A little more debug information.
This commit is contained in:
@@ -352,8 +352,8 @@ void *rtp_control_receiver(void *arg) {
|
||||
obfp += 2;
|
||||
};
|
||||
*obfp = 0;
|
||||
|
||||
|
||||
|
||||
|
||||
// get raw timestamp information
|
||||
// I think that a good way to understand these timestamps is that
|
||||
// (1) the rtlt below is the timestamp of the frame that should be playing at the
|
||||
@@ -364,19 +364,19 @@ void *rtp_control_receiver(void *arg) {
|
||||
// Thus, (3) the latency can be calculated by subtracting the second from the
|
||||
// first.
|
||||
// There must be more to it -- there something missing.
|
||||
|
||||
|
||||
// In addition, it seems that if the value of the short represented by the second
|
||||
// pair of bytes in the packet is 7
|
||||
// then an extra time lag is expected to be added, presumably by
|
||||
// the AirPort Express.
|
||||
|
||||
|
||||
// Best guess is that this delay is 11,025 frames.
|
||||
|
||||
|
||||
uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
|
||||
uint32_t rt = nctohl(&packet[16]); // raw timestamp
|
||||
|
||||
|
||||
uint32_t fl = nctohs(&packet[2]); //
|
||||
|
||||
|
||||
debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and %u,
|
||||
giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
|
||||
//debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
|
||||
@@ -2640,12 +2640,11 @@ void *rtp_buffered_audio_processor(void *arg) {
|
||||
if (have_time_information == 0) {
|
||||
int64_t play_time_since_connection = local_should_be_time - conn->connection_start_time;
|
||||
int64_t time_since_connection = get_absolute_time_in_ns() - conn->connection_start_time;
|
||||
|
||||
too_soon_after_connection =
|
||||
((play_time_since_connection < 2000000000) && (time_since_connection < 2000000000));
|
||||
if (too_soon_after_connection)
|
||||
debug(3, "time_since_connection is %f milliseconds. too_soon_after_connection is %d.",
|
||||
time_since_connection * 1E-6, too_soon_after_connection);
|
||||
debug(3, "time_since_connection is %f milliseconds. play_time_since_connection is %f milliseconds. lead_time is %f milliseconds. too_soon_after_connection is %d.",
|
||||
time_since_connection * 1E-6, play_time_since_connection * 1E-6, (play_time_since_connection - time_since_connection) * 1E-6, too_soon_after_connection);
|
||||
local_lead_time = local_should_be_time - get_absolute_time_in_ns();
|
||||
// debug(1,"local_lead_time is actually %f milliseconds.", local_lead_time * 1E-6);
|
||||
outdated = (local_lead_time < requested_lead_time_ns);
|
||||
@@ -2654,7 +2653,6 @@ void *rtp_buffered_audio_processor(void *arg) {
|
||||
// %" PRId64 " ns.", outdated, local_lead_time, requested_lead_time_ns);
|
||||
} else {
|
||||
debug(3, "Timing information not valid");
|
||||
// outdated = 1;
|
||||
}
|
||||
|
||||
if ((flush_requested) && (seq_no >= flushUntilSeq)) {
|
||||
|
||||
Reference in New Issue
Block a user