Look for at least 0.1 seconds of leadtime, ensure a master clock is at least 0.275 seconds old before use, quieten a few debug messages.
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@@ -1062,8 +1062,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
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conn->first_packet_time_to_play = should_be_time;
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int64_t lt = conn->first_packet_time_to_play - local_time_now;
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debug(1, "Connection %d: Lead time for first frame %" PRId64 ": %f seconds.",
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if (lt < 100000000)
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debug(1, "Connection %d: Short lead time for first frame %" PRId64 ": %f seconds.",
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conn->connection_number, conn->first_packet_timestamp, lt * 0.000000001);
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int64_t lateness = local_time_now - conn->first_packet_time_to_play;
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@@ -1310,7 +1310,7 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
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// if the master clock and the anchor clock are the same
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// wait at least this time before using the new master clock
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// note that mastershipo may be backdated
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if (duration_of_mastership < 100000000) {
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if (duration_of_mastership < 275000000) {
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debug(3, "master not old enough yet: %f ms", 0.000001 * duration_of_mastership);
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response = clock_not_ready;
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} else if ((duration_of_mastership > 5000000000) ||
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@@ -2254,7 +2254,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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int streaming_has_started = 0;
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int play_enabled = 0;
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uint32_t flush_from_timestamp;
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double requested_lead_time = 0.0; // normal lead time minimum -- maybe it should be about 0.1
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double requested_lead_time = 0.1; // normal lead time minimum -- maybe it should be about 0.1
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// wait until our timing information is valid
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@@ -2642,7 +2642,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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if (pkt->size) {
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// if (0) {
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if (pkt->size <= 7) { // no idea about this...
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debug(1, "malformed AAC packet skipped.");
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debug(2, "malformed AAC packet skipped.");
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} else {
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ret = avcodec_send_packet(codec_context, pkt);
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@@ -697,7 +697,7 @@ void cancel_all_RTSP_threads(airplay_stream_c stream_category, int except_this_o
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((stream_category == unspecified_stream_category) ||
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(stream_category == conns[i]->airplay_stream_category))) {
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pthread_cancel(conns[i]->thread);
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debug(1, "Connection %d: cancelled.", conns[i]->connection_number);
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debug(2, "Connection %d: cancelled.", conns[i]->connection_number);
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}
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}
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for (i = 0; i < nconns; i++) {
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@@ -706,7 +706,7 @@ void cancel_all_RTSP_threads(airplay_stream_c stream_category, int except_this_o
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((stream_category == unspecified_stream_category) ||
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(stream_category == conns[i]->airplay_stream_category))) {
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pthread_join(conns[i]->thread, NULL);
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debug(1, "Connection %d: joined.", conns[i]->connection_number);
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debug(2, "Connection %d: joined.", conns[i]->connection_number);
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free(conns[i]);
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conns[i] = NULL;
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}
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@@ -2625,7 +2625,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
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uint8_t isRemoteControlOnlyBoolean = 0;
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plist_get_bool_val(isRemoteControlOnly, &isRemoteControlOnlyBoolean);
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if (isRemoteControlOnlyBoolean != 0) {
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debug(1, "Connection %d: SETUP: Remote Control setup detected.",
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debug(2, "Connection %d: SETUP: Remote Control setup detected.",
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conn->connection_number);
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conn->airplay_stream_category = remote_control_stream;
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} else {
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@@ -2825,7 +2825,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
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conn->connection_number, err);
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}
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debug(1, "Connection %d: TCP Remote Control event port opened: %u.",
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debug(2, "Connection %d: TCP Remote Control event port opened: %u.",
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conn->connection_number, conn->local_event_port);
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if (conn->rtp_event_thread != NULL)
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debug(1, "previous rtp_event_thread allocation not freed, it seems.");
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@@ -3056,7 +3056,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
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resp->respcode = 200;
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} else if (conn->airplay_stream_category == remote_control_stream) {
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debug(2, "Connection %d: SETUP: Remote Control Stream received.", conn->connection_number);
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debug_log_rtsp_message(1, "Remote Control Stream stream (second) message", req);
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debug_log_rtsp_message(2, "Remote Control Stream stream (second) message", req);
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resp->respcode = 200;
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} else {
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