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Accept a new master clock immediately. Don't truncate the first three blocks if it's a normal start. Hacky.

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Mike Brady
2021-06-06 15:53:55 +01:00
parent 5c48580db9
commit f6a9f634c7
1 changed files with 20 additions and 13 deletions
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rtp.c
+20 -13
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@@ -1284,9 +1284,10 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
// using the master clock timing
if (actual_clock_id == conn->anchor_clock) {
// if the master clock and the anchor clock are the same
// wait at least this time before using the new master clock
if (duration_of_mastership < 700000000) {
//if the master clock and the anchor clock are the same
//wait at least this time before using the new master clock
if (duration_of_mastership < 0000000) {
debug(1,"master not old enough yet: %f ms", 0.000001 * duration_of_mastership);
response = clock_not_ready;
} else if ((duration_of_mastership > 5000000000) ||
(conn->last_anchor_info_is_valid == 0)) {
@@ -1414,7 +1415,7 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
debug(1, "Connection %d: NQPTP clock information is not valid.", conn->connection_number);
break;
default:
debug(3, "Connection %d: NQPTP clock reports an unrecognised status: %u.",
debug(1, "Connection %d: NQPTP clock reports an unrecognised status: %u.",
conn->connection_number, response);
break;
}
@@ -2139,8 +2140,8 @@ void *rtp_buffered_audio_processor(void *arg) {
av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(swr, "in_sample_rate", 44100, 0);
av_opt_set_int(swr, "out_sample_rate", config.output_rate, 0);
av_opt_set_int(swr, "in_sample_rate", conn->input_rate, 0);
av_opt_set_int(swr, "out_sample_rate", conn->input_rate, 0); // must match or the timing will be wrong`
av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
enum AVSampleFormat av_format;
@@ -2202,12 +2203,13 @@ void *rtp_buffered_audio_processor(void *arg) {
// uint32_t expected_rtptime;
uint64_t blocks_read = 0;
uint64_t blocks_read_since_flush = 0;
int flush_requested = 0;
int streaming_has_started = 0;
int play_enabled = 0;
uint32_t flush_from_timestamp;
double requested_lead_time = 0.10; // normal lead time minimum
double requested_lead_time = 0.3; // normal lead time minimum
reset_buffer(conn); // in case there is any garbage in the player
// int not_first_time_out = 0;
do {
@@ -2279,7 +2281,8 @@ void *rtp_buffered_audio_processor(void *arg) {
// do this outside the flush mutex
if (flush_newly_complete) {
debug(2, "Flush Complete.");
blocks_read = 0;
blocks_read_since_flush = 0;
}
if (play_newly_stopped != 0)
@@ -2307,10 +2310,9 @@ void *rtp_buffered_audio_processor(void *arg) {
// is there space in the player thread's buffer system?
unsigned int player_buffer_size, player_buffer_occupancy;
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);
//debug(1,"player buffer size and occupancy: %u and %u", player_buffer_size, player_buffer_occupancy);
if (player_buffer_occupancy >
((requested_lead_time + 0.4) * 44100.0 / 352)) { // must be greater than the lead time.
((requested_lead_time + 0.4) * conn->input_rate / 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
usleep(1000); // wait for a while
} else {
@@ -2331,8 +2333,10 @@ 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();
// debug(1,"lead time in buffered_audio is %f milliseconds.", lead_time * 0.000001);
if (blocks_read > 3) {
// debug(3,"lead time in buffered_audio is %f milliseconds.", lead_time * 0.000001);
// if it's the very first block (thus no priming needed)
if ((blocks_read == 1) || (blocks_read_since_flush > 3)) {
//if (1) {
if ((lead_time >= (int64_t)(requested_lead_time * 1000000000)) ||
(streaming_has_started != 0)) {
if (streaming_has_started == 0)
@@ -2373,6 +2377,8 @@ void *rtp_buffered_audio_processor(void *arg) {
pcm_buffer_read_point_rtptime += 352;
pcm_buffer_read_point += 352 * conn->input_bytes_per_frame;
} else {
debug(1,"frame to local time error");
}
} else {
usleep(1000); // wait before asking if play is enabled again
@@ -2427,6 +2433,7 @@ void *rtp_buffered_audio_processor(void *arg) {
errno, errorstring);
} else if (nread > 0) {
blocks_read++; // note, this doesn't mean they are valid audio blocks
blocks_read_since_flush++;
// debug(1, "Realtime Audio Receiver Packet of length %d received.", nread);
// now get hold of its various bits and pieces
/*