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linux-zen-server/sound/soc/qcom/qdsp6/q6asm-dai.c

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Initial commit
2023-08-30 17:53:23 +02:00
// SPDX-License-Identifier: GPL-2.0
// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
// Copyright (c) 2018, Linaro Limited
#include <linux/init.h>
#include <linux/err.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
#include <linux/spinlock.h>
#include <sound/compress_driver.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/of_device.h>
#include <sound/pcm_params.h>
#include "q6asm.h"
#include "q6routing.h"
#include "q6dsp-errno.h"
#define DRV_NAME "q6asm-fe-dai"
#define PLAYBACK_MIN_NUM_PERIODS 2
#define PLAYBACK_MAX_NUM_PERIODS 8
#define PLAYBACK_MAX_PERIOD_SIZE 65536
#define PLAYBACK_MIN_PERIOD_SIZE 128
#define CAPTURE_MIN_NUM_PERIODS 2
#define CAPTURE_MAX_NUM_PERIODS 8
#define CAPTURE_MAX_PERIOD_SIZE 4096
#define CAPTURE_MIN_PERIOD_SIZE 320
#define SID_MASK_DEFAULT 0xF
/* Default values used if user space does not set */
#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1)
#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
enum stream_state {
Q6ASM_STREAM_IDLE = 0,
Q6ASM_STREAM_STOPPED,
Q6ASM_STREAM_RUNNING,
};
struct q6asm_dai_rtd {
struct snd_pcm_substream *substream;
struct snd_compr_stream *cstream;
struct snd_codec codec;
struct snd_dma_buffer dma_buffer;
spinlock_t lock;
phys_addr_t phys;
unsigned int pcm_size;
unsigned int pcm_count;
unsigned int pcm_irq_pos; /* IRQ position */
unsigned int periods;
unsigned int bytes_sent;
unsigned int bytes_received;
unsigned int copied_total;
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
uint32_t next_track_stream_id;
bool next_track;
uint32_t stream_id;
uint16_t session_id;
enum stream_state state;
uint32_t initial_samples_drop;
uint32_t trailing_samples_drop;
bool notify_on_drain;
};
struct q6asm_dai_data {
struct snd_soc_dai_driver *dais;
int num_dais;
long long int sid;
};
static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE),
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 4,
.buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS *
CAPTURE_MAX_PERIOD_SIZE,
.period_bytes_min = CAPTURE_MIN_PERIOD_SIZE,
.period_bytes_max = CAPTURE_MAX_PERIOD_SIZE,
.periods_min = CAPTURE_MIN_NUM_PERIODS,
.periods_max = CAPTURE_MAX_NUM_PERIODS,
.fifo_size = 0,
};
static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE),
.rates = SNDRV_PCM_RATE_8000_192000,
.rate_min = 8000,
.rate_max = 192000,
.channels_min = 1,
.channels_max = 8,
.buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
PLAYBACK_MAX_PERIOD_SIZE),
.period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
.period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
.periods_min = PLAYBACK_MIN_NUM_PERIODS,
.periods_max = PLAYBACK_MAX_NUM_PERIODS,
.fifo_size = 0,
};
#define Q6ASM_FEDAI_DRIVER(num) { \
.playback = { \
.stream_name = "MultiMedia"#num" Playback", \
.rates = (SNDRV_PCM_RATE_8000_192000| \
SNDRV_PCM_RATE_KNOT), \
.formats = (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE), \
.channels_min = 1, \
.channels_max = 8, \
.rate_min = 8000, \
.rate_max = 192000, \
}, \
.capture = { \
.stream_name = "MultiMedia"#num" Capture", \
.rates = (SNDRV_PCM_RATE_8000_48000| \
SNDRV_PCM_RATE_KNOT), \
.formats = (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE), \
.channels_min = 1, \
.channels_max = 4, \
.rate_min = 8000, \
.rate_max = 48000, \
}, \
.name = "MultiMedia"#num, \
.id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
}
/* Conventional and unconventional sample rate supported */
static unsigned int supported_sample_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
88200, 96000, 176400, 192000
};
static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
.count = ARRAY_SIZE(supported_sample_rates),
.list = supported_sample_rates,
.mask = 0,
};
static const struct snd_compr_codec_caps q6asm_compr_caps = {
.num_descriptors = 1,
.descriptor[0].max_ch = 2,
.descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 88200,
96000, 176400, 192000 },
.descriptor[0].num_sample_rates = 13,
.descriptor[0].bit_rate[0] = 320,
.descriptor[0].bit_rate[1] = 128,
.descriptor[0].num_bitrates = 2,
.descriptor[0].profiles = 0,
.descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
.descriptor[0].formats = 0,
};
static void event_handler(uint32_t opcode, uint32_t token,
void *payload, void *priv)
{
struct q6asm_dai_rtd *prtd = priv;
struct snd_pcm_substream *substream = prtd->substream;
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, 0);
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
prtd->state = Q6ASM_STREAM_STOPPED;
break;
case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, 0);
break;
}
case ASM_CLIENT_EVENT_DATA_READ_DONE:
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
q6asm_read(prtd->audio_client, prtd->stream_id);
break;
default:
break;
}
}
static int q6asm_dai_prepare(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
struct q6asm_dai_rtd *prtd = runtime->private_data;
struct q6asm_dai_data *pdata;
struct device *dev = component->dev;
int ret, i;
pdata = snd_soc_component_get_drvdata(component);
if (!pdata)
return -EINVAL;
if (!prtd || !prtd->audio_client) {
dev_err(dev, "%s: private data null or audio client freed\n",
__func__);
return -EINVAL;
}
prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
prtd->pcm_irq_pos = 0;
/* rate and channels are sent to audio driver */
if (prtd->state) {
/* clear the previous setup if any */
q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6routing_stream_close(soc_prtd->dai_link->id,
substream->stream);
}
ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
prtd->phys,
(prtd->pcm_size / prtd->periods),
prtd->periods);
if (ret < 0) {
dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",
ret);
return -ENOMEM;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
FORMAT_LINEAR_PCM,
0, prtd->bits_per_sample, false);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
FORMAT_LINEAR_PCM,
prtd->bits_per_sample);
}
if (ret < 0) {
dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
goto open_err;
}
prtd->session_id = q6asm_get_session_id(prtd->audio_client);
ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
prtd->session_id, substream->stream);
if (ret) {
dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
goto routing_err;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_media_format_block_multi_ch_pcm(
prtd->audio_client, prtd->stream_id,
runtime->rate, runtime->channels, NULL,
prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
prtd->stream_id,
runtime->rate,
runtime->channels,
prtd->bits_per_sample);
/* Queue the buffers */
for (i = 0; i < runtime->periods; i++)
q6asm_read(prtd->audio_client, prtd->stream_id);
}
if (ret < 0)
dev_info(dev, "%s: CMD Format block failed\n", __func__);
else
prtd->state = Q6ASM_STREAM_RUNNING;
return ret;
routing_err:
q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
open_err:
q6asm_unmap_memory_regions(substream->stream, prtd->audio_client);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
return ret;
}
static int q6asm_dai_trigger(struct snd_soc_component *component,
struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
CMD_PAUSE);
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int q6asm_dai_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
struct q6asm_dai_rtd *prtd;
struct q6asm_dai_data *pdata;
struct device *dev = component->dev;
int ret = 0;
int stream_id;
stream_id = cpu_dai->driver->id;
pdata = snd_soc_component_get_drvdata(component);
if (!pdata) {
dev_err(dev, "Drv data not found ..\n");
return -EINVAL;
}
prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
if (prtd == NULL)
return -ENOMEM;
prtd->substream = substream;
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)event_handler, prtd, stream_id,
LEGACY_PCM_MODE);
if (IS_ERR(prtd->audio_client)) {
dev_info(dev, "%s: Could not allocate memory\n", __func__);
ret = PTR_ERR(prtd->audio_client);
kfree(prtd);
return ret;
}
/* DSP expects stream id from 1 */
prtd->stream_id = 1;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = q6asm_dai_hardware_playback;
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
runtime->hw = q6asm_dai_hardware_capture;
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_sample_rates);
if (ret < 0)
dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
/* Ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
if (ret < 0) {
dev_err(dev, "constraint for buffer bytes min max ret = %d\n",
ret);
}
}
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
if (ret < 0) {
dev_err(dev, "constraint for period bytes step ret = %d\n",
ret);
}
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
if (ret < 0) {
dev_err(dev, "constraint for buffer bytes step ret = %d\n",
ret);
}
runtime->private_data = prtd;
snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
if (pdata->sid < 0)
prtd->phys = substream->dma_buffer.addr;
else
prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
return 0;
}
static int q6asm_dai_close(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
struct q6asm_dai_rtd *prtd = runtime->private_data;
if (prtd->audio_client) {
if (prtd->state)
q6asm_cmd(prtd->audio_client, prtd->stream_id,
CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
}
q6routing_stream_close(soc_prtd->dai_link->id,
substream->stream);
kfree(prtd);
return 0;
}
static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
if (prtd->pcm_irq_pos >= prtd->pcm_size)
prtd->pcm_irq_pos = 0;
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
}
static int q6asm_dai_hw_params(struct snd_soc_component *component,
struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
prtd->pcm_size = params_buffer_bytes(params);
prtd->periods = params_periods(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
prtd->bits_per_sample = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
prtd->bits_per_sample = 24;
break;
}
return 0;
}
static void compress_event_handler(uint32_t opcode, uint32_t token,
void *payload, void *priv)
{
struct q6asm_dai_rtd *prtd = priv;
struct snd_compr_stream *substream = prtd->cstream;
unsigned long flags;
u32 wflags = 0;
uint64_t avail;
uint32_t bytes_written, bytes_to_write;
bool is_last_buffer = false;
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
spin_lock_irqsave(&prtd->lock, flags);
if (!prtd->bytes_sent) {
q6asm_stream_remove_initial_silence(prtd->audio_client,
prtd->stream_id,
prtd->initial_samples_drop);
q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, 0);
prtd->bytes_sent += prtd->pcm_count;
}
spin_unlock_irqrestore(&prtd->lock, flags);
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
spin_lock_irqsave(&prtd->lock, flags);
if (prtd->notify_on_drain) {
if (substream->partial_drain) {
/*
* Close old stream and make it stale, switch
* the active stream now!
*/
q6asm_cmd_nowait(prtd->audio_client,
prtd->stream_id,
CMD_CLOSE);
/*
* vaild stream ids start from 1, So we are
* toggling this between 1 and 2.
*/
prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
}
snd_compr_drain_notify(prtd->cstream);
prtd->notify_on_drain = false;
} else {
prtd->state = Q6ASM_STREAM_STOPPED;
}
spin_unlock_irqrestore(&prtd->lock, flags);
break;
case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
spin_lock_irqsave(&prtd->lock, flags);
bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
prtd->copied_total += bytes_written;
snd_compr_fragment_elapsed(substream);
if (prtd->state != Q6ASM_STREAM_RUNNING) {
spin_unlock_irqrestore(&prtd->lock, flags);
break;
}
avail = prtd->bytes_received - prtd->bytes_sent;
if (avail > prtd->pcm_count) {
bytes_to_write = prtd->pcm_count;
} else {
if (substream->partial_drain || prtd->notify_on_drain)
is_last_buffer = true;
bytes_to_write = avail;
}
if (bytes_to_write) {
if (substream->partial_drain && is_last_buffer) {
wflags |= ASM_LAST_BUFFER_FLAG;
q6asm_stream_remove_trailing_silence(prtd->audio_client,
prtd->stream_id,
prtd->trailing_samples_drop);
}
q6asm_write_async(prtd->audio_client, prtd->stream_id,
bytes_to_write, 0, 0, wflags);
prtd->bytes_sent += bytes_to_write;
}
if (prtd->notify_on_drain && is_last_buffer)
q6asm_cmd_nowait(prtd->audio_client,
prtd->stream_id, CMD_EOS);
spin_unlock_irqrestore(&prtd->lock, flags);
break;
default:
break;
}
}
static int q6asm_dai_compr_open(struct snd_soc_component *component,
struct snd_compr_stream *stream)
{
struct snd_soc_pcm_runtime *rtd = stream->private_data;
struct snd_compr_runtime *runtime = stream->runtime;
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct q6asm_dai_data *pdata;
struct device *dev = component->dev;
struct q6asm_dai_rtd *prtd;
int stream_id, size, ret;
stream_id = cpu_dai->driver->id;
pdata = snd_soc_component_get_drvdata(component);
if (!pdata) {
dev_err(dev, "Drv data not found ..\n");
return -EINVAL;
}
prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
if (!prtd)
return -ENOMEM;
/* DSP expects stream id from 1 */
prtd->stream_id = 1;
prtd->cstream = stream;
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)compress_event_handler,
prtd, stream_id, LEGACY_PCM_MODE);
if (IS_ERR(prtd->audio_client)) {
dev_err(dev, "Could not allocate memory\n");
ret = PTR_ERR(prtd->audio_client);
goto free_prtd;
}
size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
&prtd->dma_buffer);
if (ret) {
dev_err(dev, "Cannot allocate buffer(s)\n");
goto free_client;
}
if (pdata->sid < 0)
prtd->phys = prtd->dma_buffer.addr;
else
prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
spin_lock_init(&prtd->lock);
runtime->private_data = prtd;
return 0;
free_client:
q6asm_audio_client_free(prtd->audio_client);
free_prtd:
kfree(prtd);
return ret;
}
static int q6asm_dai_compr_free(struct snd_soc_component *component,
struct snd_compr_stream *stream)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = stream->private_data;
if (prtd->audio_client) {
if (prtd->state) {
q6asm_cmd(prtd->audio_client, prtd->stream_id,
CMD_CLOSE);
if (prtd->next_track_stream_id) {
q6asm_cmd(prtd->audio_client,
prtd->next_track_stream_id,
CMD_CLOSE);
}
}
snd_dma_free_pages(&prtd->dma_buffer);
q6asm_unmap_memory_regions(stream->direction,
prtd->audio_client);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
}
q6routing_stream_close(rtd->dai_link->id, stream->direction);
kfree(prtd);
return 0;
}
static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_codec *codec,
int stream_id)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
struct q6asm_flac_cfg flac_cfg;
struct q6asm_wma_cfg wma_cfg;
struct q6asm_alac_cfg alac_cfg;
struct q6asm_ape_cfg ape_cfg;
unsigned int wma_v9 = 0;
struct device *dev = component->dev;
int ret;
union snd_codec_options *codec_options;
struct snd_dec_flac *flac;
struct snd_dec_wma *wma;
struct snd_dec_alac *alac;
struct snd_dec_ape *ape;
codec_options = &(prtd->codec.options);
memcpy(&prtd->codec, codec, sizeof(*codec));
switch (codec->id) {
case SND_AUDIOCODEC_FLAC:
memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
flac = &codec_options->flac_d;
flac_cfg.ch_cfg = codec->ch_in;
flac_cfg.sample_rate = codec->sample_rate;
flac_cfg.stream_info_present = 1;
flac_cfg.sample_size = flac->sample_size;
flac_cfg.min_blk_size = flac->min_blk_size;
flac_cfg.max_blk_size = flac->max_blk_size;
flac_cfg.max_frame_size = flac->max_frame_size;
flac_cfg.min_frame_size = flac->min_frame_size;
ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
stream_id,
&flac_cfg);
if (ret < 0) {
dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
return -EIO;
}
break;
case SND_AUDIOCODEC_WMA:
wma = &codec_options->wma_d;
memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
wma_cfg.sample_rate = codec->sample_rate;
wma_cfg.num_channels = codec->ch_in;
wma_cfg.bytes_per_sec = codec->bit_rate / 8;
wma_cfg.block_align = codec->align;
wma_cfg.bits_per_sample = prtd->bits_per_sample;
wma_cfg.enc_options = wma->encoder_option;
wma_cfg.adv_enc_options = wma->adv_encoder_option;
wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
if (wma_cfg.num_channels == 1)
wma_cfg.channel_mask = 4; /* Mono Center */
else if (wma_cfg.num_channels == 2)
wma_cfg.channel_mask = 3; /* Stereo FL/FR */
else
return -EINVAL;
/* check the codec profile */
switch (codec->profile) {
case SND_AUDIOPROFILE_WMA9:
wma_cfg.fmtag = 0x161;
wma_v9 = 1;
break;
case SND_AUDIOPROFILE_WMA10:
wma_cfg.fmtag = 0x166;
break;
case SND_AUDIOPROFILE_WMA9_PRO:
wma_cfg.fmtag = 0x162;
break;
case SND_AUDIOPROFILE_WMA9_LOSSLESS:
wma_cfg.fmtag = 0x163;
break;
case SND_AUDIOPROFILE_WMA10_LOSSLESS:
wma_cfg.fmtag = 0x167;
break;
default:
dev_err(dev, "Unknown WMA profile:%x\n",
codec->profile);
return -EIO;
}
if (wma_v9)
ret = q6asm_stream_media_format_block_wma_v9(
prtd->audio_client, stream_id,
&wma_cfg);
else
ret = q6asm_stream_media_format_block_wma_v10(
prtd->audio_client, stream_id,
&wma_cfg);
if (ret < 0) {
dev_err(dev, "WMA9 CMD failed:%d\n", ret);
return -EIO;
}
break;
case SND_AUDIOCODEC_ALAC:
memset(&alac_cfg, 0x0, sizeof(alac_cfg));
alac = &codec_options->alac_d;
alac_cfg.sample_rate = codec->sample_rate;
alac_cfg.avg_bit_rate = codec->bit_rate;
alac_cfg.bit_depth = prtd->bits_per_sample;
alac_cfg.num_channels = codec->ch_in;
alac_cfg.frame_length = alac->frame_length;
alac_cfg.pb = alac->pb;
alac_cfg.mb = alac->mb;
alac_cfg.kb = alac->kb;
alac_cfg.max_run = alac->max_run;
alac_cfg.compatible_version = alac->compatible_version;
alac_cfg.max_frame_bytes = alac->max_frame_bytes;
switch (codec->ch_in) {
case 1:
alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
break;
case 2:
alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
break;
}
ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
stream_id,
&alac_cfg);
if (ret < 0) {
dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
return -EIO;
}
break;
case SND_AUDIOCODEC_APE:
memset(&ape_cfg, 0x0, sizeof(ape_cfg));
ape = &codec_options->ape_d;
ape_cfg.sample_rate = codec->sample_rate;
ape_cfg.num_channels = codec->ch_in;
ape_cfg.bits_per_sample = prtd->bits_per_sample;
ape_cfg.compatible_version = ape->compatible_version;
ape_cfg.compression_level = ape->compression_level;
ape_cfg.format_flags = ape->format_flags;
ape_cfg.blocks_per_frame = ape->blocks_per_frame;
ape_cfg.final_frame_blocks = ape->final_frame_blocks;
ape_cfg.total_frames = ape->total_frames;
ape_cfg.seek_table_present = ape->seek_table_present;
ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
stream_id,
&ape_cfg);
if (ret < 0) {
dev_err(dev, "APE CMD Format block failed:%d\n", ret);
return -EIO;
}
break;
default:
break;
}
return 0;
}
static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_params *params)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = stream->private_data;
int dir = stream->direction;
struct q6asm_dai_data *pdata;
struct device *dev = component->dev;
int ret;
pdata = snd_soc_component_get_drvdata(component);
if (!pdata)
return -EINVAL;
if (!prtd || !prtd->audio_client) {
dev_err(dev, "private data null or audio client freed\n");
return -EINVAL;
}
prtd->periods = runtime->fragments;
prtd->pcm_count = runtime->fragment_size;
prtd->pcm_size = runtime->fragments * runtime->fragment_size;
prtd->bits_per_sample = 16;
if (dir == SND_COMPRESS_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
params->codec.profile, prtd->bits_per_sample,
true);
if (ret < 0) {
dev_err(dev, "q6asm_open_write failed\n");
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
return ret;
}
}
prtd->session_id = q6asm_get_session_id(prtd->audio_client);
ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
prtd->session_id, dir);
if (ret) {
dev_err(dev, "Stream reg failed ret:%d\n", ret);
return ret;
}
ret = __q6asm_dai_compr_set_codec_params(component, stream,
&params->codec,
prtd->stream_id);
if (ret) {
dev_err(dev, "codec param setup failed ret:%d\n", ret);
return ret;
}
ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
(prtd->pcm_size / prtd->periods),
prtd->periods);
if (ret < 0) {
dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
return -ENOMEM;
}
prtd->state = Q6ASM_STREAM_RUNNING;
return 0;
}
static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_metadata *metadata)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
int ret = 0;
switch (metadata->key) {
case SNDRV_COMPRESS_ENCODER_PADDING:
prtd->trailing_samples_drop = metadata->value[0];
break;
case SNDRV_COMPRESS_ENCODER_DELAY:
prtd->initial_samples_drop = metadata->value[0];
if (prtd->next_track_stream_id) {
ret = q6asm_open_write(prtd->audio_client,
prtd->next_track_stream_id,
prtd->codec.id,
prtd->codec.profile,
prtd->bits_per_sample,
true);
if (ret < 0) {
dev_err(component->dev, "q6asm_open_write failed\n");
return ret;
}
ret = __q6asm_dai_compr_set_codec_params(component, stream,
&prtd->codec,
prtd->next_track_stream_id);
if (ret < 0) {
dev_err(component->dev, "q6asm_open_write failed\n");
return ret;
}
ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
prtd->next_track_stream_id,
prtd->initial_samples_drop);
prtd->next_track_stream_id = 0;
}
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
struct snd_compr_stream *stream, int cmd)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
CMD_PAUSE);
break;
case SND_COMPR_TRIGGER_NEXT_TRACK:
prtd->next_track = true;
prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
break;
case SND_COMPR_TRIGGER_DRAIN:
case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
prtd->notify_on_drain = true;
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_tstamp *tstamp)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
spin_lock_irqsave(&prtd->lock, flags);
tstamp->copied_total = prtd->copied_total;
tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
spin_unlock_irqrestore(&prtd->lock, flags);
return 0;
}
static int q6asm_compr_copy(struct snd_soc_component *component,
struct snd_compr_stream *stream, char __user *buf,
size_t count)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
u32 wflags = 0;
int avail, bytes_in_flight = 0;
void *dstn;
size_t copy;
u32 app_pointer;
u32 bytes_received;
bytes_received = prtd->bytes_received;
/**
* Make sure that next track data pointer is aligned at 32 bit boundary
* This is a Mandatory requirement from DSP data buffers alignment
*/
if (prtd->next_track)
bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
app_pointer = bytes_received/prtd->pcm_size;
app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
dstn = prtd->dma_buffer.area + app_pointer;
if (count < prtd->pcm_size - app_pointer) {
if (copy_from_user(dstn, buf, count))
return -EFAULT;
} else {
copy = prtd->pcm_size - app_pointer;
if (copy_from_user(dstn, buf, copy))
return -EFAULT;
if (copy_from_user(prtd->dma_buffer.area, buf + copy,
count - copy))
return -EFAULT;
}
spin_lock_irqsave(&prtd->lock, flags);
bytes_in_flight = prtd->bytes_received - prtd->copied_total;
if (prtd->next_track) {
prtd->next_track = false;
prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
}
prtd->bytes_received = bytes_received + count;
/* Kick off the data to dsp if its starving!! */
if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
uint32_t bytes_to_write = prtd->pcm_count;
avail = prtd->bytes_received - prtd->bytes_sent;
if (avail < prtd->pcm_count)
bytes_to_write = avail;
q6asm_write_async(prtd->audio_client, prtd->stream_id,
bytes_to_write, 0, 0, wflags);
prtd->bytes_sent += bytes_to_write;
}
spin_unlock_irqrestore(&prtd->lock, flags);
return count;
}
static int q6asm_dai_compr_mmap(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct vm_area_struct *vma)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
struct device *dev = component->dev;
return dma_mmap_coherent(dev, vma,
prtd->dma_buffer.area, prtd->dma_buffer.addr,
prtd->dma_buffer.bytes);
}
static int q6asm_dai_compr_get_caps(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_caps *caps)
{
caps->direction = SND_COMPRESS_PLAYBACK;
caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
caps->num_codecs = 5;
caps->codecs[0] = SND_AUDIOCODEC_MP3;
caps->codecs[1] = SND_AUDIOCODEC_FLAC;
caps->codecs[2] = SND_AUDIOCODEC_WMA;
caps->codecs[3] = SND_AUDIOCODEC_ALAC;
caps->codecs[4] = SND_AUDIOCODEC_APE;
return 0;
}
static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_codec_caps *codec)
{
switch (codec->codec) {
case SND_AUDIOCODEC_MP3:
*codec = q6asm_compr_caps;
break;
default:
break;
}
return 0;
}
static const struct snd_compress_ops q6asm_dai_compress_ops = {
.open = q6asm_dai_compr_open,
.free = q6asm_dai_compr_free,
.set_params = q6asm_dai_compr_set_params,
.set_metadata = q6asm_dai_compr_set_metadata,
.pointer = q6asm_dai_compr_pointer,
.trigger = q6asm_dai_compr_trigger,
.get_caps = q6asm_dai_compr_get_caps,
.get_codec_caps = q6asm_dai_compr_get_codec_caps,
.mmap = q6asm_dai_compr_mmap,
.copy = q6asm_compr_copy,
};
static int q6asm_dai_pcm_new(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm *pcm = rtd->pcm;
size_t size = q6asm_dai_hardware_playback.buffer_bytes_max;
return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
component->dev, size);
}
static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0),
};
static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.name = DRV_NAME,
.open = q6asm_dai_open,
.hw_params = q6asm_dai_hw_params,
.close = q6asm_dai_close,
.prepare = q6asm_dai_prepare,
.trigger = q6asm_dai_trigger,
.pointer = q6asm_dai_pointer,
.pcm_construct = q6asm_dai_pcm_new,
.compress_ops = &q6asm_dai_compress_ops,
.dapm_widgets = q6asm_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets),
.legacy_dai_naming = 1,
};
static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
Q6ASM_FEDAI_DRIVER(1),
Q6ASM_FEDAI_DRIVER(2),
Q6ASM_FEDAI_DRIVER(3),
Q6ASM_FEDAI_DRIVER(4),
Q6ASM_FEDAI_DRIVER(5),
Q6ASM_FEDAI_DRIVER(6),
Q6ASM_FEDAI_DRIVER(7),
Q6ASM_FEDAI_DRIVER(8),
};
static int of_q6asm_parse_dai_data(struct device *dev,
struct q6asm_dai_data *pdata)
{
struct snd_soc_dai_driver *dai_drv;
struct snd_soc_pcm_stream empty_stream;
struct device_node *node;
int ret, id, dir, idx = 0;
pdata->num_dais = of_get_child_count(dev->of_node);
if (!pdata->num_dais) {
dev_err(dev, "No dais found in DT\n");
return -EINVAL;
}
pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
GFP_KERNEL);
if (!pdata->dais)
return -ENOMEM;
memset(&empty_stream, 0, sizeof(empty_stream));
for_each_child_of_node(dev->of_node, node) {
ret = of_property_read_u32(node, "reg", &id);
if (ret || id >= MAX_SESSIONS || id < 0) {
dev_err(dev, "valid dai id not found:%d\n", ret);
continue;
}
dai_drv = &pdata->dais[idx++];
*dai_drv = q6asm_fe_dais_template[id];
ret = of_property_read_u32(node, "direction", &dir);
if (ret)
continue;
if (dir == Q6ASM_DAI_RX)
dai_drv->capture = empty_stream;
else if (dir == Q6ASM_DAI_TX)
dai_drv->playback = empty_stream;
if (of_property_read_bool(node, "is-compress-dai"))
dai_drv->compress_new = snd_soc_new_compress;
}
return 0;
}
static int q6asm_dai_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
struct device_node *node = dev->of_node;
struct of_phandle_args args;
struct q6asm_dai_data *pdata;
int rc;
pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
if (!pdata)
return -ENOMEM;
rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
if (rc < 0)
pdata->sid = -1;
else
pdata->sid = args.args[0] & SID_MASK_DEFAULT;
dev_set_drvdata(dev, pdata);
rc = of_q6asm_parse_dai_data(dev, pdata);
if (rc)
return rc;
return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
pdata->dais, pdata->num_dais);
}
#ifdef CONFIG_OF
static const struct of_device_id q6asm_dai_device_id[] = {
{ .compatible = "qcom,q6asm-dais" },
{},
};
MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
#endif
static struct platform_driver q6asm_dai_platform_driver = {
.driver = {
.name = "q6asm-dai",
.of_match_table = of_match_ptr(q6asm_dai_device_id),
},
.probe = q6asm_dai_probe,
};
module_platform_driver(q6asm_dai_platform_driver);
MODULE_DESCRIPTION("Q6ASM dai driver");
MODULE_LICENSE("GPL v2");