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Much better flushing. Delayed flushing only works for one delay. Noice artifacts removed when skipping. Pretty solid.

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This commit is contained in:
Mike Brady
2021-05-29 21:24:33 +01:00
parent 7c68f97d7f
commit 2ea28c0661
3 changed files with 40 additions and 17 deletions
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player.h
+2
View File
@@ -385,6 +385,8 @@ typedef struct {
extern pthread_mutex_t playing_conn_lock;
void reset_buffer(rtsp_conn_info *conn);
void get_audio_buffer_size_and_occupancy(unsigned int *size, unsigned int *occupancy,
rtsp_conn_info *conn);
rtp.c
+36 -16
View File
@@ -2052,13 +2052,15 @@ void *rtp_buffered_audio_processor(void *arg) {
int flush_requested = 0;
int streaming_has_started = 0;
int flush_is_delayed;
int play_enabled = 0;
uint32_t flush_from_timestamp;
int dump_block_info = 0;
double requested_lead_time = 0.100; // normal lead time minimum
do {
int flush_is_delayed = 0;
int flush_newly_requested = 0;
int flush_newly_complete = 0;
int play_newly_stopped = 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;
@@ -2105,7 +2107,7 @@ void *rtp_buffered_audio_processor(void *arg) {
// blocks_read to ensure seq_no is valid
if ((blocks_read != 0) && (seq_no >= flushUntilSeq)) {
// we have reached or overshot the flushUntilSeq block
if (flushUntilSeq != seq_no)
// if (flushUntilSeq != seq_no)
debug(1,"flushUntilSeq %u reached or overshot at %u.", flushUntilSeq, seq_no);
conn->ap2_flush_requested = 0;
flush_request_active = 0;
@@ -2116,23 +2118,34 @@ void *rtp_buffered_audio_processor(void *arg) {
flush_newly_complete = 1;
flush_requested = flush_request_active;
// flush_requested = conn->ap2_flush_requested;
int play_enabled = conn->ap2_play_enabled;
if ((play_enabled) && (conn->ap2_play_enabled == 0))
play_newly_stopped = 1;
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.");
blocks_read = 0;
}
if (play_newly_stopped != 0)
reset_buffer(conn); // stop play ASAP
if (flush_newly_requested) {
reset_buffer(conn);
if (flush_is_delayed == 0) {
debug(1,"Immediate Buffered Audio Flush Started.");
player_full_flush(conn);
//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.");
streaming_has_started = 0;
pcm_buffer_occupancy = 0;
pcm_buffer_read_point = 0;
}
@@ -2145,9 +2158,8 @@ 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 requested_lead_time = 0.10; //
if (player_buffer_occupancy >
((requested_lead_time + 0.20) * 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
@@ -2170,22 +2182,23 @@ 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 (blocks_read > 1) {
if ((lead_time >= (int64_t)(requested_lead_time * 1000000000)) ||
(streaming_has_started == 1)) {
(streaming_has_started != 0)) {
if (streaming_has_started == 0)
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;
//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);
//}
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);
pcm_buffer + pcm_buffer_read_point, 352, conn);
streaming_has_started++;
usleep(2000);
}
}
pcm_buffer_read_point_rtptime += 352;
pcm_buffer_read_point += 352 * 4;
@@ -2299,10 +2312,17 @@ void *rtp_buffered_audio_processor(void *arg) {
// debug(1,"seq_no %u, timestamp %u", seq_no, timestamp);
if ((flush_requested) && (seq_no >= flushUntilSeq)) {
uint64_t should_be_time;
if ((frame_to_local_time(timestamp, &should_be_time, conn) ==
0) && (play_enabled)) {
// play enabled will be off when this is a full flush and the anchor information is not valid
int64_t lead_time = should_be_time - get_absolute_time_in_ns();
debug(1,"flush completed to seq: %u with rtptime: %u, lead time: 0x%" PRIx64 " nanoseconds, i.e. %f sec.", seq_no, timestamp, lead_time, lead_time * 0.000000001);
} else {
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))) {
rtsp.c
+2 -1
View File
@@ -1350,6 +1350,7 @@ void handle_flushbuffered(rtsp_conn_info *conn, rtsp_message *req, rtsp_message
// if it's a flush that will be followed by a setanchor (i.e. a play) then stop play now.
if (flushFromValid == 0)
conn->ap2_play_enabled = 0;
conn->ap2_flush_from_sequence_number = flushFromSeq;
conn->ap2_flush_from_rtp_timestamp = flushFromTS;
conn->ap2_flush_until_sequence_number = flushUntilSeq;
@@ -1444,7 +1445,7 @@ void handle_setrateanchori(rtsp_conn_info *conn, rtsp_message *req, rtsp_message
debug(1, "Connection %d: No timing peer list!", conn->connection_number);
}
} else {
player_full_flush(conn);
// player_full_flush(conn);
// ptp_send_control_message_string("T"); // ensure an obsolete clock isn't picked up later.
}
}