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Clean up some of the code for accepting or diacarding buffered audio depending on its timeliness.

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Mike Brady
2022-03-16 12:38:05 +11:00
parent e7098c4529
commit 1dc481c3d3
1 changed files with 3 additions and 5 deletions
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rtp.c
+3 -5
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@@ -2515,15 +2515,13 @@ void *rtp_buffered_audio_processor(void *arg) {
if (frame_to_local_time(pcm_buffer_read_point_rtptime, &buffer_should_be_time, conn) ==
0) {
int64_t lead_time = buffer_should_be_time - get_absolute_time_in_ns();
double lead_time_ms = lead_time * 0.000001;
// debug(1,"lead time in buffered_audio is %f milliseconds.", lead_time_ms);
// it seems that some garbage blocks can be left after the flush, so
// only accept them if they have sensible lead times
if ((lead_time_ms < 5000.0) && (lead_time > -1000.0)) {
if ((lead_time < (int64_t)5000000000L) && (lead_time >= 0)) {
// if it's the very first block (thus no priming needed)
if ((blocks_read == 1) || (blocks_read_since_flush > 3)) {
if ((lead_time >= (int64_t)(requested_lead_time * 1000000000)) ||
if ((lead_time >= (int64_t)(requested_lead_time * 1000000000L)) ||
(streaming_has_started != 0)) {
if (streaming_has_started == 0)
debug(2,
@@ -2563,7 +2561,7 @@ void *rtp_buffered_audio_processor(void *arg) {
} else {
debug(2,
"Dropping packet %u from block %u with out-of-range lead_time: %.3f seconds.",
pcm_buffer_read_point_rtptime, seq_no, 0.001 * lead_time_ms);
pcm_buffer_read_point_rtptime, seq_no, 0.000000001 * lead_time);
}
pcm_buffer_read_point_rtptime += 352;