + return WEXITSTATUS(status); +} + +char *getDir(char *name) +{ + static char command[128]; + memset(command,0,128); + + strcat(command,getenv("HOME")); + strcat(command,"/"); + strcat(command,name); + + return command; +} + +char *getSoftvolNumber(int channelCount,int current) +{ + size_t i; + static char ret[4]; + memset(ret,0,5); + char buf1[4],buf2[4]; + sprintf(buf1,"%i",channelCount); + sprintf(buf2,"%i",current); + + if(strlen(buf2)<strlen(buf1)) + { + for(i=0;i<(strlen(buf1)-1);++i) + { + ret[i]='0'; + } + + strcat(ret,buf2); + + return ret; + } + else if(strlen(buf2)==strlen(buf1)) + { + strcat(ret,buf2); + return ret; + } + else + { + //this should never happen + return NULL; + } +} + +int checkSoftvolProcess(int channelcount) +{ + int i=0,c=0,ret=channelcount-1,j=0; + char nameBuf[1024]; + char pidBuf[32]; + char numBuf[4]; + char command[128]; + char fileName[64]; + + FILE *fpipe; + + for(i=0;i<channelcount;++i) + { + memset(nameBuf,0,1024); + memset(numBuf,0,4); + memset(command,0,128); + memset(fileName,0,64); + + strcat(command,"ps -A -o pid= | grep -w -f "); + strcat(command,getDir(".softvol")); + strcat(command,"/SOFTVOL"); + strcat(command,getSoftvolNumber(channelcount,i)); + strcat(command,".pid"); + + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + strcat(fileName,getSoftvolNumber(channelcount,i)); + strcat(fileName,".pid"); + + //Here we check if current process is already conected to something + FILE *temp=fopen(fileName,"r"); + if(temp!=NULL) + { + j=0; + do{ + c=fgetc(temp); + if(c!=EOF && c!='\n') + { + nameBuf[j]=c; + } + else + { + nameBuf[j]='\0'; + } + + ++j; + }while(c!=EOF); + } + else + { + memset(nameBuf,0,1024); + } + + sprintf(pidBuf,"%d",getpid()); + + if(strcmp(nameBuf,pidBuf)==0) + { + //Current process wants to re-initialize the sound - so we connect it + //to the output it is currently using + ret=i; + break; + } + else + { + //Current process is not connected to anything - so we connect it + //to the firs unused softvol output + memset(fileName,0,64); + } + } + + if(fileName[0]!=0) + { + //Current process is already using some softvol output - so we + //reconnect it to this output + for(i=0;i<channelcount;++i) + { + memset(numBuf,0,4); + memset(command,0,128); + memset(fileName,0,64); + + strcat(command,"ps -A -o pid= | grep -w -f "); + strcat(command,getDir(".softvol")); + strcat(command,"/SOFTVOL"); + strcat(command,getSoftvolNumber(channelcount,i)); + strcat(command,".pid 2> /dev/null"); + + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + strcat(fileName,getSoftvolNumber(channelcount,i)); + strcat(fileName,".pid"); + + //Here we check if process is alive... + if ( !(fpipe = (FILE*)popen(command,"r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + } + + j=0; + + do { + c = fgetc (fpipe); + if(c!=EOF) + ++j; + } while (c != EOF); + fclose (fpipe); + + //...and if it is the current process + if(j>0) + { + FILE *temp=fopen(fileName,"r"); + j=0; + do{ + c=fgetc(temp); + if(c!=EOF && c!='\n') + { + nameBuf[j]=c; + } + else + { + nameBuf[j]='\0'; + } + + ++j; + }while(c!=EOF); + + char pidBuf[32]; + sprintf(pidBuf,"%d",getpid()); + + if(strcmp(nameBuf,pidBuf)==0) + { + //Current process wants to re-initialize the sound - so we connect it + //to the output it is currently using + ret=i; + break; + } + else + { + //It is some other process - so we connect it + //to the first unused softvol output + ret=i+1; + } + } + else + { + ret=i; + } + } + } + else + { + //Process isn't using any softvol output - so we connect it + //to the first unused on the list + + for(i=0;i<channelcount;++i) + { + memset(numBuf,0,4); + memset(command,0,128); + memset(fileName,0,64); + + strcat(command,"ps -A -o pid= | grep -w -f "); + strcat(command,getDir(".softvol")); + strcat(command,"/SOFTVOL"); + strcat(command,getSoftvolNumber(channelcount,i)); + strcat(command,".pid 2> /dev/null"); + + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + strcat(fileName,getSoftvolNumber(channelcount,i)); + strcat(fileName,".pid"); + + //Here we check if process is alive... + if ( !(fpipe = (FILE*)popen(command,"r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + } + + j=0; + + do { + c = fgetc (fpipe); + if(c!=EOF) + ++j; + } while (c != EOF); + fclose (fpipe); + + //...and if it is the current process + if(j>0) + { + + } + else + { + ret=i; + break; + } + } + } + return ret; +} + +/*! \page pcm_plugins + +\section pcm_plugins_copy Plugin: copy + +This plugin copies samples from master copy PCM to given slave PCM. +The channel count, format and rate must match for both of them. + +\code +pcm.name { + type copy # Copy PCM + slave STR # Slave name + # or + slave { # Slave definition + pcm STR # Slave PCM name + # or + pcm { } # Slave PCM definition + } +} +\endcode + +\subsection pcm_plugins_copy_funcref Function reference + +<UL> + <LI>snd_pcm_copy_open() + <LI>_snd_pcm_copy_open() +</UL> + +*/ + +/** + * \brief Creates a new copy PCM + * \param pcmp Returns created PCM handle + * \param name Name of PCM + * \param root Root configuration node + * \param conf Configuration node with copy PCM description + * \param stream Stream type + * \param mode Stream mode + * \retval zero on success otherwise a negative error code + * \warning Using of this function might be dangerous in the sense + * of compatibility reasons. The prototype might be freely + * changed in future. + */ +int _snd_pcm_pavc_open(snd_pcm_t **pcmp, const char *name, + snd_config_t *root, snd_config_t *conf, + snd_pcm_stream_t stream, int mode) +{ + long channelcount=-1; + int ret=0; + char softvolName[64]; + char pidBuf[32]; + char fileName[64]; + char helpBuf[32]; + + memset(softvolName,0,64); + + snd_config_iterator_t i, next; + int err; + snd_pcm_t *spcm; + snd_config_t *slave = NULL, *sconf; + + snd_config_t *tempConfig; + err=snd_config_search(conf,"channelcount",&tempConfig); + if(err<0) + { + SNDERR("channelcount field not found"); + return -ENOENT; + } + err=snd_config_get_integer(tempConfig,&channelcount); + if(err<0) + { + SNDERR("channelcount vaule must be integer"); + return -EINVAL; + } + + snd_config_t *softvol[channelcount]; + + int counter=0; + + sprintf(softvolName,"softvol%i",counter); + + snd_config_for_each(i, next, conf) + { + snd_config_t *n = snd_config_iterator_entry(i); + const char *id; + if (snd_config_get_id(n, &id) < 0) + continue; + if (snd_pcm_conf_generic_id(id)) + continue; + if (strcmp(id, "slave") == 0) + { + slave = n; + continue; + } + if (strcmp(id, "channelcount") == 0) + { + continue; + } + if((counter<channelcount) && strcmp(id,softvolName)==0) + { + softvol[counter]=n; + ++counter; + sprintf(softvolName,"softvol%i",counter); + continue; + } + SNDERR("Unknown field %s", id); + return -EINVAL; + } + + if (!slave) { + SNDERR("slave is not defined"); + return -EINVAL; + } + + if(checkDir(".softvol")) + { + //Directory exists - we do nothing + } + else + { + //Directory doesn't exists we need to create it + if((err=createDir(".softvol"))!=0) + { + SNDERR("Unable to create ~/.softvol directory - unknown error %i",err); + return err; + } + } + + ret=checkSoftvolProcess(channelcount); + + err = snd_pcm_slave_conf(root, softvol[ret], &sconf, 0); + + if (err < 0) + return err; + err = snd_pcm_open_slave(&spcm, root, sconf, stream, mode, conf); + snd_config_delete(sconf); + if (err < 0) + return err; + err = snd_pcm_pavc_open(pcmp, name, spcm, 1); + if (err < 0) + snd_pcm_close(spcm); + + sprintf(pidBuf,"%d",getpid()); + memset(fileName,0,64); + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + + strcat(fileName,getSoftvolNumber(channelcount,ret)); + strcat(fileName,".pid"); + + savePid(pidBuf,fileName); + + return err; +} +#ifndef DOC_HIDDEN +SND_DLSYM_BUILD_VERSION(_snd_pcm_pavc_open, SND_PCM_DLSYM_VERSION); +#endif To test PAVC, we have to rebuild alsa-lib, and create .asoundrc or /etc/asound.conf that looks like this: pcm.!default { type plug slave.pcm "asymed" } pcm.asymed { type asym playback.pcm "pavcp" capture.pcm "dsnooped" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:0" period_time 0 period_size 256 #buffer_size 4096 periods 128 rate 44100 } } pcm.dsnooped { type dsnoop ipc_key 1026 slave { pcm "hw:0" channels 2 period_size 256 #buffer_size 4096 rate 44100 periods 0 period_time 0 } } pcm.softvol00 { type softvol slave { pcm "dmixer" } control { name "Softvol00" card 0 } } pcm.softvol01 { type softvol slave { pcm "dmixer" } control { name "Softvol01" card 0 } } pcm.softvol02 { type softvol slave { pcm "dmixer" } control { name "Softvol02" card 0 } } pcm.softvol03 { type softvol slave { pcm "dmixer" } control { name "Softvol03" card 0 } } pcm.softvol04 { type softvol slave { pcm "dmixer" } control { name "Softvol04" card 0 } } pcm.softvol05 { type softvol slave { pcm "dmixer" } control { name "Softvol05" card 0 } } pcm.softvol06 { type softvol slave { pcm "dmixer" } control { name "Softvol06" card 0 } } pcm.softvol07 { type softvol slave { pcm "dmixer" } control { name "Softvol07" card 0 } } pcm.softvol08 { type softvol slave { pcm "dmixer" } control { name "Softvol08" card 0 } } pcm.softvol09 { type softvol slave { pcm "dmixer" } control { name "Softvol09" card 0 } } pcm.softvol10 { type softvol slave { pcm "dmixer" } control { name "Softvol10" card 0 } } pcm.softvol11 { type softvol slave { pcm "dmixer" } control { name "Softvol11" card 0 } } ctl.dmixer { type hw card 0 } ctl.dsnooped { type hw card 0 } pcm.pavcp { type pavc slave.pcm "dmixer" channelcount 12 softvol0.pcm "softvol00" softvol1.pcm "softvol01" softvol2.pcm "softvol02" softvol3.pcm "softvol03" softvol4.pcm "softvol04" softvol5.pcm "softvol05" softvol6.pcm "softvol06" softvol7.pcm "softvol07" softvol8.pcm "softvol08" softvol9.pcm "softvol09" softvol10.pcm "softvol10" softvol11.pcm "softvol11" } With this, anything connected to "default" will be automatically redirected to the first unused softvol channel. PIDs of apps that currently use softvol channels are saved to ~/.softvol/SOFTVOLXX.pid, so this can be checked at any time. If some application will try to restart only its sound subsystem, PAVC plugin will check those files to determine if it should be connected to the new softvol channel, or the one it currently uses. I tried to mimic the way this works with OSS4. Current limitations: -Since there is no "mute and slider at the same time" support in SOFTVOL plugin - there is no mute switch for software channels (or maybe there is a way I don't know about?) -Software channels must be named SOFTVOL00,01,02 and so on -SOFTVOL channels have both PLAYBACK and CAPTURE capabilities by default - this makes some mixers go haywire, and display double sliders - maybe there is a way around this I don't know about (like setting PLAYBACK only capability)? -Code of PAVC, could probably look nicer Hopefully someone will take a look at this, of course I am open to suggestions and critics. Best regards. P.S Sorry for my English ;]

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Re: [alsa-devel] USB asynchronous mode feedback format
by lee188＠singnet.com.sg 18 Oct '10

18 Oct '10

--- Julian Scheel <julian(a)jusst.de> wrote: There may also be > >a limitation in that the Same EP number cannot be used > >for both IN and OUT. > > Hm, I'm not sure about this. You think I should move the feedback > pipe into it's own EP? > I will recheck the At91SAM7 specs later, to see if I can find a note > on such a limitation. Hi Julian, You might like to refer to this thread: http://www.avrfreaks.net/index.php?name=PNphpBB2&file=printview&t=88432&sta… As I have found out, it is not necessary to have the same EP number for IN and OUT for rate feedback to work with a Linux or OSX host. This may be a requirement for Vista/Win7 hosts though. We are still doing testing. Alex

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