nils/shairport-sync
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Enhancements

Browse Source 
Enable the AirPlay 2 build to operate an optional classic-AirPlay-only service or to gracefully degrade to classic AirPlay, as follows:
  1. Add a new command-line option '-—service-type=<type>' and an equivalent configuration entry 'service_type = "<type>"' in the 'general' section of the configuration file,
     where <type> can be "auto", "classic" or "airplay2":
      1. "auto" (default) means that the service will be AirPlay 2 if NQPTP is running. If NQPTP is not running, classic AirPlay service will be provided instead, and in that case,
      "(Classic)" will be appended to the default AirPlay service name visible to AirPlay clients like Apple Music, for example "RaspberryPi3B (Classic)".
      2. "classic" means the service will be classic AirPlay (aka AirPlay 1).
      3. "airplay2" means the service will be the modern AirPlay 2. In this case, as distinct from "auto", if NQPTP is not running, Shairport Sync will log an error and terminate.
  2. In the systemd service file, NQPTP is now a "Want" rather than a "Require". If it's present, then it will be launched before Shairport Sync. If it's absent, Shairport Sync will launch anyway.
  3. Improve the delivery of input format changes and emit 'sdsc' metadata when changes occur.
  4. Emit new format information in the log if statistics is enabled.

Docker Changes
  1. Support for linux/arm/v6 has been dropped, as Docker is no longer supported.
  2. NQPTP is not started in the AirPlay 2 Docker image if '--service-type=classic' or '--service-type=airplay1' is in the command line options at the end of the docker run command.
     The purpose is to ensure that ports 319 and 320 are left alone when the AirPlay 2 image is set to provide Classic service only.
     Note that setting the configuration file 'service_type' to 'classic' will not prevent NQPTP from starting up -- you must use the command line option.
  3. A new "dev" target has been added. It is a large image containing the custom-built FFmpeg library, NQPTP, Avahi and D-Bus along with the Shairport Sync source and
     all necessary development tools. When started, Avahi, D-Bus and NQPTP are all installed and running. The bash shell has also been added and is entered.

Stability Improvements
  Reorganise session preemption to fully terminate the existing session before starting a new one.
  Don't delay closing the event port to wait for it to be closed at the client end.
  Add a safe_socket_close() function to ensure sockets are fully closed. Use -1 to designate closed rather than 0, to prevent attempts to reclose sockets, causing mayhem.
  Re-order FFmpeg decommissioning during teardown.
This commit is contained in:
Mike Brady
2026-05-18 09:57:12 +01:00
parent 0bd9271ea9
commit f30dd7aee6
69 changed files with 2897 additions and 2716 deletions
Download Patch File Download Diff File Expand all files Collapse all files
ap2_buffered_audio_processor.c
+71 -52
View File
@@ -30,6 +30,7 @@
#include "rtp.h"
#include "utilities/buffered_read.h"
#include "utilities/mod23.h"
#include "utilities/network_utilities.h"
#include <sodium.h>
#include <stdint.h>
@@ -91,10 +92,9 @@ void addADTStoPacket(uint8_t *packet, int packetLen, int rate, int channel_confi
void rtp_buffered_audio_cleanup_handler(__attribute__((unused)) void *arg) {
debug(2, "Buffered Audio Receiver Cleanup Start.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
close(conn->buffered_audio_socket);
safe_socket_close(&conn->buffered_audio_socket);
debug(3, "Connection %d: closing TCP Buffered Audio port: %u.", conn->connection_number,
conn->local_buffered_audio_port);
conn->buffered_audio_socket = 0;
debug(2, "Connection %d: rtp_buffered_audio_processor exit.", conn->connection_number);
}
@@ -198,7 +198,7 @@ void *rtp_buffered_audio_processor(void *arg) {
int packets_played_in_this_sequence = 0;
int play_enabled = 0;
int very_early_packets_signalled = 0;
// double requested_lead_time = 0.0; // normal lead time minimum -- maybe it should be about 0.1
@@ -276,20 +276,19 @@ void *rtp_buffered_audio_processor(void *arg) {
if (payload_ssrc != SSRC_NONE)
previous_ssrc = payload_ssrc;
payload_ssrc = nctohl(&packet[8]);
if ((payload_ssrc != previous_ssrc) && (payload_ssrc != SSRC_NONE)) {
if (ssrc_is_recognised(payload_ssrc) == 0) {
debug(2, "Unrecognised SSRC: %u.", payload_ssrc);
} else {
debug(2, "Connection %d: incoming audio encoding is%s \"%s\".",
conn->connection_number, previous_ssrc == SSRC_NONE ? "" : " switching to", get_ssrc_name(payload_ssrc));
conn->connection_number, previous_ssrc == SSRC_NONE ? "" : " switching to",
get_ssrc_name(payload_ssrc));
}
}
if ((payload_ssrc != previous_ssrc) && (ssrc_is_recognised(payload_ssrc) == 0)) {
debug(2, "Unrecognised SSRC: %u.", payload_ssrc);
debug(2, "Unrecognised SSRC: %u.", payload_ssrc);
}
if (blocks_read_since_play_began == 1) {
@@ -327,7 +326,8 @@ void *rtp_buffered_audio_processor(void *arg) {
finished = 1;
} else if (nread < 0) {
char errorstring[1024];
(void) !strerror_r(errno, (char *)errorstring, sizeof(errorstring)); // (void) ! to suppress unused response warning
(void)!strerror_r(errno, (char *)errorstring,
sizeof(errorstring)); // (void) ! to suppress unused response warning
debug(1, "error in rtp_buffered_audio_processor %d: \"%s\". Could not recv a data_len .",
errno, errorstring);
finished = 1;
@@ -343,34 +343,42 @@ void *rtp_buffered_audio_processor(void *arg) {
debug(2, "immediate flush started at sequence number %u until sequence number of %u.",
seq_no, conn->ap2_immediate_flush_until_sequence_number);
}
if ((blocks_read != 0) && ((a_minus_b_mod23(seq_no, conn->ap2_immediate_flush_until_sequence_number) > 0))) {
debug(1, "immediate flush may have escaped its endpoint! Seq_no is %u, conn->ap2_immediate_flush_until_sequence_number is %u.", seq_no, conn->ap2_immediate_flush_until_sequence_number);
if ((blocks_read != 0) &&
((a_minus_b_mod23(seq_no, conn->ap2_immediate_flush_until_sequence_number) > 0))) {
debug(1,
"immediate flush may have escaped its endpoint! Seq_no is %u, "
"conn->ap2_immediate_flush_until_sequence_number is %u.",
seq_no, conn->ap2_immediate_flush_until_sequence_number);
}
if ((blocks_read != 0) && ((a_minus_b_mod23(seq_no, conn->ap2_immediate_flush_until_sequence_number) >= 0))) {
debug(2, "immediate flush completed at seq_no: %u, conn->ap2_immediate_flush_until_sequence_number: %u.", seq_no, conn->ap2_immediate_flush_until_sequence_number);
if ((blocks_read != 0) &&
((a_minus_b_mod23(seq_no, conn->ap2_immediate_flush_until_sequence_number) >= 0))) {
debug(2,
"immediate flush completed at seq_no: %u, "
"conn->ap2_immediate_flush_until_sequence_number: %u.",
seq_no, conn->ap2_immediate_flush_until_sequence_number);
conn->ap2_immediate_flush_requested = 0;
ap2_immediate_flush_requested = 0;
// turn off all deferred requests. Not sure if this is right...
unsigned int f = 0;
for (f = 0; f < MAX_DEFERRED_FLUSH_REQUESTS; f++) {
if ((conn->ap2_deferred_flush_requests[f].inUse != 0) && (conn->ap2_deferred_flush_requests[f].active = 0)) {
if ((conn->ap2_deferred_flush_requests[f].inUse != 0) &&
(conn->ap2_deferred_flush_requests[f].active = 0)) {
debug(1,
"deferred flush cancelled by an immediate flush: flushFromTS: %12u, flushFromSeq: %12u, "
"flushUntilTS: %12u, flushUntilSeq: %12u, timestamp: %12u.",
conn->ap2_deferred_flush_requests[f].flushFromTS,
conn->ap2_deferred_flush_requests[f].flushFromSeq,
conn->ap2_deferred_flush_requests[f].flushUntilTS,
conn->ap2_deferred_flush_requests[f].flushUntilSeq, timestamp);
"deferred flush cancelled by an immediate flush: flushFromTS: %12u, "
"flushFromSeq: %12u, "
"flushUntilTS: %12u, flushUntilSeq: %12u, timestamp: %12u.",
conn->ap2_deferred_flush_requests[f].flushFromTS,
conn->ap2_deferred_flush_requests[f].flushFromSeq,
conn->ap2_deferred_flush_requests[f].flushUntilTS,
conn->ap2_deferred_flush_requests[f].flushUntilSeq, timestamp);
}
conn->ap2_deferred_flush_requests[f].inUse = 0;
conn->ap2_deferred_flush_requests[f].active = 0;
}
} else {
debug(4, "immediate flush of block %u until block %u", seq_no,
conn->ap2_immediate_flush_until_sequence_number);
@@ -426,11 +434,10 @@ void *rtp_buffered_audio_processor(void *arg) {
} else if (conn->ap2_deferred_flush_requests[f].active != 0) {
new_audio_block_needed = 1;
debug(4,
"deferred flush of block: %u, timestamp: %u, SSRC: \"%s\". flushFromTS: %12u, flushFromSeq: %12u, "
"deferred flush of block: %u, timestamp: %u, SSRC: \"%s\". flushFromTS: %12u, "
"flushFromSeq: %12u, "
"flushUntilTS: %12u, flushUntilSeq: %12u, timestamp: %12u.",
seq_no,
timestamp,
get_ssrc_name(payload_ssrc),
seq_no, timestamp, get_ssrc_name(payload_ssrc),
conn->ap2_deferred_flush_requests[f].flushFromTS,
conn->ap2_deferred_flush_requests[f].flushFromSeq,
conn->ap2_deferred_flush_requests[f].flushUntilTS,
@@ -453,22 +460,28 @@ 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...
// 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;
// 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();
// 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
// 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.
@@ -476,34 +489,35 @@ void *rtp_buffered_audio_processor(void *arg) {
// 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) &&
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
// && (audio_decoded_buffer_below_desired_length != 0)
) {
very_early_packets_signalled = 0; // reset very early packet warning signaller
// try to identify blocks that are timed to before the last buffer, and drop 'em
int64_t time_from_last_buffer_time =
buffer_should_be_time - previous_buffer_should_be_time;
if ((packets_played_in_this_sequence == 0) || (time_from_last_buffer_time > 0)) {
payload_length = 0;
if (ssrc_is_recognised(payload_ssrc) != 0) {
// prepare_decoding_chain(conn, payload_ssrc);
unsigned long long new_payload_length = 0;
payload_pointer = m + leading_free_space_length;
if (lead_time >= 0) { // only decipher the packet if it's not too late
int response = -1; // guess that there is a problem
int response = -1; // guess that there is a problem
if (conn->session_key != NULL) {
unsigned char nonce[12];
memset(nonce, 0, sizeof(nonce));
@@ -661,11 +675,16 @@ void *rtp_buffered_audio_processor(void *arg) {
}
new_audio_block_needed = 1; // the block has been used up and is no longer current
} else {
if ((have_valid_time != 0) && (very_early_packets_signalled == 0) && (lead_time * 1E-9 > (config.audio_decoded_buffer_desired_length + 0.2))) {
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);
if ((have_valid_time != 0) && (very_early_packets_signalled == 0) &&
(lead_time * 1E-9 > (config.audio_decoded_buffer_desired_length + 0.2))) {
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(((1000000 * conn->frames_per_packet) / conn->input_rate) * 2); // wait for approximately the length of two packets
usleep(((1000000 * conn->frames_per_packet) / conn->input_rate) *
2); // wait for approximately the length of two packets
}
}
}