November 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] OSS: keep index within bounds of midi_devs[]
by Roel Kluin 16 Nov '09

16 Nov '09

When the {orig,midi}_dev equals num_midis, that's one too large already. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- In MIDIbuf_open() sound/oss/midibuf.c:166 if (num_midis > MAX_MIDI_DEV) { printk(KERN_ERR "midi: Too many midi interfaces\n"); num_midis = MAX_MIDI_DEV; } num_midis is at most MAX_MIDI_DEV. `git grep -n MAX_MIDI_DEV' and note that when the index equals num_midis this causes reads/writes outside array bounds for midi_devs, midi2synth, dev_conf and others (below check in patch). sound/oss/midi_synth.c | 2 +- sound/oss/mpu401.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 9e45098..3bc7104 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -426,7 +426,7 @@ midi_synth_open(int dev, int mode) int err; struct midi_input_info *inc; - if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL) + if (orig_dev < 0 || orig_dev >= num_midis || midi_devs[orig_dev] == NULL) return -ENXIO; midi2synth[orig_dev] = dev; diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 734b8f9..0af9d24 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -770,7 +770,7 @@ static int mpu_synth_ioctl(int dev, unsigned int cmd, void __user *arg) midi_dev = synth_devs[dev]->midi_dev; - if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL) + if (midi_dev < 0 || midi_dev >= num_midis || midi_devs[midi_dev] == NULL) return -ENXIO; devc = &dev_conf[midi_dev];

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-kmirror: ALSA kernel mirror repository branch, master now at v1.0.21-334-g6739046
by Takashi Iwai 16 Nov '09

16 Nov '09

At Thu, 12 Nov 2009 10:54:02 +0100 (CET), noreply-git(a)alsa-project.org wrote: > > commit 6739046df36c7adf80c961bcba4870270e66dbf6 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Thu Nov 12 10:15:48 2009 +0100 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Thu Nov 12 10:51:48 2009 +0100 > > ALSA: hda - proc - add support for dynamic controls to mixer<->NID mapping > > This patch adds support for dynamically created controls to proc codec file > (Control: lines). > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> > > commit c45b73bf328cd8ace53cf39994328cf9d6548c4f > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Wed Nov 11 13:43:01 2009 +0100 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Thu Nov 12 10:51:16 2009 +0100 > > ALSA: hda - proc - introduce Control: lines to show mixer<->NID assignment > > This is an initial patch to show universal control<->NID assigment in > proc codec file. The change helps to debug codec related problems. > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I find the second one is the nice hack, but the first (newer) one isn't good since it abuses the subdevice field of the ctl id. It is a part of API/ABI, and if we do any changes the semantics, we should define the changed behavior *beforehand* publicly. Also, both commits give many warnings via checkpatch.pl... So I postpone the merge so far. thanks, Takashi

2 9

[alsa-devel] [PATCH 5/7] S3C AUDIO: Add header to pass platform data to device drivers
by Jassi Brar 16 Nov '09

16 Nov '09

Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c/include/plat/audio.h | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) create mode 100644 arch/arm/plat-s3c/include/plat/audio.h diff --git a/arch/arm/plat-s3c/include/plat/audio.h b/arch/arm/plat-s3c/include/plat/audio.h new file mode 100644 index 0000000..f22d23b --- /dev/null +++ b/arch/arm/plat-s3c/include/plat/audio.h @@ -0,0 +1,17 @@ +/* arch/arm/plat-s3c/include/plat/audio.h + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +/** + * struct s3c_audio_pdata - common platform data for audio device drivers + * @cfg_gpio: Callback function to setup mux'ed pins in I2S/PCM/AC97 mode + */ +struct s3c_audio_pdata { + int (*cfg_gpio)(struct platform_device *); +}; -- 1.6.2.5

3 2

[alsa-devel] [PATCH 2/7] S3C AUDIO: Rename 's3c24xx-pcm' driver to 's3c_audio'
by Jassi Brar 16 Nov '09

16 Nov '09

Making room for namespace for the PCM Controller driver the platform driver(s3c24xx-pcm) has been renamed to SoC agnostic name 's3c-audio'. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Makefile | 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/s3c-audio.c | 480 ++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-audio.h | 31 ++ sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 480 ------------------------ sound/soc/s3c24xx/s3c24xx-pcm.h | 31 -- 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.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 2 +- 20 files changed, 527 insertions(+), 527 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c-audio.c create mode 100644 sound/soc/s3c24xx/s3c-audio.h delete mode 100644 sound/soc/s3c24xx/s3c24xx-pcm.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-pcm.h diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 7790406..3bdb74d 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,5 +1,5 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c24xx-pcm.o +snd-soc-s3c24xx-objs := s3c-audio.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/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 93e6c87..ac3a709 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 "s3c24xx-pcm.h" +#include "s3c-audio.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 12c7148..3896e52 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -24,7 +24,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c24xx-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 26409a9..1d05137 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 "s3c24xx-pcm.h" +#include "s3c-audio.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 77de6c5..9183166 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 "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c24xx-i2s.h" /* define the scenarios */ diff --git a/sound/soc/s3c24xx/s3c-audio.c b/sound/soc/s3c24xx/s3c-audio.c new file mode 100644 index 0000000..cb49400 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-audio.c @@ -0,0 +1,481 @@ +/* + * s3c-audio.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-audio.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; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; + unsigned long totbytes = params_buffer_bytes(params); + 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->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform s3c24xx_soc_platform = { + .name = "s3c24xx-audio", + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); + +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-audio.h b/sound/soc/s3c24xx/s3c-audio.h new file mode 100644 index 0000000..8cbc071 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-audio.h @@ -0,0 +1,31 @@ +/* + * s3c-audio.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_soc_platform s3c24xx_soc_platform; +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 5a442aa..13c99ab 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -35,7 +35,7 @@ #include <mach/dma.h> #include "s3c-i2s-v2.h" -#include "s3c24xx-pcm.h" +#include "s3c-audio.h" #undef S3C_IIS_V2_SUPPORTED diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 23718ea..553c480 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -37,7 +37,7 @@ #include <mach/regs-gpio.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 678b176..000dc79 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -35,7 +35,7 @@ #include <asm/dma.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c24xx-ac97.h" struct s3c24xx_ac97_info { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index afb4bc9..ed83747 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 "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c24xx-i2s.h" static struct s3c2410_dma_client s3c24xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c deleted file mode 100644 index cb49400..0000000 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ /dev/null @@ -1,480 +0,0 @@ -/* - * s3c24xx-pcm.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 "s3c24xx-pcm.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; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; - unsigned long totbytes = params_buffer_bytes(params); - 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->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform s3c24xx_soc_platform = { - .name = "s3c24xx-audio", - .pcm_ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); - -static int __init s3c24xx_soc_platform_init(void) -{ - return snd_soc_register_platform(&s3c24xx_soc_platform); -} -module_init(s3c24xx_soc_platform_init); - -static void __exit s3c24xx_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&s3c24xx_soc_platform); -} -module_exit(s3c24xx_soc_platform_exit); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h deleted file mode 100644 index 8cbc071..0000000 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * s3c24xx-pcm.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_soc_platform s3c24xx_soc_platform; -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 1966e0d..ef75416 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 "s3c24xx-pcm.h" +#include "s3c-audio.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 8346bd9..61166a5 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 "s3c24xx-pcm.h" +#include "s3c-audio.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 25797e0..af0bf57 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 "s3c24xx-pcm.h" +#include "s3c-audio.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 c215d32..e8d6cc2 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 "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c24xx-i2s.h" #include "../codecs/uda134x.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 719d63c..64c034b 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -35,7 +35,7 @@ #include <mach/map.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c64xx-i2s.h" static struct s3c2410_dma_client s3c64xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index a2a4f53..106c72d 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -20,7 +20,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c24xx-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 cb8a916..7956a9c 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 "s3c24xx-pcm.h" +#include "s3c-audio.h" #include "s3c64xx-i2s.h" #define S3C64XX_I2S_V4 2 -- 1.6.2.5

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[alsa-devel] [PATCH 4/7] S3C64XX PCM: Added PCLK clock source for the PCM controllers
by Jassi Brar 16 Nov '09

16 Nov '09

Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c64xx/clock.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-s3c64xx/clock.c b/arch/arm/plat-s3c64xx/clock.c index 7a36e89..3ab564e 100644 --- a/arch/arm/plat-s3c64xx/clock.c +++ b/arch/arm/plat-s3c64xx/clock.c @@ -129,6 +129,18 @@ static struct clk init_clocks_disable[] = { .enable = s3c64xx_pclk_ctrl, .ctrlbit = S3C_CLKCON_PCLK_IIS1, }, { + .name = "pcm", + .id = 0, + .parent = &clk_p, + .enable = s3c64xx_pclk_ctrl, + .ctrlbit = S3C_CLKCON_PCLK_PCM0, + }, { + .name = "pcm", + .id = 1, + .parent = &clk_p, + .enable = s3c64xx_pclk_ctrl, + .ctrlbit = S3C_CLKCON_PCLK_PCM1, + }, { .name = "spi", .id = 0, .parent = &clk_p, -- 1.6.2.5

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[alsa-devel] [PATCH 3/7] S3C64XX PCM: Define PCM Controller base registers
by Jassi Brar 16 Nov '09

16 Nov '09

Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/mach-s3c6400/include/mach/map.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c6400/include/mach/map.h b/arch/arm/mach-s3c6400/include/mach/map.h index fc8b223..866be31 100644 --- a/arch/arm/mach-s3c6400/include/mach/map.h +++ b/arch/arm/mach-s3c6400/include/mach/map.h @@ -48,6 +48,8 @@ #define S3C64XX_PA_IIS1 (0x7F003000) #define S3C64XX_PA_TIMER (0x7F006000) #define S3C64XX_PA_IIC0 (0x7F004000) +#define S3C64XX_PA_PCM0 (0x7F009000) +#define S3C64XX_PA_PCM1 (0x7F00A000) #define S3C64XX_PA_IISV4 (0x7F00D000) #define S3C64XX_PA_IIC1 (0x7F00F000) -- 1.6.2.5

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[alsa-devel] PC Beep or PC Speaker or just Beep? and HDA Beep code...
by Jaroslav Kysela 16 Nov '09

16 Nov '09

Hi, I'm changing the behaviour of Beep / PC Beep controls for HDA drivers (comments?): http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=d605dbeed1b10… I would like to introduce uniform naming for these controls to not confuse users. HDA driver: Beep, PC Beep AC97: PC Speaker, Beep, PC Beep etc.. I think that the original purpose of PC Speaker is a beep generator. Also, many SoC platforms use AC97 codecs, so removing abbreviation PC might make sense. It would be probably best to change all control names in drivers to 'Beep'. Comments? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH 0/2] fix softfloat handling
by Marc Kleine-Budde 16 Nov '09

16 Nov '09

Hello, this series of two patches fixes the softfloat handling within the alsa-lib. Please review and consider to apply. cheers, Marc

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[alsa-devel] [patch] make some bitfields unsigned
by Dan Carpenter 16 Nov '09

16 Nov '09

This is a clean up and doesn't change the behavior. Bit fields should always be unsigned. Otherwise pm_suspend_enabled will be -1 when you want it to be 1. The other bad thing is that the sparse checker will complain 36 times if they aren't unsigned. The other bitfields in that struct are unsigned already. regards, dan carpenter Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- orig/sound/pci/ice1712/ice1712.h 2009-11-14 12:09:26.000000000 +0200 +++ devel/sound/pci/ice1712/ice1712.h 2009-11-14 12:10:01.000000000 +0200 @@ -382,8 +382,8 @@ #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); - int pm_suspend_enabled:1; - int pm_saved_is_spdif_master:1; + unsigned int pm_suspend_enabled:1; + unsigned int pm_saved_is_spdif_master:1; unsigned int pm_saved_spdif_ctrl; unsigned char pm_saved_spdif_cfg; unsigned int pm_saved_route;

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[alsa-devel] [PATCH - Full support for Toshiba NB200 1/1] Support for Toshiba NB200
by cjg＠cruxppc.org 16 Nov '09

16 Nov '09

From: Giuseppe Coviello <cjg(a)cruxppc.org> Signed-off-by: Giuseppe Coviello <cjg(a)cruxppc.org> --- sound/pci/hda/patch_realtek.c | 86 ++++++++++++++++++++++++++++++++++++++++- 1 files changed, 85 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0a98e7..69484b8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -191,6 +191,7 @@ enum { ALC272_DELL, ALC272_DELL_ZM1, ALC272_SAMSUNG_NC10, + ALC272_TOSHIBA_NB200, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -16355,6 +16356,40 @@ static struct hda_verb alc272_dell_init_verbs[] = { {} }; +static struct snd_kcontrol_new alc272_toshiba_mixer[] = { + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc272_toshiba_init_verbs[] = { + /* Front Pin: output 0 (0x0c) */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* 0x14 */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* 0x14 */ + /* Front Mic pin: input vref at 80% */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* 0x19 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* 0x19 */ + {0x17, AC_VERB_SET_EAPD_BTLENABLE, 2}, /* 0x14 */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* originally 0x12 */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + + static struct snd_kcontrol_new alc662_auto_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), @@ -16579,6 +16614,40 @@ static void alc663_m51va_inithook(struct hda_codec *codec) alc_mic_automute(codec); } +static void alc272_toshiba_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + AMP_OUT_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 1, + AMP_OUT_MUTE, bits); +} + +static void alc272_toshiba_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc272_toshiba_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +static void alc272_toshiba_inithook(struct hda_codec *codec) +{ + alc272_toshiba_speaker_automute(codec); + alc_mic_automute(codec); +} + /* ***************** Mode1 ******************************/ #define alc663_mode1_unsol_event alc663_m51va_unsol_event #define alc663_mode1_setup alc663_m51va_setup @@ -16830,6 +16899,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC663_ASUS_MODE6] = "asus-mode6", [ALC272_DELL] = "dell", [ALC272_DELL_ZM1] = "dell-zm1", + [ALC272_TOSHIBA_NB200] = "toshiba-nb200", [ALC272_SAMSUNG_NC10] = "samsung-nc10", [ALC662_AUTO] = "auto", }; @@ -16838,6 +16908,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL), SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC272_TOSHIBA_NB200), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), @@ -16888,7 +16959,6 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), @@ -17175,6 +17245,20 @@ static struct alc_config_preset alc662_presets[] = { .setup = alc663_mode4_setup, .init_hook = alc663_mode4_inithook, }, + [ALC272_TOSHIBA_NB200] = { + .mixers = { alc663_m51va_mixer, alc272_toshiba_mixer }, + .init_verbs = { alc662_init_verbs, alc272_toshiba_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .adc_nids = alc662_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), + .capsrc_nids = alc662_capsrc_nids, + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_capture_source, + .unsol_event = alc272_toshiba_unsol_event, + .init_hook = alc272_toshiba_inithook, + }, }; -- 1.6.4.2

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