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flaky but some interesting stuff

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This commit is contained in:
Mike Brady
2021-05-29 15:59:11 +01:00
parent 9cf4ef032d
commit 7c68f97d7f
3 changed files with 48 additions and 17 deletions
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player.c
+5 -6
View File
@@ -930,6 +930,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
flush_needed = 1;
drop_request = 1;
} else {
debug(1,"flush");
if ((conn->ab_synced) && ((conn->ab_write - conn->ab_read) > 0)) {
abuf_t *firstPacket = conn->audio_buffer + BUFIDX(conn->ab_read);
abuf_t *lastPacket = conn->audio_buffer + BUFIDX(conn->ab_write - 1);
@@ -2308,18 +2309,16 @@ void *player_thread_func(void *arg) {
int64_t filler_length =
(int64_t)(config.resyncthreshold * config.output_rate); // number of samples
if ((sync_error > 0) && (sync_error > filler_length)) {
debug(1, "Large positive sync error of: %" PRId64 " frames (%f seconds).",
sync_error, (sync_error * 1.0) / config.output_rate);
debug(1, "Large positive sync error of: %" PRId64 " frames (%f seconds), with frame: %u.",
sync_error, (sync_error * 1.0) / config.output_rate, inframe->given_timestamp);
int64_t local_frames_to_drop = sync_error / conn->output_sample_ratio;
uint32_t frames_to_drop_sized = local_frames_to_drop;
do_flush(inframe->given_timestamp + frames_to_drop_sized, conn);
} else if ((sync_error < 0) && ((-sync_error) > filler_length)) {
debug(1,
"Large negative sync error of: %" PRId64
" frames (%f seconds), with should_be_frame_32 of %" PRIu32 ", nt of %" PRId64
" and current_delay of %" PRId64 ".",
sync_error, (sync_error * 1.0) / config.output_rate, should_be_frame_32,
inframe->given_timestamp * conn->output_sample_ratio, current_delay);
" frames (%f seconds), with frame: %" PRIu32 ".",
sync_error, (sync_error * 1.0) / config.output_rate,inframe->given_timestamp);
int64_t silence_length = -sync_error;
if (silence_length > (filler_length * 5))
silence_length = filler_length * 5;
rtp.c
+42 -10
View File
@@ -2053,6 +2053,8 @@ void *rtp_buffered_audio_processor(void *arg) {
int streaming_has_started = 0;
int flush_is_delayed;
uint32_t flush_from_timestamp;
int dump_block_info = 0;
do {
int flush_newly_requested = 0;
@@ -2067,12 +2069,13 @@ void *rtp_buffered_audio_processor(void *arg) {
if (conn->ap2_flush_from_valid == 0) {// i.e. a flush from right now
flush_request_active = 1;
flush_is_delayed = 0;
} else {
flush_is_delayed = 1;
dump_block_info = 1;
flush_from_timestamp = conn->ap2_flush_from_rtp_timestamp;
int32_t blocks_to_start_of_flush = conn->ap2_flush_from_sequence_number - seq_no;
if (blocks_to_start_of_flush <= 0) {
flush_request_active = 1;
flush_is_delayed = 1;
}
}
}
@@ -2080,7 +2083,7 @@ void *rtp_buffered_audio_processor(void *arg) {
if (flush_request_active) {
if (flush_requested == 0) {
// here, a flush has been newly requested
/*
debug(1,"Flush requested:");
if (conn->ap2_flush_from_valid) {
debug(1," fromTS: %u", conn->ap2_flush_from_rtp_timestamp);
@@ -2096,7 +2099,7 @@ void *rtp_buffered_audio_processor(void *arg) {
uint32_t endTS = fib + pcm_buffer_read_point_rtptime;
debug(1," currentTS_End: %u", endTS); // a frame occupies 4 bytes
debug(1," currentSeq: %u", seq_no);
*/
flush_newly_requested = 1;
}
// blocks_read to ensure seq_no is valid
@@ -2119,18 +2122,19 @@ void *rtp_buffered_audio_processor(void *arg) {
// do this outside the flush mutex
if (flush_newly_complete) {
debug(1,"Flush Complete.");
if (flush_is_delayed == 0)
player_full_flush(conn);
}
if (flush_newly_requested) {
if (flush_is_delayed == 0) {
debug(1,"Immediate Buffered Audio Flush Started.");
// player_full_flush(conn);
// streaming_has_started = 0;
player_full_flush(conn);
streaming_has_started = 0;
pcm_buffer_occupancy = 0;
pcm_buffer_read_point = 0;
dump_block_info = 0;
} else {
debug(1,"Delayed Buffered Audio Flush Started.");
pcm_buffer_occupancy = 0;
pcm_buffer_read_point = 0;
}
}
@@ -2143,7 +2147,7 @@ void *rtp_buffered_audio_processor(void *arg) {
// player_buffer_occupancy);
double requested_lead_time = 0.10; //
if (player_buffer_occupancy >
((requested_lead_time + 0.10) * 44100.0 / 352)) { // must be greater than the lead time.
((requested_lead_time + 0.20) * 44100.0 / 352)) { // must be greater than the lead time.
// if there is enough stuff in the player's buffer, sleep for a while and try again
// debug(1,"sleep for 20 ms");
usleep(20000); // wait for a while
@@ -2256,6 +2260,18 @@ void *rtp_buffered_audio_processor(void *arg) {
// uint8_t marker = 0;
// uint8_t payload_type = 0;
/*
if (dump_block_info != 0) {
if ((flush_requested) && (seq_no < flushUntilSeq))
debug(1, "block %u, rtptime %u, should be skipped.", seq_no, timestamp);
else
debug(1, "block %u, rtptime %u, should be decoded and used.", seq_no, timestamp);
if (((flush_requested != 0) && (seq_no == flushUntilSeq)) || ((flush_requested == 0) && (new_buffer_needed))) {
debug(1, "block %u, rtptime %u, will be decoded and used.", seq_no, timestamp);
}
}
*/
/*
if ((blocks_read != 0) && (seq_no != previous_seq_no + 1)) {
@@ -2282,7 +2298,13 @@ void *rtp_buffered_audio_processor(void *arg) {
// to decipher it first
// debug(1,"seq_no %u, timestamp %u", seq_no, timestamp);
if (((flush_requested) && (seq_no == flushUntilSeq)) || (new_buffer_needed)) {
if ((flush_requested) && (seq_no >= flushUntilSeq)) {
debug(1,"flush completed to seq: %u with rtptime: %u.", seq_no, timestamp);
}
if (((flush_requested != 0) && (seq_no == flushUntilSeq)) || ((flush_requested == 0) && (new_buffer_needed))) {
// if we are here because of a flush request, it must be the case that
// flushing the pcm buffer wasn't enough, as the request would have been turned off by now
@@ -2382,6 +2404,16 @@ void *rtp_buffered_audio_processor(void *arg) {
// debug(1,"generated %d bytes of PCM", dst_bufsize);
// copy the PCM audio into the PCM buffer.
// make sure it's big enough first
// also, check it if needs to be truncated but to an impending delayed flush_is_delayed
if (flush_is_delayed) {
// see if the flush_from_timestamp is in the buffer
int32_t samples_remaining = (flush_from_timestamp - pcm_buffer_read_point_rtptime);
if ((samples_remaining > 0) && ((samples_remaining * 4) < dst_bufsize)) {
debug(1,"samples remaining before flush: %d, number of samples %d. flushFromTS: %u, pcm_buffer_read_point_rtptime: %u.", samples_remaining, dst_bufsize/4, flush_from_timestamp, pcm_buffer_read_point_rtptime);
dst_bufsize = samples_remaining * 4;
}
}
if ((pcm_buffer_size - pcm_buffer_occupancy) < dst_bufsize) {
debug(1,
"pcm_buffer_read_point (frames): %u, pcm_buffer_occupancy "
rtsp.c
+1 -1
View File
@@ -3822,7 +3822,7 @@ static void *rtsp_conversation_thread_func(void *pconn) {
rtsp_message *req, *resp;
#ifdef CONFIG_AIRPLAY_2
conn->ap2_audio_buffer_size = 1024 * 1024 * 128;
conn->ap2_audio_buffer_size = 1024 * 1024 * 8;
#endif
while (conn->stop == 0) {