nils/shairport-sync
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Lot of changes here.

Browse Source 
1. check_classic_mac_basic.yml — update to latest macOS-latest
2. docker-on-push-tag-or-pr — dependant updates
3. FFTConvolver — fix a compilation warning when the DEBUG flag is turned off
4. Reorganise some source files — e.g. move most of the metadata stuff into a subdirectory
5. Remove unused special-purpose AirPlay 2 RC event handler.
6. Reorganise source files and source code relating to the standard event handler.
7. Check configuration settings a little more carefully to offer more meaningful diagnostics.
8. Add password to AirPlay 2 and reorganise initialisation to set AirPlay 2 flags accordingly.  Also, remove redundant airplay_pin.
9. Remove redundant debug facility definitions from common.h
10. Make it possible to omit all the dacp stuff from a build.
11. Make it possible to omit metadata multicast from a build.
12. Add AP2-native metadata by setting bit 50 of features to true.
13. Improve stability when pre-empting an existing play session.
14. Fix a few compiler warnings in the pair_ap library.
15. Ensure that the audio format is known before starting synchronisation.
16. Improve handle_configure to set/clear a new config setting. Doesn’t seem to do anything…
17. Experimentally turn off bit 46 of AirPlay 2 features — seems to stop failing pair-verify attempts.
This commit is contained in:
Mike Brady
2026-04-26 10:23:18 +01:00
parent e2bb272d93
commit 11052b7896
56 changed files with 2672 additions and 1947 deletions
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ap2_buffered_audio_processor.c
+33 -8
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@@ -235,6 +235,7 @@ void *rtp_buffered_audio_processor(void *arg) {
uint16_t data_len;
// here we read from the buffer that our thread has been reading
// debug(1,"read a block");
size_t bytes_remaining_in_buffer;
nread =
read_sized_block(buffered_audio, &data_len, sizeof(data_len), &bytes_remaining_in_buffer);
@@ -249,6 +250,7 @@ void *rtp_buffered_audio_processor(void *arg) {
// get the block itself
// debug(1,"buffered audio packet of size %u detected.", data_len - 2);
nread = read_sized_block(buffered_audio, packet, data_len - 2, &bytes_remaining_in_buffer);
// debug(1,"block read");
// diagnostic
if ((conn->ap2_audio_buffer_minimum_size < 0) ||
@@ -442,7 +444,7 @@ void *rtp_buffered_audio_processor(void *arg) {
// to decode it and pass it to the player
if (new_audio_block_needed == 0) {
// is there space in the player thread's buffer system?
size_t player_buffer_occupancy = get_audio_buffer_occupancy(conn);
// size_t player_buffer_occupancy = get_audio_buffer_occupancy(conn);
// debug(1,"player buffer size and occupancy: %u and %u", player_buffer_size,
// player_buffer_occupancy);
@@ -451,17 +453,41 @@ void *rtp_buffered_audio_processor(void *arg) {
// and send it to the player. Otherwise, keep the block and sleep for a while.
// calculate if there is room in the decoded audio buffer...
int audio_decoded_buffer_below_desired_length = ((1.0 * player_buffer_occupancy * conn->frames_per_packet) / conn->input_rate) <= config.audio_decoded_buffer_desired_length;
// debug(1, "frames buffered: %f seconds, desired length: %f seconds.", (1.0 * player_buffer_occupancy * conn->frames_per_packet) / conn->input_rate, config.audio_decoded_buffer_desired_length);
// int audio_decoded_buffer_below_desired_length = ((1.0 * player_buffer_occupancy * conn->frames_per_packet) / conn->input_rate) <= config.audio_decoded_buffer_desired_length;
uint64_t buffer_should_be_time;
int have_valid_time = (frame_to_local_time(timestamp, &buffer_should_be_time, conn) == 0);
// calculate the lead time to make sure it's not too early...
int64_t lead_time = buffer_should_be_time - get_absolute_time_in_ns();
if ((play_enabled != 0) && (have_valid_time != 0) &&
(audio_decoded_buffer_below_desired_length != 0) &&
(lead_time * 1E-9 < (config.audio_decoded_buffer_desired_length + 0.1))) {
// debug(1,"play_enabled: %d, have_valid_time: %d, audio_decoded_buffer_below_desired_length: %d, lead_time * 1E-9: %f, (config.audio_decoded_buffer_desired_length + 0.1): %f, player_buffer_occupancy: %zu",
// play_enabled, have_valid_time, audio_decoded_buffer_below_desired_length, lead_time * 1E-9, (config.audio_decoded_buffer_desired_length + 0.1), player_buffer_occupancy
// );
// A slight problem here is that counting the number of buffers may not be sufficient,
// because the actual device may be
// taking data in large quantities at a single time.
// So we just have to ensure that there
// is enough of a lead time maintained for sufficient audio to be available to prevent
// the device from under-running.
// If means that the Shairport Sync player might riun out of audio occasionally, but
// as long as the device has enough in its buffer, everything is fine.
// But it also means that Shairport Sync's buffers must be sufficient to hold all the
// entire lead-time's amount of audio in case the device has a zero-sized buffer.
if ((play_enabled != 0) && (have_valid_time != 0)
// (audio_decoded_buffer_below_desired_length != 0) &&
&& (lead_time * 1E-9 < (config.audio_decoded_buffer_desired_length + 0.1))
// && (audio_decoded_buffer_below_desired_length != 0)
) {
very_early_packets_signalled = 0; //reset very early packet warning signaller
@@ -639,8 +665,7 @@ void *rtp_buffered_audio_processor(void *arg) {
debug(1, "incoming frame suddenly (?) has a lead time of %f seconds, with a desired decoded buffer length of %f.", 1.0 * lead_time * 1E-9, config.audio_decoded_buffer_desired_length);
very_early_packets_signalled = 1;
}
usleep(20000); // wait for a while
usleep(((1000000 * conn->frames_per_packet) / conn->input_rate) * 2); // wait for approximately the length of two packets
}
}
}