On 12/14/2017 11:03 PM, srinivas.kandaga...@linaro.org wrote:
From: Srinivas Kandagatla <srinivas.kandaga...@linaro.org>

This patch adds support to open, write and media format commands
in the q6asm module.
[..]
+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;
+
ac could get freed by q6asm_audio_client_free during the execution of q6asm_callback as they are running in different thread.
Add synchronization.
+       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;
+}
+
[..]
+/**
+ * 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;
asm_multi_channel_pcm_fmt_blk_v4 is now being used in latest adsp. Better to add adsp version based support to handle different struct
+       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,

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