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Subject[RESEND PATCH v2 08/15] ASoC: qcom: q6asm: add support to audio stream apis
Date
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

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

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 530 ++++++++++++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 42 ++++
2 files changed, 571 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 4be92441f524..dabd6509ef99 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -8,16 +8,34 @@
#include <linux/soc/qcom/apr.h>
#include <linux/device.h>
#include <linux/platform_device.h>
+#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
#include "q6asm.h"
#include "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 DEFAULT_POPP_TOPOLOGY 0x00010BE4
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
#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_SESSION_CMD_RUN_V2 0x00010DAA
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
+#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
+#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
+
+#define ASM_LEGACY_STREAM_SESSION 0
+#define ASM_END_POINT_DEVICE_MATRIX 0
+#define DEFAULT_APP_TYPE 0
+#define TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
#define TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
#define SYNC_IO_MODE 0x0001
#define ASYNC_IO_MODE 0x0002
@@ -42,6 +60,49 @@ 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 apr_hdr hdr;
+ 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[8];
+} __packed;
+
+struct asm_data_cmd_write_v2 {
+ struct apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ u32 flags;
+ u32 time_lsw;
+ u32 time_msw;
+} __packed;
+
struct audio_buffer {
dma_addr_t phys;
uint32_t used;
@@ -408,6 +469,111 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
return NULL;
}

+static int32_t q6asm_callback(struct apr_device *adev,
+ struct apr_client_data *data, int session_id)
+{
+ struct audio_client *ac;// = (struct audio_client *)priv;
+ uint32_t token;
+ uint32_t *payload;
+ uint32_t wakeup_flag = 1;
+ uint32_t client_event = 0;
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+
+ if (data == NULL)
+ return -EINVAL;
+
+ ac = q6asm_get_audio_client(q6asm, session_id);
+ if (!q6asm_is_valid_audio_client(ac))
+ return -EINVAL;
+
+ payload = data->payload;
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ token = data->token;
+ switch (payload[0]) {
+ 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;
+ 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_FLUSH_READBUFS:
+ if (token != ac->session) {
+ dev_err(ac->dev, "session invalid\n");
+ return -EINVAL;
+ }
+ case ASM_STREAM_CMD_CLOSE:
+ client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
+ break;
+ case ASM_STREAM_CMD_OPEN_WRITE_V3:
+ case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+ if (payload[1] != 0) {
+ dev_err(ac->dev,
+ "cmd = 0x%x returned error = 0x%x\n",
+ payload[0], payload[1]);
+ if (wakeup_flag) {
+ ac->cmd_state = payload[1];
+ wake_up(&ac->cmd_wait);
+ }
+ return 0;
+ }
+ break;
+ default:
+ dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+ payload[0]);
+ break;
+ }
+
+ if (ac->cmd_state && wakeup_flag) {
+ ac->cmd_state = 0;
+ wake_up(&ac->cmd_wait);
+ }
+ if (ac->cb)
+ ac->cb(client_event, data->token,
+ data->payload, ac->priv);
+
+ return 0;
+ }
+
+ switch (data->opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2:{
+ struct audio_port_data *port =
+ &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+ client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+
+ if (ac->io_mode & SYNC_IO_MODE) {
+ dma_addr_t phys = port->buf[data->token].phys;
+
+ if (lower_32_bits(phys) != payload[0] ||
+ upper_32_bits(phys) != payload[1]) {
+ dev_err(ac->dev, "Expected addr %pa\n",
+ &port->buf[data->token].phys);
+ return -EINVAL;
+ }
+ token = data->token;
+ port->buf[token].used = 1;
+ }
+ break;
+ }
+ }
+ if (ac->cb)
+ ac->cb(client_event, data->token, data->payload, ac->priv);
+
+ return 0;
+}
+
static int q6asm_srvc_callback(struct apr_device *adev, struct apr_client_data *data)
{
struct q6asm *a, *q6asm = dev_get_drvdata(&adev->dev);
@@ -415,12 +581,16 @@ static int q6asm_srvc_callback(struct apr_device *adev, struct apr_client_data *
struct audio_port_data *port;
uint32_t dir = 0;
uint32_t sid = 0;
+ int dest_port;
uint32_t *payload;

if (!data) {
dev_err(&adev->dev, "%s: Invalid CB\n", __func__);
return 0;
}
+ dest_port = (data->dest_port >> 8) & 0xFF;
+ if (dest_port)
+ return q6asm_callback(adev, data, dest_port);

payload = data->payload;
sid = (data->token >> 8) & 0x0F;
@@ -540,6 +710,364 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
}
EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);

+static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample, uint32_t stream_id,
+ bool is_gapless_mode)
+{
+ struct asm_stream_cmd_open_write_v3 open;
+ int rc;
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, stream_id);
+ ac->cmd_state = -1;
+
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
+ open.mode_flags = 0x00;
+ open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
+ if (is_gapless_mode)
+ open.mode_flags |= 1 << ASM_SHIFT_GAPLESS_MODE_FLAG;
+
+ /* source endpoint : matrix */
+ open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+ open.bits_per_sample = bits_per_sample;
+ open.postprocopo_id = DEFAULT_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);
+ return -EINVAL;
+ }
+ rc = apr_send_pkt(ac->adev, (uint32_t *) &open);
+ if (rc < 0)
+ return rc;
+
+ rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "timeout on open write\n");
+ return -ETIMEDOUT;
+ }
+
+ if (ac->cmd_state > 0)
+ return adsp_err_get_lnx_err_code(ac->cmd_state);
+
+ ac->io_mode |= TUN_WRITE_IO_MODE;
+
+ return 0;
+}
+
+/**
+ * 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)
+{
+ return __q6asm_open_write(ac, format, bits_per_sample,
+ ac->stream_id, false);
+}
+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;
+ int rc;
+
+ q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id);
+ ac->cmd_state = -1;
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run.flags = flags;
+ run.time_lsw = lsw_ts;
+ run.time_msw = msw_ts;
+
+ rc = apr_send_pkt(ac->adev, (uint32_t *) &run);
+ if (rc < 0)
+ return rc;
+
+ if (wait) {
+ rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0),
+ 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "timeout on run cmd\n");
+ return -ETIMEDOUT;
+ }
+ if (ac->cmd_state > 0)
+ return adsp_err_get_lnx_err_code(ac->cmd_state);
+ }
+
+ return 0;
+}
+
+/**
+ * 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.
+ * @use_default_chmap: flag to use default ch map.
+ * @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,
+ bool use_default_chmap,
+ char *channel_map,
+ uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
+ u8 *channel_mapping;
+ int rc = 0;
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id);
+ ac->cmd_state = -1;
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+ 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 (use_default_chmap) {
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ dev_err(ac->dev, " map channels failed %d\n", channels);
+ return -EINVAL;
+ }
+ } else {
+ memcpy(channel_mapping, channel_map,
+ PCM_FORMAT_MAX_NUM_CHANNEL);
+ }
+
+ rc = apr_send_pkt(ac->adev, (uint32_t *) &fmt);
+ if (rc < 0)
+ goto fail_cmd;
+
+ rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "timeout on format update\n");
+ return -ETIMEDOUT;
+ }
+ if (ac->cmd_state > 0)
+ return adsp_err_get_lnx_err_code(ac->cmd_state);
+
+ return 0;
+fail_cmd:
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+/**
+ * q6asm_write_nolock() - non blocking write
+ *
+ * @ac: audio client pointer
+ * @len: lenght in bytes
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ * @flags: flags associated with write
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags)
+{
+ struct asm_data_cmd_write_v2 write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int dsp_buf = 0;
+ int rc = 0;
+
+ if (ac->io_mode & SYNC_IO_MODE) {
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ q6asm_add_hdr(ac, &write.hdr, sizeof(write), false,
+ ac->stream_id);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.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 (flags == NO_TIMESTAMP)
+ write.flags = (flags & 0x800000FF);
+ else
+ write.flags = (0x80000000 | flags);
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->max_buf_cnt)
+ port->dsp_buf = 0;
+
+ rc = apr_send_pkt(ac->adev, (uint32_t *) &write);
+ if (rc < 0)
+ return rc;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_nolock);
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+ int cnt = 0;
+ int loopcnt = 0;
+ int used;
+ struct audio_port_data *port = NULL;
+
+ if (ac->io_mode & SYNC_IO_MODE) {
+ used = (ac->io_mode & TUN_WRITE_IO_MODE ? 1 : 0);
+ mutex_lock(&ac->cmd_lock);
+ for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE;
+ loopcnt++) {
+ port = &ac->port[loopcnt];
+ cnt = port->max_buf_cnt - 1;
+ port->dsp_buf = 0;
+ while (cnt >= 0) {
+ if (!port->buf)
+ continue;
+ port->buf[cnt].used = used;
+ cnt--;
+ }
+ }
+ mutex_unlock(&ac->cmd_lock);
+ }
+}
+
+static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+{
+ int stream_id = ac->stream_id;
+ struct apr_hdr hdr;
+ int rc;
+
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), true, stream_id);
+ ac->cmd_state = -1;
+ switch (cmd) {
+ case CMD_PAUSE:
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_SUSPEND:
+ hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+ break;
+ case CMD_FLUSH:
+ hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ break;
+ case CMD_OUT_FLUSH:
+ hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ break;
+ case CMD_EOS:
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ ac->cmd_state = 0;
+ break;
+ case CMD_CLOSE:
+ hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ rc = apr_send_pkt(ac->adev, (uint32_t *) &hdr);
+ if (rc < 0)
+ return rc;
+
+ if (!wait)
+ return 0;
+
+ rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "timeout response for opcode[0x%x]\n",
+ hdr.opcode);
+ return -ETIMEDOUT;
+ }
+ if (ac->cmd_state > 0)
+ return adsp_err_get_lnx_err_code(ac->cmd_state);
+
+ 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)
{
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index e1409c368600..b4896059da79 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -2,7 +2,34 @@
#ifndef __Q6_ASM_H__
#define __Q6_ASM_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 MSM_FRONTEND_DAI_MULTIMEDIA1 0
+#define MSM_FRONTEND_DAI_MULTIMEDIA2 1
+#define MSM_FRONTEND_DAI_MULTIMEDIA3 2
+#define MSM_FRONTEND_DAI_MULTIMEDIA4 3
+#define MSM_FRONTEND_DAI_MULTIMEDIA5 4
+#define MSM_FRONTEND_DAI_MULTIMEDIA6 5
+#define MSM_FRONTEND_DAI_MULTIMEDIA7 6
+#define MSM_FRONTEND_DAI_MULTIMEDIA8 7
+
#define MAX_SESSIONS 16
+#define NO_TIMESTAMP 0xFF00
+#define FORMAT_LINEAR_PCM 0x0000

typedef void (*app_cb) (uint32_t opcode, uint32_t token,
uint32_t *payload, void *priv);
@@ -10,6 +37,21 @@ struct audio_client;
struct audio_client *q6asm_audio_client_alloc(struct device *dev,
app_cb cb, void *priv);
void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_write_nolock(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_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ bool use_default_chmap,
+ char *channel_map,
+ 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,
--
2.15.0
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