android_kernel_xiaomi_sm8350/sound/soc/au1x/dbdma2.c
Manuel Lauss 4a161d235b ALSA: ASoC: Au12x0/Au1550 PSC Audio support
Audio for Au12x0/Au1550 PSCs in AC97 and I2S mode, for ASoC v1 framework.

- DBDMA, AC97 and I2S drivers
- sample AC97 machine code (Db1200)

Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-10 09:33:07 +02:00

422 lines
11 KiB
C

/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <mano@roarinelk.homelinux.net>
*
* 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.
*
* NOTE: all of these drivers can only work with a SINGLE instance
* of a PSC. Multiple independent audio devices are impossible
* with ASoC v1.
*/
#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) */
unsigned long dma_area; /* address of queued DMA area */
unsigned long 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;
};
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
/*
* 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,
.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_flags(cd->ddma_chan,
(void *)phys_to_virt(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_flags(cd->ddma_chan,
(void *)phys_to_virt(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 == PCM_RX)
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 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_TYPE(substream);
pcd = au1xpsc_audio_pcmdma[stype];
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 = (unsigned long)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 =
au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)];
au1xxx_dbdma_reset(pcd->ddma_chan);
if (SUBSTREAM_TYPE(substream) == PCM_RX) {
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 = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(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,
au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos);
}
static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
{
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(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]);
return 0;
}
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_card *card,
struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
return 0;
}
static int au1xpsc_pcm_probe(struct platform_device *pdev)
{
struct resource *r;
int ret;
if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX])
return -EBUSY;
/* TX DMA */
au1xpsc_audio_pcmdma[PCM_TX]
= kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
if (!au1xpsc_audio_pcmdma[PCM_TX])
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!r) {
ret = -ENODEV;
goto out1;
}
(au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start;
/* RX DMA */
au1xpsc_audio_pcmdma[PCM_RX]
= kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
if (!au1xpsc_audio_pcmdma[PCM_RX])
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!r) {
ret = -ENODEV;
goto out2;
}
(au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start;
return 0;
out2:
kfree(au1xpsc_audio_pcmdma[PCM_RX]);
au1xpsc_audio_pcmdma[PCM_RX] = NULL;
out1:
kfree(au1xpsc_audio_pcmdma[PCM_TX]);
au1xpsc_audio_pcmdma[PCM_TX] = NULL;
return ret;
}
static int au1xpsc_pcm_remove(struct platform_device *pdev)
{
int i;
for (i = 0; i < 2; i++) {
if (au1xpsc_audio_pcmdma[i]) {
au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]);
kfree(au1xpsc_audio_pcmdma[i]);
au1xpsc_audio_pcmdma[i] = NULL;
}
}
return 0;
}
/* au1xpsc audio platform */
struct snd_soc_platform au1xpsc_soc_platform = {
.name = "au1xpsc-pcm-dbdma",
.probe = au1xpsc_pcm_probe,
.remove = au1xpsc_pcm_remove,
.pcm_ops = &au1xpsc_pcm_ops,
.pcm_new = au1xpsc_pcm_new,
.pcm_free = au1xpsc_pcm_free_dma_buffers,
};
EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
static int __init au1xpsc_audio_dbdma_init(void)
{
au1xpsc_audio_pcmdma[PCM_TX] = NULL;
au1xpsc_audio_pcmdma[PCM_RX] = NULL;
return 0;
}
static void __exit au1xpsc_audio_dbdma_exit(void)
{
}
module_init(au1xpsc_audio_dbdma_init);
module_exit(au1xpsc_audio_dbdma_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");