From: Srinivas Kandagatla <srinivas.kandaga...@linaro.org>

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

Signed-off-by: Srinivas Kandagatla <srinivas.kandaga...@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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