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Correct the treatment of audio backend latency offset in Realtime streams. Fix a warning if the offset is too negative and that would require frames to be played before they could possibly arrive.

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This commit is contained in:
Mike Brady
2026-08-20 10:55:30 +01:00
parent d8165c1143
commit b69e01cdf9
1 changed files with 26 additions and 45 deletions
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rtp.c
+26 -45
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@@ -1619,6 +1619,7 @@ int32_t decipher_player_put_packet(uint8_t *ciphered_audio_alt, ssize_t nread,
}
void *rtp_ap2_control_receiver(void *arg) {
const int32_t ap2_realttime_stream_latency_fudge_factor = 11025; // seems to bring everything into sync
// #include <syscall.h>
// debug(1, "rtp_ap2_control_receiver PID %d", syscall(SYS_gettid));
pthread_cleanup_push(rtp_ap2_control_handler_cleanup_handler, arg);
@@ -1677,22 +1678,6 @@ void *rtp_ap2_control_receiver(void *arg) {
switch (packet[1]) {
case 215: // code 215, effectively an anchoring announcement
{
// struct timespec tnr;
// clock_gettime(CLOCK_REALTIME, &tnr);
// uint64_t local_realtime_now = timespec_to_ns(&tnr);
/*
char obf[4096];
char *obfp = obf;
int obfc;
for (obfc=0;obfc<nread;obfc++) {
snprintf(obfp, 3, "%02X", packet[obfc]);
obfp+=2;
};
*obfp=0;
debug(1,"AP2 Timing Control Received: \"%s\"",obf);
*/
uint64_t remote_packet_time_ns = nctoh64(packet + 8);
check64conversion("remote_packet_time_ns", packet + 8, remote_packet_time_ns);
uint64_t clock_id = nctoh64(packet + 20);
@@ -1702,49 +1687,45 @@ void *rtp_ap2_control_receiver(void *arg) {
// debug(1,"remote_packet_time_ns: %" PRIx64 ", local_realtime_now_ns: %" PRIx64
// ".", remote_packet_time_ns, local_realtime_now);
uint32_t frame_1 =
nctohl(packet + 4); // this seems to be the frame with latency of 77165 included
nctohl(packet + 4); // this seems to be the frame with latency of 77175 included
check32conversion("frame_1", packet + 4, frame_1);
uint32_t frame_2 =
nctohl(packet + 16); // this seems to be the frame the time refers to
check32conversion("frame_2", packet + 16, frame_2);
// this just updates the anchor information contained in the packet
// the frame and its remote time
// add in the audio_backend_latency_offset;
int32_t notified_latency = frame_2 - frame_1;
if (notified_latency != 77175)
debug(1, "Notified latency is %d frames.", notified_latency);
int32_t added_latency =
(int32_t)(config.audio_backend_latency_offset * conn->input_rate);
// the actual latency is the notified latency plus the fixed latency + the added
// latency
// this updates the anchor information contained in the packet
int32_t stream_specified_latency = frame_2 - frame_1; // this is the latency expected
if (stream_specified_latency != 77175)
debug(1, "Stream-specified latency is %d frames. Normally it is 77175.", stream_specified_latency);
int32_t net_source_latency = stream_specified_latency + ap2_realttime_stream_latency_fudge_factor;
int32_t net_latency =
notified_latency + 11035 +
added_latency; // this is the latency between incoming frames and the DAC
net_latency = net_latency - (int32_t)(config.audio_backend_buffer_desired_length *
// Now to accommodate a backend buffer of the desired length.
// Note that it's in input-rate frames, not output-rate frames!
net_source_latency = net_source_latency - (int32_t)(config.audio_backend_buffer_desired_length *
conn->input_rate);
// debug(1, "Net latency is %d frames.", net_latency);
// Now we want to check the user-specified latency offset.
// We want to warn the user if they have asked for a negative latency that is too great --
// one that would require packets to arrive before they actually do,
// (which is about two seconds before they are to be played).
int32_t net_latency = net_source_latency + (int32_t)(config.audio_backend_latency_offset * conn->input_rate);
if (net_latency <= 0) {
if (conn->latency_warning_issued == 0) {
warn("The stream latency (%f seconds) it too short to accommodate an offset of "
"%f "
"seconds and a backend buffer of %f seconds.",
((notified_latency + 11035) * 1.0) / conn->input_rate,
warn("The stream latency (%g seconds) is too short to accommodate an audio backend latency offset of "
"%g seconds and a backend buffer of %g seconds. The audio_backend_latency_offset has been set to zero.",
((stream_specified_latency + ap2_realttime_stream_latency_fudge_factor) * 1.0) / conn->input_rate,
config.audio_backend_latency_offset,
config.audio_backend_buffer_desired_length);
warn("(FYI the stream latency needed would be %f seconds.)",
config.audio_backend_buffer_desired_length -
config.audio_backend_latency_offset);
config.audio_backend_latency_offset = 0.0;
net_latency = net_source_latency;
conn->latency_warning_issued = 1;
}
conn->latency = notified_latency + 11035;
} else {
conn->latency = notified_latency + 11035 + added_latency;
}
set_ptp_anchor_info(conn, clock_id, frame_1 - 11035 - added_latency,
remote_packet_time_ns);
conn->latency = net_latency; // this is the time window within which packets can be accepted without being too late
set_ptp_anchor_info(conn, clock_id, frame_1 - ap2_realttime_stream_latency_fudge_factor, remote_packet_time_ns);
if (conn->anchor_clock != clock_id) {
debug(2, "Connection %d: Change Anchor Clock: %" PRIx64 ".",
conn->connection_number, clock_id);