nils/shairport-sync
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Fix a bug that caused SPS to stop if another speaker with its own master clock (e.g. a HomePod mini) was included and then removed from the output group.

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This commit is contained in:
Mike Brady
2022-06-19 13:13:09 +01:00
parent 2260853ebc
commit 6556a79dbc
1 changed files with 19 additions and 48 deletions
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rtp.c
+19 -48
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@@ -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 "
@@ -1274,7 +1274,8 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
void set_ptp_anchor_info(rtsp_conn_info *conn, uint64_t clock_id, uint32_t rtptime,
uint64_t networktime) {
// debug(1,"clock: %" PRIx64 ", rtptime: %" PRIu32 ".", clock_id, rtptime);
// debug(1,"set_ptp_anchor_info: clock: %" PRIx64 ", rtptime: %" PRIu32 ", networktime: %" PRIx64
// ".", clock_id, rtptime, networktime);
if (conn->anchor_clock != clock_id) {
debug(2, "Connection %d: Set Anchor Clock: %" PRIx64 ".", conn->connection_number, clock_id);
}
@@ -1318,12 +1319,6 @@ void set_ptp_anchor_info(rtsp_conn_info *conn, uint64_t clock_id, uint32_t rtpti
conn->anchor_rtptime = rtptime;
conn->anchor_time = networktime;
conn->anchor_clock = clock_id;
// these are used to identify when the master clock becomes equal to the
// actual anchor clock information, so it can be used to avoid accumulating errors
conn->actual_anchor_rtptime = rtptime;
conn->actual_anchor_time = networktime;
conn->actual_anchor_clock = clock_id;
}
void reset_ptp_anchor_info(rtsp_conn_info *conn) {
@@ -1378,50 +1373,22 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
int64_t duration_of_mastership = time_now - start_of_mastership;
debug(2,
"Connection %d: Master clock has changed to %" PRIx64
". History: %f milliseconds.",
". History: %.3f milliseconds.",
conn->connection_number, actual_clock_id, 0.000001 * duration_of_mastership);
// Here we adjust the time of the anchor rtptime.
// We know its local time, so we use the new clocks's offset to
// calculate what time that must be on the new clock.
// Now, the thing is that while the anchor clock and master clock for a
// buffered session starts off the same,
// buffered session start off the same,
// the master clock can change without the anchor clock changing.
// SPS gives the new master clock time to settle down and then
// calculates the appropriate offset to it by
// calculating back from the local anchor information and the new clock's
// advertised offset. Of course, small errors will occur.
// Equally importantly, the new master clock(s) and the original one will
// drift at different rates. So, after all this, if the original master
// clock becomes the master again, there could be quite a difference
// in the time information that was calculated through all the clock
// changes and the actual master clock's time information.
// What do we do? We can hardly ignore this new and reliable information
// so we'll take it. Maybe we should add code to slowly correct towards it
// but at present, we just take it.
// advertised offset.
// So, if the master clock has again become equal to the actual anchor clock
// then we can reinstate it all.
// First, let us calculate the cumulative offset after swapping all the clocks...
conn->anchor_time = conn->last_anchor_local_time + actual_offset;
// Here we can check how much of a deviation there was going from clock to clock and
// back around to the master clock
if (actual_clock_id == conn->actual_anchor_clock) {
int64_t cumulative_deviation = conn->anchor_time - conn->actual_anchor_time;
debug(2,
"The original master clock has become the master clock again."
"The estimated clock time "
"was %f ms ahead(+) or behind (-) the original master clock time.",
0.000001 * cumulative_deviation);
conn->anchor_clock = conn->actual_anchor_clock;
conn->anchor_time = conn->actual_anchor_time;
conn->anchor_rtptime = conn->actual_anchor_rtptime;
} else {
conn->anchor_clock = actual_clock_id;
}
conn->anchor_clock = actual_clock_id;
}
} else {
response = clock_not_valid; // no current clock information and no previous clock info
@@ -2624,8 +2591,12 @@ void *rtp_buffered_audio_processor(void *arg) {
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. 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);
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);