Re: [alsa-devel] [RESEND PATCH v2 08/15] ASoC: qcom: q6asm: add support to audio stream apis

On 12/14/2017 11:03 PM, srinivas.kandagatla@linaro.org wrote:

From: Srinivas Kandagatla srinivas.kandagatla@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,