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Subject[PATCH v3 11/25] 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>
---
include/dt-bindings/sound/qcom,q6asm.h | 22 ++
sound/soc/qcom/qdsp6/q6asm.c | 503 ++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 41 +++
3 files changed, 564 insertions(+), 2 deletions(-)
create mode 100644 include/dt-bindings/sound/qcom,q6asm.h

diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h
new file mode 100644
index 000000000000..4e85bf804cec
--- /dev/null
+++ b/include/dt-bindings/sound/qcom,q6asm.h
@@ -0,0 +1,22 @@
+// SPDX-License-Identifier: GPL-2.0
+#ifndef __DT_BINDINGS_Q6_ASM_H__
+#define __DT_BINDINGS_Q6_ASM_H__
+
+#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 MSM_FRONTEND_DAI_MULTIMEDIA9 8
+#define MSM_FRONTEND_DAI_MULTIMEDIA10 9
+#define MSM_FRONTEND_DAI_MULTIMEDIA11 10
+#define MSM_FRONTEND_DAI_MULTIMEDIA12 11
+#define MSM_FRONTEND_DAI_MULTIMEDIA13 12
+#define MSM_FRONTEND_DAI_MULTIMEDIA14 13
+#define MSM_FRONTEND_DAI_MULTIMEDIA15 14
+#define MSM_FRONTEND_DAI_MULTIMEDIA16 15
+
+#endif /* __DT_BINDINGS_Q6_ASM_H__ */
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 412275edb15c..0ee1e30a8d8e 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -10,6 +10,7 @@
#include <linux/soc/qcom/apr.h>
#include <linux/device.h>
#include <linux/of.h>
+#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
@@ -17,10 +18,26 @@
#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_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 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 +63,49 @@ 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[PCM_FORMAT_MAX_NUM_CHANNEL];
+} __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 {
phys_addr_t phys;
uint32_t used;
@@ -131,7 +191,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,

rc = wait_event_timeout(a->mem_wait, (a->mem_state <= 0), 5 * HZ);
if (!rc) {
- dev_err(a->dev, "CMD timeout \n");
+ dev_err(a->dev, "CMD timeout\n");
rc = -ETIMEDOUT;
} else if (a->mem_state < 0) {
rc = q6dsp_errno(a->mem_state);
@@ -395,6 +455,108 @@ void *q6asm_get_dai_data(struct device *dev)
}
EXPORT_SYMBOL_GPL(q6asm_get_dai_data);

+static int32_t q6asm_stream_callback(struct apr_device *adev,
+ struct apr_client_message *data,
+ int session_id)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct audio_port_data *port;
+ struct audio_client *ac;
+ uint32_t token;
+ uint32_t client_event = 0;
+
+ ac = q6asm_get_audio_client(q6asm, session_id);
+ if (!ac)/* Audio client might already be freed by now */
+ return 0;
+
+ if (!q6asm_is_valid_audio_client(ac))
+ return -EINVAL;
+
+ result = data->payload;
+
+ switch (data->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ token = data->token;
+ 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;
+ 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 (result->status != 0) {
+ dev_err(ac->dev,
+ "cmd = 0x%x returned error = 0x%x\n",
+ result->opcode, result->status);
+ ac->cmd_state = -result->status;
+ wake_up(&ac->cmd_wait);
+ return 0;
+ }
+ break;
+ default:
+ dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+
+ if (ac->cmd_state) {
+ ac->cmd_state = 0;
+ wake_up(&ac->cmd_wait);
+ }
+ if (ac->cb)
+ ac->cb(client_event, data->token,
+ data->payload, ac->priv);
+
+ return 0;
+
+ case ASM_DATA_EVENT_WRITE_DONE_V2:
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+ client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ phys_addr_t phys = port->buf[data->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[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_message *data)
{
@@ -404,6 +566,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct audio_port_data *port;
uint32_t dir = 0;
uint32_t sid = 0;
+ int session_id;
+
+ session_id = (data->dest_port >> 8) & 0xFF;
+ if (session_id)
+ return q6asm_stream_callback(adev, data, session_id);

result = data->payload;
sid = (data->token >> 8) & 0x0F;
@@ -519,6 +686,338 @@ 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, void *cmd)
+{
+ int rc;
+
+ mutex_lock(&ac->lock);
+ ac->cmd_state = 1;
+
+ rc = apr_send_pkt(ac->adev, cmd);
+ if (rc < 0)
+ goto err;
+
+ rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state <= 0), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ goto err;
+ }
+
+ if (ac->cmd_state > 0)
+ rc = q6dsp_errno(ac->cmd_state);
+
+err:
+ mutex_unlock(&ac->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;
+ int rc;
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id);
+
+ open.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_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 = q6asm_ac_send_cmd_sync(ac, &open);
+ if (rc < 0)
+ return rc;
+
+ ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+ return 0;
+}
+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;
+
+ q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run.flags = flags;
+ run.time_lsw = lsw_ts;
+ run.time_msw = msw_ts;
+ if (wait)
+ return q6asm_ac_send_cmd_sync(ac, &run);
+ else
+ return apr_send_pkt(ac->adev, &run);
+
+}
+
+/**
+ * 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,
+ u8 channel_map[PCM_FORMAT_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
+ u8 *channel_mapping;
+ int rc;
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id);
+
+ 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 (channel_map) {
+ memcpy(channel_mapping, channel_map,
+ PCM_FORMAT_MAX_NUM_CHANNEL);
+ } else {
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ dev_err(ac->dev, " map channels failed %d\n", channels);
+ return -EINVAL;
+ }
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, &fmt);
+ if (rc < 0)
+ goto fail_cmd;
+
+ return 0;
+fail_cmd:
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+/**
+ * q6asm_write_async() - 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_async(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 rc = 0;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return 0;
+
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ q6asm_add_hdr(ac, &write.hdr, sizeof(write), false,
+ ac->stream_id);
+
+ ab = &port->buf[port->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->num_periods)
+ port->dsp_buf = 0;
+
+ rc = apr_send_pkt(ac->adev, &write);
+ if (rc < 0)
+ return rc;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_async);
+
+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 & ASM_SYNC_IO_MODE))
+ return;
+
+ used = (ac->io_mode & ASM_TUN_WRITE_IO_MODE ? 1 : 0);
+ mutex_lock(&ac->lock);
+ for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE;
+ loopcnt++) {
+ port = &ac->port[loopcnt];
+ cnt = port->num_periods - 1;
+ port->dsp_buf = 0;
+ while (cnt >= 0) {
+ if (!port->buf)
+ continue;
+ port->buf[cnt].used = used;
+ cnt--;
+ }
+ }
+ mutex_unlock(&ac->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);
+
+ 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;
+ break;
+ case CMD_CLOSE:
+ hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (wait)
+ rc = q6asm_ac_send_cmd_sync(ac, &hdr);
+ else
+ return apr_send_pkt(ac->adev, &hdr);
+
+ 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)
{
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index a4f9fe636b7e..b5ef90bb724b 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -1,8 +1,35 @@
// 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
+
+enum {
+ LEGACY_PCM_MODE = 0,
+ LOW_LATENCY_PCM_MODE,
+ ULTRA_LOW_LATENCY_PCM_MODE,
+ ULL_POST_PROCESSING_PCM_MODE,
+};

#define MAX_SESSIONS 16
+#define NO_TIMESTAMP 0xFF00
+#define FORMAT_LINEAR_PCM 0x0000

void q6asm_set_dai_data(struct device *dev, void *data);
void *q6asm_get_dai_data(struct device *dev);
@@ -14,6 +41,20 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
q6asm_cb cb, void *priv,
int session_id);
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_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ u8 channel_map[PCM_FORMAT_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,
--
2.15.1
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