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Version 5.0 Major Release.

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New Features:
Multi-Channel and High-Resolution Audio Support
48,000 frames per second ("48k") operation.
48k lossless stereo support.
5.1 and 7.1 surround sound support.
Multi-channel and multi-rate operation on ALSA, PipeWire, PulseAudio, FreeBSD, stdout and Unix pipe output backends.
Automatic Audio Format Selection
Flexible and controllable output format selection.
Automatic rate, sample format, and channel count selection.
Full FFmpeg Integration
Support for transcoding.
Advanced resampling capabilities.
New audio format support.
Enhanced Resampling
New vernier resampling and interpolation method optimized for low-power CPUs.
Better performance on resource-constrained devices.

Convolution and Loudness Enhancements:
Convolution system is now multithreaded and works on stereo and multichannel audio at 48k and 44.1k.
Multiple impulse response (IR) files can now be provided via convolution_ir_files setting.
New convolution_thread_pool_size setting for multithreaded processing (defaults to 1).
Loudness processing now works with stereo and multichannel audio at 48k and 44.1k.
Updated to the most recent HiFi-LoFi FFT convolver.

MQTT Enhancements:
Added new publish_retain boolean option. When enabled, published MQTT messages have the retain flag set, so the MQTT broker stores the last message per topic and new subscribers receive the most recent value immediately. Thanks to lululombard for PR #2142.

D-Bus Enhancements:
Added new dbus_default_message_bus command-line argument (can be system or session) to set the default message bus for both D-Bus native service and MPRIS service.

Performance Improvements:
Enhanced compatibility with AirPlay 2 AutoMix and Smart Tracklists resulting in less unexplained track skipping.
Better operation on low-power devices down to Raspberry Pi B.
Improved efficiency on embedded systems.
Enhanced timestamp handling for better synchronization.
Improved sync error calculation.
Rebuilt buffered audio processor for cleaner handling of immediate and deferred flush requests.

Docker Enhancements:
Reduced Docker image sizes with slimmed-down FFmpeg library.
Removed dhclient from Docker images for smaller footprint.

Bug Fixes:
Fixed MQTT warning on service startup: "Could not establish a mqtt connection". The startup script now correctly states that the mosquitto service is required. Thanks to Hugo Villeneuve for PR #2137.
Fixed compatibility with mbedtls library version 3.4+ (present on recent Linux versions). Thanks to Christian Beier for finding and fixing the bug.
Fixed PulseAudio backend so that PA_ERR_NODATA returns "No latency data yet". Thanks to Vladimir Shakov for the report and fix.
Ensured old flush requests are deleted when a new play session starts. Thanks to saujanyashah for the report.
Fixed format warnings on 64-bit and 32-bit systems
Removed compilation warnings on 32-bit builds
Improved argument checking for debug(), inform(), warn() and die() functions
Fixed "daemon" typos throughout codebase. Thanks to Chris Boot for PR #1981.
Added warning if a convolution impulse response file cannot be read due to bad path or permissions

Build System Improvements:
Unified service file with variable substitution for Avahi support, making it easier to add future service dependencies. Thanks to Hugo Villeneuve.
Network interface selection now only considers interfaces that are up, running and not loopback interfaces. Thanks to Carl Johnson for the suggestion.
Configuration File Changes and Deprecations

New settings: convolution_ir_files (replaces convolution_ir_file), convolution_enabled (replaces convolution), convolution_max_length_in_seconds (replaces convolution_max_length), loudness_enabled (replaces loudness).
New convolution_thread_pool_size setting (defaults to 1).
Deprecated settings: convolution_ir_file, convolution, convolution_max_length, loudness.
Corresponding D-Bus methods and properties have been updated.

Deprecation Notice:
The Jack Audio and soundio backends are deprecated and will be removed in a future release. Consider using the updated PipeWire backend instead.

Documentation Updates
Updated BUILD.md with latest build instructions.
Updated AIRPLAY2.md with feature information.
Enhanced convolution and loudness documentation.

Maintenance:
Fixed FFmpeg deprecation warnings.
Bumped actions/checkout from 6.0.1 to 6.0.2.
Bumped docker/login-action from 3.6.0 to 3.7.0.
Bumped docker/build-push-action from 6.13.0 to 6.15.0.
Bumped docker/setup-qemu-action from 3.4.0 to 3.6.0.
Bumped docker/setup-buildx-action from 3.9.0 to 3.10.0.
This commit is contained in:
Mike Brady
2026-02-13 15:17:40 +00:00
parent 3c8ceb7c97
commit b405c0b896
169 changed files with 17206 additions and 10625 deletions
Download Patch File Download Diff File Expand all files Collapse all files
audio_ao.c
+176 -35
View File
@@ -1,7 +1,7 @@
/*
* libao output driver. This file is part of Shairport.
* Copyright (c) James Laird 2013
* Copyright (c) Mike Brady 2014 -- 2022
* Copyright (c) Mike Brady 2014--2025
* All rights reserved.
*
* Permission is hereby granted, free of charge, to any person
@@ -32,10 +32,161 @@
#include <stdio.h>
#include <unistd.h>
typedef struct {
sps_format_t sps_format;
int bits_per_sample;
int byte_format;
} sps_ao_t;
// these are the only formats that audio_pw will ever allow itself to be configured with
static sps_ao_t format_lookup[] = {{SPS_FORMAT_S16_LE, 16, AO_FMT_LITTLE},
{SPS_FORMAT_S16_BE, 16, AO_FMT_BIG},
{SPS_FORMAT_S32_LE, 32, AO_FMT_LITTLE},
{SPS_FORMAT_S32_BE, 32, AO_FMT_BIG}};
// use an SPS_FORMAT_... to find an entry in the format_lookup table or return NULL
static sps_ao_t *sps_format_lookup(sps_format_t to_find) {
sps_ao_t *response = NULL;
unsigned int i = 0;
while ((response == NULL) && (i < sizeof(format_lookup) / sizeof(sps_ao_t))) {
if (format_lookup[i].sps_format == to_find)
response = &format_lookup[i];
else
i++;
}
return response;
}
ao_device *dev = NULL;
ao_option *ao_opts = NULL;
ao_sample_format fmt;
ao_sample_format ao_output_format;
int driver = 0;
static int32_t current_encoded_output_format = 0;
static int check_settings(sps_format_t sample_format, unsigned int sample_rate,
unsigned int channel_count) {
// debug(1, "ao check_settings: configuration: %u/%s/%u.", sample_rate,
// sps_format_description_string(sample_format), channel_count);
int response = EINVAL;
sps_ao_t *format_info = sps_format_lookup(sample_format);
if (format_info != NULL) {
ao_sample_format check_fmt;
memset(&check_fmt, 0, sizeof(check_fmt));
check_fmt.bits = format_info->bits_per_sample;
check_fmt.rate = sample_rate;
check_fmt.channels = channel_count;
check_fmt.byte_format = format_info->byte_format;
// fmt.matrix = strdup("L,R");
ao_device *check_dev = ao_open_live(driver, &check_fmt, ao_opts);
if (check_dev != NULL) {
debug(3, "ao: check_settings %u/%s/%u works!", sample_rate,
sps_format_description_string(sample_format), channel_count);
if (ao_close(check_dev) == 0)
debug(1, "ao check_settings: error at ao_close");
response = 0;
} else {
debug(3, "ao: check_settings %u/%s/%u is not available!", sample_rate,
sps_format_description_string(sample_format), channel_count);
}
// response = 0;
}
// debug(1, "pa check_settings: configuration: %u/%s/%u %s.", sample_rate,
// sps_format_description_string(sample_format), channel_count,
// response == 0 ? "is okay" : "can not be configured");
return response;
}
static int check_configuration(unsigned int channels, unsigned int rate, unsigned int format) {
return check_settings(format, rate, channels);
}
static int32_t get_configuration(unsigned int channels, unsigned int rate, unsigned int format) {
uint64_t start_time = get_absolute_time_in_ns();
int32_t response =
search_for_suitable_configuration(channels, rate, format, &check_configuration);
int64_t elapsed_time = get_absolute_time_in_ns() - start_time;
debug(1, "ao: get_configuration took %0.3f mS.", elapsed_time * 0.000001);
return response;
}
static int configure(int32_t requested_encoded_format, char **channel_map) {
debug(3, "a0: configure %s.", short_format_description(requested_encoded_format));
int response = EINVAL;
if (current_encoded_output_format != requested_encoded_format) {
uint64_t start_time = get_absolute_time_in_ns();
if (current_encoded_output_format == 0)
debug(1, "a0: setting output configuration to %s.",
short_format_description(requested_encoded_format));
else
// note -- can't use short_format_description twice in one call because it returns the same
// string buffer each time
debug(1, "a0: changing output configuration to %s.",
short_format_description(requested_encoded_format));
current_encoded_output_format = requested_encoded_format;
sps_ao_t *format_info =
sps_format_lookup(FORMAT_FROM_ENCODED_FORMAT(current_encoded_output_format));
if (format_info == NULL)
die("ao: can't find format information!");
if (dev != NULL) {
if (ao_close(dev) == 0) {
debug(1, "ao configure: error closing existing device ao_close");
}
}
dev = NULL;
memset(&ao_output_format, 0, sizeof(ao_output_format));
ao_output_format.bits = format_info->bits_per_sample;
ao_output_format.rate = RATE_FROM_ENCODED_FORMAT(current_encoded_output_format);
ao_output_format.channels = CHANNELS_FROM_ENCODED_FORMAT(current_encoded_output_format);
ao_output_format.byte_format = format_info->byte_format;
switch (CHANNELS_FROM_ENCODED_FORMAT(current_encoded_output_format)) {
case 2:
ao_output_format.matrix = strdup("L,R");
break;
case 6:
ao_output_format.matrix = strdup("L,R,C,LFE,BR,BL");
break;
case 8:
ao_output_format.matrix = strdup("L,R,C,LFE,BR,BL,SL,SR");
break;
default:
break;
}
// fmt.matrix = strdup("L,R");
/*
dev = ao_open_live(driver, &ao_output_format, ao_opts);
if (dev != NULL) {
debug(
1, "ao: configure %u/%s/%u opened!",
RATE_FROM_ENCODED_FORMAT(current_encoded_output_format),
sps_format_description_string(FORMAT_FROM_ENCODED_FORMAT(current_encoded_output_format)),
CHANNELS_FROM_ENCODED_FORMAT(current_encoded_output_format));
response = 0;
} else {
debug(
1, "ao: configure %u/%s/%u is not available!",
RATE_FROM_ENCODED_FORMAT(current_encoded_output_format),
sps_format_description_string(FORMAT_FROM_ENCODED_FORMAT(current_encoded_output_format)),
CHANNELS_FROM_ENCODED_FORMAT(current_encoded_output_format));
}
*/
int64_t elapsed_time = get_absolute_time_in_ns() - start_time;
debug(3, "pa: configure took %0.3f mS.", elapsed_time * 0.000001);
} else {
debug(3, "pa: setting output configuration -- configuration unchanged, so nothing done.");
}
if ((response == 0) && (channel_map != NULL)) {
*channel_map = NULL; // nothing back here
}
return response;
}
static void help(void) {
printf(" -d driver set the output driver\n"
@@ -69,12 +220,10 @@ static void help(void) {
}
static int init(int argc, char **argv) {
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
ao_initialize();
driver = ao_default_driver_id();
if (driver == -1) {
warn("libao can not find a usable driver!");
warn("the ao backend can not find a usable output device with the default driver!");
} else {
// set up default values first
@@ -85,8 +234,14 @@ static int init(int argc, char **argv) {
// get settings from settings file first, allow them to be overridden by
// command line options
// do the "general" audio options. Note, these options are in the "general" stanza!
parse_general_audio_options();
// get settings from settings file, passing in defaults for format_set, rate_set and channel_set
// Note, these options may be in the "general" stanza or the named stanza
#ifdef CONFIG_FFMPEG
parse_audio_options("ao", SPS_FORMAT_SET, SPS_RATE_SET, SPS_CHANNEL_SET);
#else
parse_audio_options("ao", SPS_FORMAT_NON_FFMPEG_SET, SPS_RATE_NON_FFMPEG_SET,
SPS_CHANNNEL_NON_FFMPEG_SET);
#endif
optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour
argv--; // so we shift the arguments to satisfy getopt()
@@ -127,66 +282,52 @@ static int init(int argc, char **argv) {
if (optind < argc)
die("Invalid audio argument: %s", argv[optind]);
memset(&fmt, 0, sizeof(fmt));
fmt.bits = 16;
fmt.rate = 44100;
fmt.channels = 2;
fmt.byte_format = AO_FMT_NATIVE;
fmt.matrix = strdup("L,R");
// get_permissible_configuration_settings();
}
pthread_setcancelstate(oldState, NULL);
return 0;
}
static void deinit(void) {
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
if (dev != NULL)
ao_close(dev);
dev = NULL;
ao_shutdown();
pthread_setcancelstate(oldState, NULL);
}
static void start(__attribute__((unused)) int sample_rate,
__attribute__((unused)) int sample_format) {
// debug(1,"libao start");
}
static int play(void *buf, int samples, __attribute__((unused)) int sample_type,
__attribute__((unused)) uint32_t timestamp,
__attribute__((unused)) uint64_t playtime) {
int response = 0;
if (dev == NULL) {
debug(1, "ao play(): ao_open_live to play %d samples, bytes per sample: %u, channels: %u",
samples, ao_output_format.bits / 8, ao_output_format.channels);
dev = ao_open_live(driver, &ao_output_format, ao_opts);
}
// debug(1,"ao play(): play %d samples, bytes per sample: %u, channels: %u", samples,
// ao_output_format.bits/8, ao_output_format.channels);
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
if (driver != -1) {
if (dev == NULL)
dev = ao_open_live(driver, &fmt, ao_opts);
if (dev != NULL)
response = ao_play(dev, buf, samples * 4);
}
response = ao_play(dev, buf, samples * ao_output_format.bits / 8 * ao_output_format.channels);
pthread_setcancelstate(oldState, NULL);
return response;
}
static void stop(void) {
// debug(1,"libao stop");
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
if (dev != NULL) {
debug(1, "ao stop(): ao_close");
ao_close(dev);
dev = NULL;
}
pthread_setcancelstate(oldState, NULL);
}
audio_output audio_ao = {.name = "ao",
.help = &help,
.init = &init,
.deinit = &deinit,
.prepare = NULL,
.start = &start,
.configure = &configure,
.get_configuration = &get_configuration,
// .start = &start,
.stop = &stop,
.is_running = NULL,
.flush = NULL,