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Improving the flush behaviour.

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This commit is contained in:
Mike Brady
2021-05-29 10:54:44 +01:00
parent 2986b282d3
commit 9cf4ef032d
2 changed files with 41 additions and 17 deletions
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rtp.c
+37 -16
View File
@@ -2046,12 +2046,17 @@ void *rtp_buffered_audio_processor(void *arg) {
int pcm_buffer_occupancy = 0;
int pcm_buffer_read_point = 0; // offset to where the next buffer should come from
uint32_t pcm_buffer_read_point_rtptime = 0;
uint32_t expected_rtptime;
uint64_t blocks_read = 0;
int flush_requested = 0;
int streaming_has_started = 0;
int flush_is_delayed;
do {
int flush_newly_requested = 0;
int flush_newly_complete = 0;
// are we in in flush mode, or just about to leave it?
debug_mutex_lock(&conn->flush_mutex, 10000, 1); // 10ms is a long time to wait!
uint32_t flushUntilSeq = conn->ap2_flush_until_sequence_number;
@@ -2061,11 +2066,13 @@ void *rtp_buffered_audio_processor(void *arg) {
if (conn->ap2_flush_requested) {
if (conn->ap2_flush_from_valid == 0) {// i.e. a flush from right now
flush_request_active = 1;
flush_is_delayed = 0;
} else {
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;
// debug(1, "delayed flush request going active");
flush_is_delayed = 1;
}
}
}
@@ -2102,16 +2109,29 @@ void *rtp_buffered_audio_processor(void *arg) {
flush_newly_requested = 0;
}
}
if ((flush_requested) && (flush_request_active == 0))
flush_newly_complete = 1;
flush_requested = flush_request_active;
// flush_requested = conn->ap2_flush_requested;
int play_enabled = conn->ap2_play_enabled;
debug_mutex_unlock(&conn->flush_mutex, 3);
// 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) {
streaming_has_started = 0;
pcm_buffer_occupancy = 0;
pcm_buffer_read_point = 0;
if (flush_is_delayed == 0) {
debug(1,"Immediate Buffered Audio Flush Started.");
// player_full_flush(conn);
// streaming_has_started = 0;
pcm_buffer_occupancy = 0;
pcm_buffer_read_point = 0;
} else {
debug(1,"Delayed Buffered Audio Flush Started.");
}
}
// now, if a flush is not requested, we can do the normal stuff
@@ -2121,9 +2141,9 @@ void *rtp_buffered_audio_processor(void *arg) {
get_audio_buffer_size_and_occupancy(&player_buffer_size, &player_buffer_occupancy, conn);
// debug(1,"player buffer size and occupancy: %u and %u", player_buffer_size,
// player_buffer_occupancy);
double reqested_lead_time = 0.10; //
double requested_lead_time = 0.10; //
if (player_buffer_occupancy >
((reqested_lead_time + 0.1) * 44100.0 / 352)) { // must be greater than the lead time.
((requested_lead_time + 0.10) * 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
@@ -2146,15 +2166,21 @@ void *rtp_buffered_audio_processor(void *arg) {
0) {
int64_t lead_time = buffer_should_be_time - get_absolute_time_in_ns();
// debug(1,"lead time in buffered_audio is %f milliseconds.", lead_time * 0.000001);
if ((lead_time >= (int64_t)(reqested_lead_time * 1000000000)) ||
if ((lead_time >= (int64_t)(requested_lead_time * 1000000000)) ||
(streaming_has_started == 1)) {
if (streaming_has_started == 0)
debug(3, "Connection %d: buffered audio starting frame: %u, lead time: %f seconds.", conn->connection_number, pcm_buffer_read_point_rtptime,
debug(1, "Connection %d: buffered audio starting frame: %u, lead time: %f seconds.", conn->connection_number, pcm_buffer_read_point_rtptime,
0.000000001 * lead_time);
else {
if (expected_rtptime != pcm_buffer_read_point_rtptime)
debug(1,"actual rtptime is %u, expected rtptime is %u.", pcm_buffer_read_point_rtptime, expected_rtptime);
}
streaming_has_started = 1;
expected_rtptime = pcm_buffer_read_point_rtptime + 352;
player_put_packet(0, 0, pcm_buffer_read_point_rtptime,
pcm_buffer + pcm_buffer_read_point, 352, conn);
usleep(2000);
}
pcm_buffer_read_point_rtptime += 352;
@@ -2230,9 +2256,9 @@ void *rtp_buffered_audio_processor(void *arg) {
// uint8_t marker = 0;
// uint8_t payload_type = 0;
/*
if ((blocks_read != 0) && (seq_no != previous_seq_no + 1)) {
if (previous_seq_no != 0)
debug(1, "block discontinuity: from sequence number %u to sequence number %u.",
previous_seq_no, seq_no);
@@ -2243,13 +2269,13 @@ void *rtp_buffered_audio_processor(void *arg) {
pcm_buffer_read_point_rtptime, previous_seq_no, timestamp, seq_no);
pcm_buffer_occupancy = 0;
}
// if still playing, this is a problem -- need to reset the player
if (play_enabled)
player_full_flush(conn);
}
*/
// previous_seq_no = seq_no;
// at this point, we can check if we can to flush this packet -- we won't have
@@ -2258,11 +2284,6 @@ void *rtp_buffered_audio_processor(void *arg) {
if (((flush_requested) && (seq_no == flushUntilSeq)) || (new_buffer_needed)) {
if ((flush_requested) && (seq_no == flushUntilSeq)) {
player_full_flush(conn);
debug(1,"Flush completed.");
}
// 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
// so we better indicate that the pcm buffer is empty and its contents invalid
rtsp.c
+4 -1
View File
@@ -1357,7 +1357,10 @@ void handle_flushbuffered(rtsp_conn_info *conn, rtsp_message *req, rtsp_message
conn->ap2_flush_requested = 1;
debug_mutex_unlock(&conn->flush_mutex, 3);
// player_full_flush(conn);
if (flushFromValid)
debug(1,"Deferred Flush Requested");
else
debug(1,"Immediate Flush Requested");
resp->respcode = 200;
}