ASoC: qdsp6: q6asm: Add support to audio stream apis

This patch adds support to open, write and media format commands
in the q6asm module.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Srinivas Kandagatla 2018-05-18 13:56:03 +01:00 коммит произвёл Mark Brown
Родитель a2a5d30218
Коммит 68fd8480bb
Не найден ключ, соответствующий данной подписи
Идентификатор ключа GPG: 24D68B725D5487D0
2 изменённых файлов: 849 добавлений и 0 удалений

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@ -11,6 +11,8 @@
#include <linux/spinlock.h>
#include <linux/kref.h>
#include <linux/of.h>
#include <linux/of_platform.h>
#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
@ -19,10 +21,36 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
#define ASM_STREAM_CMD_CLOSE 0x00010BCD
#define ASM_STREAM_CMD_FLUSH 0x00010BCE
#define ASM_SESSION_CMD_PAUSE 0x00010BD3
#define ASM_DATA_CMD_EOS 0x00010BDB
#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
#define ASM_STREAM_POSTPROC_TOPO_ID_NONE 0x00010C68
#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
#define ASM_LEGACY_STREAM_SESSION 0
/* Bit shift for the stream_perf_mode subfield. */
#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
#define ASM_END_POINT_DEVICE_MATRIX 0
#define ASM_DEFAULT_APP_TYPE 0
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
@ -46,6 +74,89 @@ struct avs_cmd_shared_mem_unmap_regions {
u32 mem_map_handle;
} __packed;
struct asm_data_cmd_media_fmt_update_v2 {
u32 fmt_blk_size;
} __packed;
struct asm_multi_channel_pcm_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 num_channels;
u16 bits_per_sample;
u32 sample_rate;
u16 is_signed;
u16 reserved;
u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
} __packed;
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
} __packed;
struct asm_enc_cfg_blk_param_v2 {
u32 frames_per_buf;
u32 enc_cfg_blk_size;
} __packed;
struct asm_multi_channel_pcm_enc_cfg_v2 {
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
uint16_t num_channels;
uint16_t bits_per_sample;
uint32_t sample_rate;
uint16_t is_signed;
uint16_t reserved;
uint8_t channel_mapping[8];
} __packed;
struct asm_data_cmd_read_v2 {
u32 buf_addr_lsw;
u32 buf_addr_msw;
u32 mem_map_handle;
u32 buf_size;
u32 seq_id;
} __packed;
struct asm_data_cmd_read_v2_done {
u32 status;
u32 buf_addr_lsw;
u32 buf_addr_msw;
};
struct asm_stream_cmd_open_read_v3 {
u32 mode_flags;
u32 src_endpointype;
u32 preprocopo_id;
u32 enc_cfg_id;
u16 bits_per_sample;
u16 reserved;
} __packed;
struct asm_data_cmd_write_v2 {
u32 buf_addr_lsw;
u32 buf_addr_msw;
u32 mem_map_handle;
u32 buf_size;
u32 seq_id;
u32 timestamp_lsw;
u32 timestamp_msw;
u32 flags;
} __packed;
struct asm_stream_cmd_open_write_v3 {
uint32_t mode_flags;
uint16_t sink_endpointype;
uint16_t bits_per_sample;
uint32_t postprocopo_id;
uint32_t dec_fmt_id;
} __packed;
struct asm_session_cmd_run_v2 {
u32 flags;
u32 time_lsw;
u32 time_msw;
} __packed;
struct audio_buffer {
phys_addr_t phys;
uint32_t size; /* size of buffer */
@ -409,6 +520,149 @@ err:
return ac;
}
static int32_t q6asm_stream_callback(struct apr_device *adev,
struct apr_resp_pkt *data,
int session_id)
{
struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
struct aprv2_ibasic_rsp_result_t *result;
struct apr_hdr *hdr = &data->hdr;
struct audio_port_data *port;
struct audio_client *ac;
uint32_t client_event = 0;
int ret = 0;
ac = q6asm_get_audio_client(q6asm, session_id);
if (!ac)/* Audio client might already be freed by now */
return 0;
result = data->payload;
switch (hdr->opcode) {
case APR_BASIC_RSP_RESULT:
switch (result->opcode) {
case ASM_SESSION_CMD_PAUSE:
client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
break;
case ASM_SESSION_CMD_SUSPEND:
client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
break;
case ASM_DATA_CMD_EOS:
client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
break;
case ASM_STREAM_CMD_FLUSH:
client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
break;
case ASM_SESSION_CMD_RUN_V2:
client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
break;
case ASM_STREAM_CMD_CLOSE:
client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
break;
case ASM_STREAM_CMD_FLUSH_READBUFS:
client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
break;
case ASM_STREAM_CMD_OPEN_WRITE_V3:
case ASM_STREAM_CMD_OPEN_READ_V3:
case ASM_STREAM_CMD_OPEN_READWRITE_V2:
case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
if (result->status != 0) {
dev_err(ac->dev,
"cmd = 0x%x returned error = 0x%x\n",
result->opcode, result->status);
ac->result = *result;
wake_up(&ac->cmd_wait);
ret = 0;
goto done;
}
break;
default:
dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
result->opcode);
break;
}
ac->result = *result;
wake_up(&ac->cmd_wait);
if (ac->cb)
ac->cb(client_event, hdr->token,
data->payload, ac->priv);
ret = 0;
goto done;
case ASM_DATA_EVENT_WRITE_DONE_V2:
client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
if (ac->io_mode & ASM_SYNC_IO_MODE) {
phys_addr_t phys;
unsigned long flags;
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
if (!port->buf) {
spin_unlock_irqrestore(&ac->lock, flags);
ret = 0;
goto done;
}
phys = port->buf[hdr->token].phys;
if (lower_32_bits(phys) != result->opcode ||
upper_32_bits(phys) != result->status) {
dev_err(ac->dev, "Expected addr %pa\n",
&port->buf[hdr->token].phys);
spin_unlock_irqrestore(&ac->lock, flags);
ret = -EINVAL;
goto done;
}
spin_unlock_irqrestore(&ac->lock, flags);
}
break;
case ASM_DATA_EVENT_READ_DONE_V2:
client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
if (ac->io_mode & ASM_SYNC_IO_MODE) {
struct asm_data_cmd_read_v2_done *done = data->payload;
unsigned long flags;
phys_addr_t phys;
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
if (!port->buf) {
spin_unlock_irqrestore(&ac->lock, flags);
ret = 0;
goto done;
}
phys = port->buf[hdr->token].phys;
if (upper_32_bits(phys) != done->buf_addr_msw ||
lower_32_bits(phys) != done->buf_addr_lsw) {
dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
&port->buf[hdr->token].phys,
done->buf_addr_lsw,
done->buf_addr_msw);
spin_unlock_irqrestore(&ac->lock, flags);
ret = -EINVAL;
goto done;
}
spin_unlock_irqrestore(&ac->lock, flags);
}
break;
}
if (ac->cb)
ac->cb(client_event, hdr->token, data->payload, ac->priv);
done:
kref_put(&ac->refcount, q6asm_audio_client_release);
return ret;
}
static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_resp_pkt *data)
{
@ -420,6 +674,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct q6asm *a;
uint32_t sid = 0;
uint32_t dir = 0;
int session_id;
session_id = (hdr->dest_port >> 8) & 0xFF;
if (session_id)
return q6asm_stream_callback(adev, data, session_id);
sid = (hdr->token >> 8) & 0x0F;
ac = q6asm_get_audio_client(q6asm, sid);
@ -540,6 +799,547 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
}
EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
{
struct apr_hdr *hdr = &pkt->hdr;
int rc;
mutex_lock(&ac->cmd_lock);
ac->result.opcode = 0;
ac->result.status = 0;
rc = apr_send_pkt(ac->adev, pkt);
if (rc < 0)
goto err;
rc = wait_event_timeout(ac->cmd_wait,
(ac->result.opcode == hdr->opcode), 5 * HZ);
if (!rc) {
dev_err(ac->dev, "CMD timeout\n");
rc = -ETIMEDOUT;
goto err;
}
if (ac->result.status > 0) {
dev_err(ac->dev, "DSP returned error[%x]\n",
ac->result.status);
rc = -EINVAL;
} else {
rc = 0;
}
err:
mutex_unlock(&ac->cmd_lock);
return rc;
}
/**
* q6asm_open_write() - Open audio client for writing
*
* @ac: audio client pointer
* @format: audio sample format
* @bits_per_sample: bits per sample
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_open_write(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_write_v3 *open;
struct apr_pkt *pkt;
void *p;
int rc, pkt_size;
pkt_size = APR_HDR_SIZE + sizeof(*open);
p = kzalloc(pkt_size, GFP_KERNEL);
if (!p)
return -ENOMEM;
pkt = p;
open = p + APR_HDR_SIZE;
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
open->mode_flags = 0x00;
open->mode_flags |= ASM_LEGACY_STREAM_SESSION;
/* source endpoint : matrix */
open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
open->bits_per_sample = bits_per_sample;
open->postprocopo_id = ASM_NULL_POPP_TOPOLOGY;
switch (format) {
case FORMAT_LINEAR_PCM:
open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
goto err;
}
rc = q6asm_ac_send_cmd_sync(ac, pkt);
if (rc < 0)
goto err;
ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
err:
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_open_write);
static int __q6asm_run(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts, bool wait)
{
struct asm_session_cmd_run_v2 *run;
struct apr_pkt *pkt;
int pkt_size, rc;
void *p;
pkt_size = APR_HDR_SIZE + sizeof(*run);
p = kzalloc(pkt_size, GFP_ATOMIC);
if (!p)
return -ENOMEM;
pkt = p;
run = p + APR_HDR_SIZE;
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
run->flags = flags;
run->time_lsw = lsw_ts;
run->time_msw = msw_ts;
if (wait) {
rc = q6asm_ac_send_cmd_sync(ac, pkt);
} else {
rc = apr_send_pkt(ac->adev, pkt);
if (rc == pkt_size)
rc = 0;
}
kfree(pkt);
return rc;
}
/**
* q6asm_run() - start the audio client
*
* @ac: audio client pointer
* @flags: flags associated with write
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_run(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
}
EXPORT_SYMBOL_GPL(q6asm_run);
/**
* q6asm_run_nowait() - start the audio client withou blocking
*
* @ac: audio client pointer
* @flags: flags associated with write
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
}
EXPORT_SYMBOL_GPL(q6asm_run_nowait);
/**
* q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
*
* @ac: audio client pointer
* @rate: audio sample rate
* @channels: number of audio channels.
* @channel_map: channel map pointer
* @bits_per_sample: bits per sample
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
u8 channel_map[PCM_MAX_NUM_CHANNEL],
uint16_t bits_per_sample)
{
struct asm_multi_channel_pcm_fmt_blk_v2 *fmt;
struct apr_pkt *pkt;
u8 *channel_mapping;
void *p;
int rc, pkt_size;
pkt_size = APR_HDR_SIZE + sizeof(*fmt);
p = kzalloc(pkt_size, GFP_KERNEL);
if (!p)
return -ENOMEM;
pkt = p;
fmt = p + APR_HDR_SIZE;
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
fmt->num_channels = channels;
fmt->bits_per_sample = bits_per_sample;
fmt->sample_rate = rate;
fmt->is_signed = 1;
channel_mapping = fmt->channel_mapping;
if (channel_map) {
memcpy(channel_mapping, channel_map, PCM_MAX_NUM_CHANNEL);
} else {
if (q6dsp_map_channels(channel_mapping, channels)) {
dev_err(ac->dev, " map channels failed %d\n", channels);
rc = -EINVAL;
goto err;
}
}
rc = q6asm_ac_send_cmd_sync(ac, pkt);
err:
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
* @ac: audio client pointer
* @rate: audio sample rate
* @channels: number of audio channels.
* @bits_per_sample: bits per sample
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
{
struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg;
struct apr_pkt *pkt;
u8 *channel_mapping;
u32 frames_per_buf = 0;
int pkt_size, rc;
void *p;
pkt_size = APR_HDR_SIZE + sizeof(*enc_cfg);
p = kzalloc(pkt_size, GFP_KERNEL);
if (!p)
return -ENOMEM;
pkt = p;
enc_cfg = p + APR_HDR_SIZE;
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
enc_cfg->encdec.param_size = sizeof(*enc_cfg) - sizeof(enc_cfg->encdec);
enc_cfg->encblk.frames_per_buf = frames_per_buf;
enc_cfg->encblk.enc_cfg_blk_size = enc_cfg->encdec.param_size -
sizeof(struct asm_enc_cfg_blk_param_v2);
enc_cfg->num_channels = channels;
enc_cfg->bits_per_sample = bits_per_sample;
enc_cfg->sample_rate = rate;
enc_cfg->is_signed = 1;
channel_mapping = enc_cfg->channel_mapping;
if (q6dsp_map_channels(channel_mapping, channels)) {
rc = -EINVAL;
goto err;
}
rc = q6asm_ac_send_cmd_sync(ac, pkt);
err:
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
/**
* q6asm_read() - read data of period size from audio client
*
* @ac: audio client pointer
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_read(struct audio_client *ac)
{
struct asm_data_cmd_read_v2 *read;
struct audio_port_data *port;
struct audio_buffer *ab;
struct apr_pkt *pkt;
unsigned long flags;
int pkt_size;
int rc = 0;
void *p;
pkt_size = APR_HDR_SIZE + sizeof(*read);
p = kzalloc(pkt_size, GFP_ATOMIC);
if (!p)
return -ENOMEM;
pkt = p;
read = p + APR_HDR_SIZE;
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
ab = &port->buf[port->dsp_buf];
pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
read->buf_addr_lsw = lower_32_bits(ab->phys);
read->buf_addr_msw = upper_32_bits(ab->phys);
read->mem_map_handle = port->mem_map_handle;
read->buf_size = ab->size;
read->seq_id = port->dsp_buf;
pkt->hdr.token = port->dsp_buf;
port->dsp_buf++;
if (port->dsp_buf >= port->num_periods)
port->dsp_buf = 0;
spin_unlock_irqrestore(&ac->lock, flags);
rc = apr_send_pkt(ac->adev, pkt);
if (rc == pkt_size)
rc = 0;
else
pr_err("read op[0x%x]rc[%d]\n", pkt->hdr.opcode, rc);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_read);
static int __q6asm_open_read(struct audio_client *ac,
uint32_t format, uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_read_v3 *open;
struct apr_pkt *pkt;
int pkt_size, rc;
void *p;
pkt_size = APR_HDR_SIZE + sizeof(*open);
p = kzalloc(pkt_size, GFP_KERNEL);
if (!p)
return -ENOMEM;
pkt = p;
open = p + APR_HDR_SIZE;
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
/* Stream prio : High, provide meta info with encoded frames */
open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
open->preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
open->bits_per_sample = bits_per_sample;
open->mode_flags = 0x0;
open->mode_flags |= ASM_LEGACY_STREAM_SESSION <<
ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
switch (format) {
case FORMAT_LINEAR_PCM:
open->mode_flags |= 0x00;
open->enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
default:
pr_err("Invalid format[%d]\n", format);
}
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
/**
* q6asm_open_read() - Open audio client for reading
*
* @ac: audio client pointer
* @format: audio sample format
* @bits_per_sample: bits per sample
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_open_read(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_read(ac, format, bits_per_sample);
}
EXPORT_SYMBOL_GPL(q6asm_open_read);
/**
* q6asm_write_async() - non blocking write
*
* @ac: audio client pointer
* @len: lenght in bytes
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
* @wflags: flags associated with write
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t wflags)
{
struct asm_data_cmd_write_v2 *write;
struct audio_port_data *port;
struct audio_buffer *ab;
unsigned long flags;
struct apr_pkt *pkt;
int pkt_size;
int rc = 0;
void *p;
pkt_size = APR_HDR_SIZE + sizeof(*write);
p = kzalloc(pkt_size, GFP_ATOMIC);
if (!p)
return -ENOMEM;
pkt = p;
write = p + APR_HDR_SIZE;
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
ab = &port->buf[port->dsp_buf];
pkt->hdr.token = port->dsp_buf;
pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2;
write->buf_addr_lsw = lower_32_bits(ab->phys);
write->buf_addr_msw = upper_32_bits(ab->phys);
write->buf_size = len;
write->seq_id = port->dsp_buf;
write->timestamp_lsw = lsw_ts;
write->timestamp_msw = msw_ts;
write->mem_map_handle =
ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
if (wflags == NO_TIMESTAMP)
write->flags = (wflags & 0x800000FF);
else
write->flags = (0x80000000 | wflags);
port->dsp_buf++;
if (port->dsp_buf >= port->num_periods)
port->dsp_buf = 0;
spin_unlock_irqrestore(&ac->lock, flags);
rc = apr_send_pkt(ac->adev, pkt);
if (rc == pkt_size)
rc = 0;
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_write_async);
static void q6asm_reset_buf_state(struct audio_client *ac)
{
struct audio_port_data *port = NULL;
unsigned long flags;
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
port->dsp_buf = 0;
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
port->dsp_buf = 0;
spin_unlock_irqrestore(&ac->lock, flags);
}
static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
{
int stream_id = ac->stream_id;
struct apr_pkt pkt;
int rc;
q6asm_add_hdr(ac, &pkt.hdr, APR_HDR_SIZE, true, stream_id);
switch (cmd) {
case CMD_PAUSE:
pkt.hdr.opcode = ASM_SESSION_CMD_PAUSE;
break;
case CMD_SUSPEND:
pkt.hdr.opcode = ASM_SESSION_CMD_SUSPEND;
break;
case CMD_FLUSH:
pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH;
break;
case CMD_OUT_FLUSH:
pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
break;
case CMD_EOS:
pkt.hdr.opcode = ASM_DATA_CMD_EOS;
break;
case CMD_CLOSE:
pkt.hdr.opcode = ASM_STREAM_CMD_CLOSE;
break;
default:
return -EINVAL;
}
if (wait)
rc = q6asm_ac_send_cmd_sync(ac, &pkt);
else
return apr_send_pkt(ac->adev, &pkt);
if (rc < 0)
return rc;
if (cmd == CMD_FLUSH)
q6asm_reset_buf_state(ac);
return 0;
}
/**
* q6asm_cmd() - run cmd on audio client
*
* @ac: audio client pointer
* @cmd: command to run on audio client.
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_cmd(struct audio_client *ac, int cmd)
{
return __q6asm_cmd(ac, cmd, true);
}
EXPORT_SYMBOL_GPL(q6asm_cmd);
/**
* q6asm_cmd_nowait() - non blocking, run cmd on audio client
*
* @ac: audio client pointer
* @cmd: command to run on audio client.
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
{
return __q6asm_cmd(ac, cmd, false);
}
EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
static int q6asm_probe(struct apr_device *adev)
{

Просмотреть файл

@ -1,8 +1,36 @@
/* SPDX-License-Identifier: GPL-2.0 */
#ifndef __Q6_ASM_H__
#define __Q6_ASM_H__
#include "q6dsp-common.h"
#include <dt-bindings/sound/qcom,q6asm.h>
/* ASM client callback events */
#define CMD_PAUSE 0x0001
#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE 0x1001
#define CMD_FLUSH 0x0002
#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE 0x1002
#define CMD_EOS 0x0003
#define ASM_CLIENT_EVENT_CMD_EOS_DONE 0x1003
#define CMD_CLOSE 0x0004
#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE 0x1004
#define CMD_OUT_FLUSH 0x0005
#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE 0x1005
#define CMD_SUSPEND 0x0006
#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE 0x1006
#define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008
#define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009
#define ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a
enum {
LEGACY_PCM_MODE = 0,
LOW_LATENCY_PCM_MODE,
ULTRA_LOW_LATENCY_PCM_MODE,
ULL_POST_PROCESSING_PCM_MODE,
};
#define MAX_SESSIONS 8
#define NO_TIMESTAMP 0xFF00
#define FORMAT_LINEAR_PCM 0x0000
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
@ -11,6 +39,27 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
q6asm_cb cb, void *priv,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags);
int q6asm_open_write(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
int q6asm_open_read(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
int q6asm_read(struct audio_client *ac);
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
u8 channel_map[PCM_MAX_NUM_CHANNEL],
uint16_t bits_per_sample);
int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
uint32_t lsw_ts);
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
uint32_t lsw_ts);
int q6asm_cmd(struct audio_client *ac, int cmd);
int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
int q6asm_get_session_id(struct audio_client *ac);
int q6asm_map_memory_regions(unsigned int dir,
struct audio_client *ac,