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

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On 12/14/2017 11:03 PM, srinivas.kandagatla@xxxxxxxxxx wrote:
From: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx>

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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