nils/shairport-sync
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Clean up some of the sync timing. Ignore a flush if playing AP2. Set the requested lead time to zero.

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This commit is contained in:
Mike Brady
2021-05-11 10:46:56 +01:00
parent ca0c625b22
commit 2b88215f09
3 changed files with 21 additions and 23 deletions
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player.c
+15 -13
View File
@@ -1059,12 +1059,12 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
int64_t lt = conn->first_packet_time_to_play - local_time_now;
debug(1, "Connection %d: Lead time: %f seconds.", conn->connection_number,
debug(1, "Connection %d: Lead time for first frame %" PRId64 ": %f seconds.", conn->connection_number, conn->first_packet_timestamp,
lt * 0.000000001);
if (local_time_now > conn->first_packet_time_to_play) {
uint64_t lateness = local_time_now - conn->first_packet_time_to_play;
debug(2, "First packet is %" PRIu64 " nanoseconds late! Flushing 0.5 seconds",
int64_t lateness = local_time_now - conn->first_packet_time_to_play;
if (lateness > 0) {
debug(1, "First packet is %" PRId64 " nanoseconds late! Flushing 0.5 seconds",
lateness);
do_flush(conn->first_packet_timestamp + 5 * 4410, conn);
}
@@ -1094,18 +1094,17 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (fabs(0.000001 * change_in_should_be_time) >
0.001) // the clock drift estimation might be nudging the estimate, and we can
// ignore this unless if's more than a microsecond
debug(2, "Change in estimated first_packet_time: %8.4f milliseconds.",
debug(2, "Change in estimated first_packet_time: %f milliseconds for first_packet .",
0.000001 * change_in_should_be_time);
conn->first_packet_time_to_play = should_be_time;
if (local_time_now > conn->first_packet_time_to_play) {
uint64_t lateness = local_time_now - conn->first_packet_time_to_play;
debug(2, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness);
int64_t lead_time = conn->first_packet_time_to_play - local_time_now; // negative means late
if (lead_time < 0) {
debug(1, "Gone past starting time for %u by %" PRId64 " nanoseconds.", conn->first_packet_timestamp, -lead_time);
conn->ab_buffering = 0;
} else {
// do some calculations
int64_t lead_time = conn->first_packet_time_to_play - local_time_now;
if ((config.audio_backend_silent_lead_in_time_auto == 1) ||
(lead_time <=
(int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) {
@@ -1131,7 +1130,6 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (exact_frame_gap <= frames_needed_to_maintain_desired_buffer) {
fs = conn->max_frames_per_packet * 2;
}
// if we are very close to the end of buffering, i.e. within two frame-lengths,
// add the remaining silence needed and end buffering
if (exact_frame_gap <= conn->max_frames_per_packet * 2) {
@@ -1153,7 +1151,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
generate_zero_frames(silence, fs, config.output_format,
conn->enable_dither, conn->previous_random_number);
config.output->play(silence, fs);
// debug(1, "Sent %" PRId64 " frames of silence", fs);
debug(2, "Sent %" PRId64 " frames of silence", fs);
free(silence);
have_sent_prefiller_silence = 1;
}
@@ -1172,7 +1170,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
}
}
} else {
debug(2, "Unexpected response to getting dac delay: %d.", resp);
debug(1, "Unexpected response to getting dac delay: %d.", resp);
}
}
}
@@ -1978,6 +1976,7 @@ void *player_thread_func(void *arg) {
pthread_testcancel(); // allow a pthread_cancel request to take effect.
abuf_t *inframe = buffer_get_frame(conn); // this has cancellation point(s), but it's not
// guaranteed that they'll always be executed
if (inframe) {
inbuf = inframe->data;
inbuflength = inframe->length;
@@ -2208,6 +2207,8 @@ void *player_thread_func(void *arg) {
sync_error = delay; // int64_t from int64_t - int32_t, so okay
// debug(1, "Sync error on frame %u is %" PRId64 " frames.", inframe->given_timestamp, sync_error);
if (at_least_one_frame_seen_this_session == 0) {
at_least_one_frame_seen_this_session = 1;
@@ -2226,7 +2227,7 @@ void *player_thread_func(void *arg) {
// remove the bias when reporting the error to make it the true error
debug(2,
debug(1,
"first frame sync error (positive --> late): %" PRId64
" frames, %.3f mS at %d frames per second output.",
sync_error + first_frame_early_bias,
@@ -3025,6 +3026,7 @@ void player_flush(uint32_t timestamp, rtsp_conn_info *conn) {
}
void player_full_flush(rtsp_conn_info *conn) {
debug(3,"player_full_flush");
// this basically flushes everything from the player
// here, find the rtptime of the last from in the buffer and add 1 to it
// so as to ask to flush everything
rtp.c
+1 -1
View File
@@ -2100,7 +2100,7 @@ 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 = 0.4; //
double reqested_lead_time = 0.0; //
if (player_buffer_occupancy >
((reqested_lead_time + 0.1) * 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
rtsp.c
+5 -9
View File
@@ -1810,7 +1810,7 @@ void handle_flush(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) {
rtptime = uatoi(p + 1); // unsigned integer -- up to 2^32-1
}
}
// debug(1,"RTSP Flush Requested: %u.",rtptime);
debug(1,"RTSP Flush Requested: %u.",rtptime);
if (have_play_lock(conn)) {
#ifdef CONFIG_METADATA
if (p)
@@ -1818,19 +1818,15 @@ void handle_flush(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) {
else
send_metadata('ssnc', 'flsr', NULL, 0, NULL, 0);
#endif
player_flush(rtptime, conn); // will not crash even it there is no player thread.
// hack -- ignore it for airplay 2
if (conn->airplay_type != ap_2)
player_flush(rtptime, conn); // will not crash even it there is no player thread.
resp->respcode = 200;
} else {
warn("Connection %d FLUSH %u received without having the player (no ANNOUNCE?)",
warn("Connection %d FLUSH %u received without having the player",
conn->connection_number, rtptime);
resp->respcode = 451;
// #ifdef CONFIG_AIRPLAY_2
// debug(1, "Connection %d FLUSH %u received without having the player but do it anyway...",
// conn->connection_number, rtptime);
// player_flush(rtptime, conn); // will not crash even it there is no player thread.
// resp->respcode = 200;
// #endif
}
}