ASoC: msm: Check stream id for ASM callbacks in compress driver
Send stream id to DSP for asm commands which requires stream and
session information. Stream and session id is added as token
parameter for DSP commands. Use token parameter to interpret the
stream for which the callback is received.
Conflicts:
sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
Change-Id: I58a40d84f60f4a4172ec7ee79302c8bfb24c910b
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
diff --git a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
index c78e436..41ef3e3 100644
--- a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
@@ -66,9 +66,22 @@
const DECLARE_TLV_DB_LINEAR(msm_compr_vol_gain, 0,
COMPRESSED_LR_VOL_MAX_STEPS);
+/*
+ * LSB 8 bits is used as stream id for some DSP
+ * commands for compressed playback.
+ */
+#define STREAM_ID_FROM_TOKEN(i) (i & 0xFF)
+
+/* Stream id switches between 1 and 2 */
+#define NEXT_STREAM_ID(stream_id) ((stream_id & 1) + 1)
+
+#define STREAM_ARRAY_INDEX(stream_id) (stream_id - 1)
+
+#define MAX_NUMBER_OF_STREAMS 2
+
struct msm_compr_gapless_state {
bool set_next_stream_id;
- int32_t stream_opened[2];
+ int32_t stream_opened[MAX_NUMBER_OF_STREAMS];
uint32_t initial_samples_drop;
uint32_t trailing_samples_drop;
uint32_t gapless_transition;
@@ -210,7 +223,8 @@
pr_debug("%s: bytes_received = %d copied_total = %d\n",
__func__, prtd->bytes_received, prtd->copied_total);
if (prtd->first_buffer && prtd->gapless_state.use_dsp_gapless_mode)
- q6asm_send_meta_data(prtd->audio_client,
+ q6asm_stream_send_meta_data(prtd->audio_client,
+ prtd->audio_client->stream_id,
prtd->gapless_state.initial_samples_drop,
prtd->gapless_state.trailing_samples_drop);
@@ -260,6 +274,7 @@
uint32_t chan_mode = 0;
uint32_t sample_rate = 0;
int bytes_available, stream_id;
+ uint32_t stream_index;
pr_debug("%s opcode =%08x\n", __func__, opcode);
switch (opcode) {
@@ -317,8 +332,10 @@
spin_unlock(&prtd->lock);
break;
case ASM_DATA_EVENT_RENDERED_EOS:
- pr_debug("ASM_DATA_CMDRSP_EOS\n");
spin_lock(&prtd->lock);
+ pr_debug("%s: ASM_DATA_CMDRSP_EOS token 0x%x,stream id %d\n",
+ __func__, token, STREAM_ID_FROM_TOKEN(token));
+ stream_id = STREAM_ID_FROM_TOKEN(token);
if (atomic_read(&prtd->eos) &&
!prtd->gapless_state.set_next_stream_id) {
pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
@@ -326,13 +343,22 @@
wake_up(&prtd->eos_wait);
}
atomic_set(&prtd->eos, 0);
- stream_id = ac->stream_id^1; /*prev stream */
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS ||
+ stream_index < 0) {
+ pr_err("%s: Invalid stream index %d", __func__,
+ stream_index);
+ spin_unlock(&prtd->lock);
+ break;
+ }
+
if (prtd->gapless_state.set_next_stream_id &&
- prtd->gapless_state.stream_opened[stream_id]) {
- q6asm_stream_cmd_nowait(prtd->audio_client,
- CMD_CLOSE, stream_id);
+ prtd->gapless_state.stream_opened[stream_index]) {
+ pr_debug("%s: CMD_CLOSE stream_id %d\n",
+ __func__, stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_CLOSE, stream_id);
atomic_set(&prtd->close, 1);
- prtd->gapless_state.stream_opened[stream_id] = 0;
+ prtd->gapless_state.stream_opened[stream_index] = 0;
prtd->gapless_state.set_next_stream_id = false;
}
if (prtd->gapless_state.gapless_transition)
@@ -369,25 +395,35 @@
spin_unlock(&prtd->lock);
break;
case ASM_STREAM_CMD_FLUSH:
- pr_debug("ASM_STREAM_CMD_FLUSH\n");
+ pr_debug("%s: ASM_STREAM_CMD_FLUSH:", __func__);
+ pr_debug("token 0x%x, stream id %d\n", token,
+ STREAM_ID_FROM_TOKEN(token));
prtd->cmd_ack = 1;
wake_up(&prtd->flush_wait);
break;
case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
- pr_debug("ASM_DATA_CMD_REMOVE_INITIAL_SILENCE\n");
+ pr_debug("%s: ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:",
+ __func__);
+ pr_debug("token 0x%x, stream id = %d\n", token,
+ STREAM_ID_FROM_TOKEN(token));
break;
case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:
- pr_debug("ASM_DATA_CMD_REMOVE_TRAILING_SILENCE\n");
+ pr_debug("%s: ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:",
+ __func__);
+ pr_debug("token = 0x%x, stream id = %d\n", token,
+ STREAM_ID_FROM_TOKEN(token));
break;
case ASM_STREAM_CMD_CLOSE:
- pr_debug("ASM_DATA_CMD_CLOSE\n");
+ pr_debug("%s: ASM_DATA_CMD_CLOSE:", __func__);
+ pr_debug("token 0x%x, stream id %d\n", token,
+ STREAM_ID_FROM_TOKEN(token));
/*
* wakeup wait for stream avail on stream 3
* after stream 1 ends.
*/
if (prtd->next_stream) {
pr_debug("%s:CLOSE:wakeup wait for stream\n",
- __func__);
+ __func__);
prtd->stream_available = 1;
wake_up(&prtd->wait_for_stream_avail);
prtd->next_stream = 0;
@@ -405,10 +441,12 @@
break;
}
case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3:
- pr_debug("ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n");
+ pr_debug("%s: ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n",
+ __func__);
break;
case RESET_EVENTS:
- pr_err("Received reset events CB, move to error state");
+ pr_err("%s: Received reset events CB, move to error state",
+ __func__);
spin_lock(&prtd->lock);
snd_compr_fragment_elapsed(cstream);
prtd->copied_total = prtd->bytes_received;
@@ -416,7 +454,8 @@
spin_unlock(&prtd->lock);
break;
default:
- pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ pr_debug("%s: Not Supported Event opcode[0x%x]\n",
+ __func__, opcode);
break;
}
}
@@ -486,6 +525,7 @@
uint16_t bits_per_sample = 16;
int dir = IN, ret = 0;
struct audio_client *ac = prtd->audio_client;
+ uint32_t stream_index;
struct asm_softpause_params softpause = {
.enable = SOFT_PAUSE_ENABLE,
.period = SOFT_PAUSE_PERIOD,
@@ -498,7 +538,7 @@
.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
};
- pr_debug("%s\n", __func__);
+ pr_debug("%s: stream_id %d\n", __func__, ac->stream_id);
ret = q6asm_stream_open_write_v2(ac,
prtd->codec, bits_per_sample,
ac->stream_id,
@@ -508,7 +548,13 @@
return -ENOMEM;
}
- prtd->gapless_state.stream_opened[ac->stream_id] = 1;
+ stream_index = STREAM_ARRAY_INDEX(ac->stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) {
+ pr_err("%s: Invalid stream index:%d", __func__, stream_index);
+ return -EINVAL;
+ }
+
+ prtd->gapless_state.stream_opened[stream_index] = 1;
pr_debug("%s be_id %d\n", __func__, soc_prtd->dai_link->be_id);
msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
ac->perf_mode,
@@ -529,7 +575,7 @@
pr_err("%s: Send SoftVolume Param failed ret=%d\n",
__func__, ret);
- ret = q6asm_set_io_mode(ac, (COMPRESSED_IO | ASYNC_IO_MODE));
+ ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE));
if (ret < 0) {
pr_err("%s: Set IO mode failed\n", __func__);
return -EINVAL;
@@ -673,6 +719,7 @@
struct audio_client *ac = prtd->audio_client;
int dir = IN, ret = 0, stream_id;
unsigned long flags;
+ uint32_t stream_index;
pr_debug("%s\n", __func__);
@@ -693,13 +740,19 @@
spin_lock_irqsave(&prtd->lock, flags);
stream_id = ac->stream_id;
- if (prtd->gapless_state.stream_opened[stream_id^1]) {
+ stream_index = STREAM_ARRAY_INDEX(NEXT_STREAM_ID(stream_id));
+
+ if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) &&
+ (prtd->gapless_state.stream_opened[stream_index])) {
spin_unlock_irqrestore(&prtd->lock, flags);
- pr_debug(" close stream %d", stream_id^1);
- q6asm_stream_cmd(ac, CMD_CLOSE, stream_id^1);
+ pr_debug(" close stream %d", NEXT_STREAM_ID(stream_id));
+ q6asm_stream_cmd(ac, CMD_CLOSE, NEXT_STREAM_ID(stream_id));
spin_lock_irqsave(&prtd->lock, flags);
}
- if (prtd->gapless_state.stream_opened[stream_id]) {
+
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) &&
+ (prtd->gapless_state.stream_opened[stream_index])) {
spin_unlock_irqrestore(&prtd->lock, flags);
pr_debug("close stream %d", stream_id);
q6asm_stream_cmd(ac, CMD_CLOSE, stream_id);
@@ -719,8 +772,6 @@
pr_debug("%s: ocmem_req: %d\n", __func__,
atomic_read(&pdata->audio_ocmem_req));
- /* client buf alloc was with stream id 0, so free with the same */
- ac->stream_id = 0;
q6asm_audio_client_buf_free_contiguous(dir, ac);
q6asm_audio_client_free(ac);
@@ -892,6 +943,7 @@
int bytes_to_write;
unsigned long flags;
int stream_id;
+ uint32_t stream_index;
if (cstream->direction != SND_COMPRESS_PLAYBACK) {
pr_err("%s: Unsupported stream type\n", __func__);
@@ -944,9 +996,9 @@
atomic_set(&prtd->drain, 0);
}
prtd->last_buffer = 0;
- pr_debug("issue CMD_FLUSH\n");
prtd->cmd_ack = 0;
if (!prtd->gapless_state.gapless_transition) {
+ pr_debug("issue CMD_FLUSH stream_id %d\n", stream_id);
spin_unlock_irqrestore(&prtd->lock, flags);
rc = q6asm_stream_cmd(
prtd->audio_client, CMD_FLUSH, stream_id);
@@ -981,7 +1033,9 @@
pr_debug("SNDRV_PCM_TRIGGER_PAUSE_PUSH transition %d\n",
prtd->gapless_state.gapless_transition);
if (!prtd->gapless_state.gapless_transition) {
- q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ pr_debug("issue CMD_PAUSE stream_id %d\n",
+ ac->stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id);
atomic_set(&prtd->start, 0);
}
break;
@@ -1057,7 +1111,8 @@
/* send EOS */
prtd->cmd_ack = 0;
- q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ pr_debug("issue CMD_EOS stream_id %d\n", ac->stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id);
pr_info("PARTIAL DRAIN, do not wait for EOS ack\n");
/* send a zero length buffer */
@@ -1094,8 +1149,9 @@
}
/* move to next stream and reset vars */
- pr_debug("%s: Moving to next stream in gapless\n", __func__);
- ac->stream_id ^= 1;
+ pr_debug("%s: Moving to next stream in gapless\n",
+ __func__);
+ ac->stream_id = NEXT_STREAM_ID(ac->stream_id);
prtd->byte_offset = 0;
prtd->app_pointer = 0;
prtd->first_buffer = 1;
@@ -1122,11 +1178,11 @@
stream can be used for gapless playback
*/
prtd->gapless_state.set_next_stream_id = false;
- pr_debug("%s: CMD_EOS\n", __func__);
+ pr_debug("%s:CMD_EOS stream_id %d\n", __func__, ac->stream_id);
prtd->cmd_ack = 0;
atomic_set(&prtd->eos, 1);
- q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id);
spin_unlock_irqrestore(&prtd->lock, flags);
@@ -1138,7 +1194,8 @@
if (rc < 0)
pr_err("%s: EOS wait failed\n", __func__);
- pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait for EOS\n", __func__);
+ pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait for EOS\n",
+ __func__);
if (prtd->cmd_interrupt)
rc = -EINTR;
@@ -1150,12 +1207,14 @@
* so prepare the current stream in session for gapless playback
*/
spin_lock_irqsave(&prtd->lock, flags);
- pr_debug("%s: issue CMD_PAUSE ", __func__);
- q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ pr_debug("%s:issue CMD_PAUSE stream_id %d",
+ __func__, ac->stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id);
prtd->cmd_ack = 0;
spin_unlock_irqrestore(&prtd->lock, flags);
- pr_debug("%s: issue CMD_FLUSH", __func__);
- q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ pr_debug("%s:issue CMD_FLUSH ac->stream_id %d",
+ __func__, ac->stream_id);
+ q6asm_stream_cmd(ac, CMD_FLUSH, ac->stream_id);
wait_event_timeout(prtd->flush_wait,
prtd->cmd_ack, 1 * HZ / 4);
@@ -1191,14 +1250,25 @@
pr_debug("%s: SND_COMPR_TRIGGER_NEXT_TRACK\n", __func__);
spin_lock_irqsave(&prtd->lock, flags);
rc = 0;
- stream_id = ac->stream_id^1; /*next stream in gapless*/
+ /* next stream in gapless */
+ stream_id = NEXT_STREAM_ID(ac->stream_id);
/*
* Wait if stream 1 has not completed before honoring next
* track for stream 3. Scenario happens if second clip is
* small and fills in one buffer so next track will be
* called immediately.
*/
- if (prtd->gapless_state.stream_opened[stream_id]) {
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS ||
+ stream_index < 0) {
+ pr_err("%s: Invalid stream index: %d", __func__,
+ stream_index);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ rc = -EINVAL;
+ break;
+ }
+
+ if (prtd->gapless_state.stream_opened[stream_index]) {
if (prtd->gapless_state.gapless_transition) {
rc = msm_compr_wait_for_stream_avail(prtd,
&flags);
@@ -1229,6 +1299,7 @@
}
break;
}
+ pr_debug("%s: open_write stream_id %d", __func__, stream_id);
rc = q6asm_stream_open_write_v2(prtd->audio_client,
prtd->codec, 16,
stream_id,
@@ -1245,7 +1316,7 @@
break;
}
spin_lock_irqsave(&prtd->lock, flags);
- prtd->gapless_state.stream_opened[stream_id] = 1;
+ prtd->gapless_state.stream_opened[stream_index] = 1;
prtd->gapless_state.set_next_stream_id = true;
spin_unlock_irqrestore(&prtd->lock, flags);
break;