linux/sound/soc/au1x/dbdma2.c

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/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <manuel.lauss@gmail.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.
*
* DMA glue for Au1x-PSC audio.
*
*/
#include <linux/module.h>
#include <linux/init.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/mach-au1x00/au1000.h>
#include <asm/mach-au1x00/au1xxx_dbdma.h>
#include <asm/mach-au1x00/au1xxx_psc.h>
#include "psc.h"
/*#define PCM_DEBUG*/
#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x)
#ifdef PCM_DEBUG
#define DBG MSG
#else
#define DBG(x...) do {} while (0)
#endif
struct au1xpsc_audio_dmadata {
/* DDMA control data */
unsigned int ddma_id; /* DDMA direction ID for this PSC */
u32 ddma_chan; /* DDMA context */
/* PCM context (for irq handlers) */
struct snd_pcm_substream *substream;
unsigned long curr_period; /* current segment DDMA is working on */
unsigned long q_period; /* queue period(s) */
dma_addr_t dma_area; /* address of queued DMA area */
dma_addr_t dma_area_s; /* start address of DMA area */
unsigned long pos; /* current byte position being played */
unsigned long periods; /* number of SG segments in total */
unsigned long period_bytes; /* size in bytes of one SG segment */
/* runtime data */
int msbits;
};
/*
* These settings are somewhat okay, at least on my machine audio plays
* almost skip-free. Especially the 64kB buffer seems to help a LOT.
*/
#define AU1XPSC_PERIOD_MIN_BYTES 1024
#define AU1XPSC_BUFFER_MIN_BYTES 65536
#define AU1XPSC_PCM_FMTS \
(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \
SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \
0)
/* PCM hardware DMA capabilities - platform specific */
static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
.formats = AU1XPSC_PCM_FMTS,
.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
.period_bytes_max = 4096 * 1024 - 1,
.periods_min = 2,
.periods_max = 4096, /* 2 to as-much-as-you-like */
.buffer_bytes_max = 4096 * 1024 - 1,
.fifo_size = 16, /* fifo entries of AC97/I2S PSC */
};
static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
{
au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area,
cd->period_bytes, DDMA_FLAGS_IE);
/* update next-to-queue period */
++cd->q_period;
cd->dma_area += cd->period_bytes;
if (cd->q_period >= cd->periods) {
cd->q_period = 0;
cd->dma_area = cd->dma_area_s;
}
}
static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
{
au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area,
cd->period_bytes, DDMA_FLAGS_IE);
/* update next-to-queue period */
++cd->q_period;
cd->dma_area += cd->period_bytes;
if (cd->q_period >= cd->periods) {
cd->q_period = 0;
cd->dma_area = cd->dma_area_s;
}
}
static void au1x_pcm_dmatx_cb(int irq, void *dev_id)
{
struct au1xpsc_audio_dmadata *cd = dev_id;
cd->pos += cd->period_bytes;
if (++cd->curr_period >= cd->periods) {
cd->pos = 0;
cd->curr_period = 0;
}
snd_pcm_period_elapsed(cd->substream);
au1x_pcm_queue_tx(cd);
}
static void au1x_pcm_dmarx_cb(int irq, void *dev_id)
{
struct au1xpsc_audio_dmadata *cd = dev_id;
cd->pos += cd->period_bytes;
if (++cd->curr_period >= cd->periods) {
cd->pos = 0;
cd->curr_period = 0;
}
snd_pcm_period_elapsed(cd->substream);
au1x_pcm_queue_rx(cd);
}
static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd)
{
if (pcd->ddma_chan) {
au1xxx_dbdma_stop(pcd->ddma_chan);
au1xxx_dbdma_reset(pcd->ddma_chan);
au1xxx_dbdma_chan_free(pcd->ddma_chan);
pcd->ddma_chan = 0;
pcd->msbits = 0;
}
}
/* in case of missing DMA ring or changed TX-source / RX-dest bit widths,
* allocate (or reallocate) a 2-descriptor DMA ring with bit depth according
* to ALSA-supplied sample depth. This is due to limitations in the dbdma api
* (cannot adjust source/dest widths of already allocated descriptor ring).
*/
static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
int stype, int msbits)
{
/* DMA only in 8/16/32 bit widths */
if (msbits == 24)
msbits = 32;
/* check current config: correct bits and descriptors allocated? */
if ((pcd->ddma_chan) && (msbits == pcd->msbits))
goto out; /* all ok! */
au1x_pcm_dbdma_free(pcd);
if (stype == SNDRV_PCM_STREAM_CAPTURE)
pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
DSCR_CMD0_ALWAYS,
au1x_pcm_dmarx_cb, (void *)pcd);
else
pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS,
pcd->ddma_id,
au1x_pcm_dmatx_cb, (void *)pcd);
if (!pcd->ddma_chan)
return -ENOMEM;
au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits);
au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2);
pcd->msbits = msbits;
au1xxx_dbdma_stop(pcd->ddma_chan);
au1xxx_dbdma_reset(pcd->ddma_chan);
out:
return 0;
}
static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss)
{
struct snd_soc_pcm_runtime *rtd = ss->private_data;
struct au1xpsc_audio_dmadata *pcd =
snd_soc_platform_get_drvdata(rtd->platform);
return &pcd[ss->stream];
}
static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct au1xpsc_audio_dmadata *pcd;
int stype, ret;
ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
if (ret < 0)
goto out;
stype = substream->stream;
pcd = to_dmadata(substream);
DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
"runtime->min_align %d\n",
(unsigned long)runtime->dma_area,
(unsigned long)runtime->dma_addr, runtime->dma_bytes,
runtime->min_align);
DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits,
params_periods(params), params_period_bytes(params), stype);
ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits);
if (ret) {
MSG("DDMA channel (re)alloc failed!\n");
goto out;
}
pcd->substream = substream;
pcd->period_bytes = params_period_bytes(params);
pcd->periods = params_periods(params);
pcd->dma_area_s = pcd->dma_area = runtime->dma_addr;
pcd->q_period = 0;
pcd->curr_period = 0;
pcd->pos = 0;
ret = 0;
out:
return ret;
}
static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_lib_free_pages(substream);
return 0;
}
static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
{
struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
au1xxx_dbdma_reset(pcd->ddma_chan);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
au1x_pcm_queue_rx(pcd);
au1x_pcm_queue_rx(pcd);
} else {
au1x_pcm_queue_tx(pcd);
au1x_pcm_queue_tx(pcd);
}
return 0;
}
static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
u32 c = to_dmadata(substream)->ddma_chan;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
au1xxx_dbdma_start(c);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
au1xxx_dbdma_stop(c);
break;
default:
return -EINVAL;
}
return 0;
}
static snd_pcm_uframes_t
au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
{
return bytes_to_frames(substream->runtime, to_dmadata(substream)->pos);
}
static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
{
struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int stype = substream->stream, *dmaids;
dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
if (!dmaids)
return -ENODEV; /* whoa, has ordering changed? */
pcd->ddma_id = dmaids[stype];
snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
return 0;
}
static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
{
au1x_pcm_dbdma_free(to_dmadata(substream));
return 0;
}
static struct snd_pcm_ops au1xpsc_pcm_ops = {
.open = au1xpsc_pcm_open,
.close = au1xpsc_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = au1xpsc_pcm_hw_params,
.hw_free = au1xpsc_pcm_hw_free,
.prepare = au1xpsc_pcm_prepare,
.trigger = au1xpsc_pcm_trigger,
.pointer = au1xpsc_pcm_pointer,
};
static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
{
snd_pcm_lib_preallocate_free_for_all(pcm);
}
static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
struct snd_pcm *pcm = rtd->pcm;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
return 0;
}
/* au1xpsc audio platform */
static struct snd_soc_platform_driver au1xpsc_soc_platform = {
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
.ops = &au1xpsc_pcm_ops,
.pcm_new = au1xpsc_pcm_new,
.pcm_free = au1xpsc_pcm_free_dma_buffers,
};
static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
{
struct au1xpsc_audio_dmadata *dmadata;
dmadata = devm_kzalloc(&pdev->dev,
2 * sizeof(struct au1xpsc_audio_dmadata),
GFP_KERNEL);
if (!dmadata)
return -ENOMEM;
platform_set_drvdata(pdev, dmadata);
return snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform);
}
static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
snd_soc_unregister_platform(&pdev->dev);
return 0;
}
static struct platform_driver au1xpsc_pcm_driver = {
.driver = {
.name = "au1xpsc-pcm",
.owner = THIS_MODULE,
},
.probe = au1xpsc_pcm_drvprobe,
.remove = __devexit_p(au1xpsc_pcm_drvremove),
};
module_platform_driver(au1xpsc_pcm_driver);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
MODULE_AUTHOR("Manuel Lauss");