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shairport-sync/audio_pw.c
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Mike Brady 60143518c2 This squashed commit comprises a long series of improvements, bug fixes, optimizations, and new feature additions: 
- 48,000 frames per second operation,
 - 48k lossless stereo, 5.1 and 7.1 surround sound
 - Multichannel and multi rate operation on ALSA, PipeWire, PulseAudio, FreeBSD, stdout and unix pipe output backends.

Automatic, flexible and controllable output format (rate, sample format and channel count) selection.

Full FFmpeg integration to support transcoding, resampling, and new audio formats.

Better operation on lower powered devices down to e.g. Raspberry Pi B.

Reduced Docker image sizes with a slimmed-down FFmpeg library.

Enhanced timestamp handling for better synchronization.

Improved the sync error calculation.

A new "vernier" resampling and interpolation method for low-power CPUs.

Bug fixes and minor enhancements.

Note -- there are many breaking changes from previous versions of Shairport Sync!
2025-02-23 13:31:17 +00:00

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/*
* Asynchronous PipeWire Backend. This file is part of Shairport Sync.
* Copyright (c) Mike Brady 2024--2025
* All rights reserved.
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or
* sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
// This uses ideas from the tone generator sample code at:
// https://github.com/PipeWire/pipewire/blob/master/src/examples/audio-src.c
// Thanks to Wim Taymans.
#include "audio.h"
#include "common.h"
#include <errno.h>
#include <pthread.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <pipewire/pipewire.h>
#include <spa/param/audio/format-utils.h>
static char channel_map_mono[] = "FC";
static char channel_map_stereo[] = "FL FR";
static char channel_map_2p1[] = "FL FR LFE";
static char channel_map_4p0[] = "FL FR FC BC";
static char channel_map_5p0[] = "FL FR FC BL BR";
static char channel_map_5p1[] = "FL FR FC LFE BL BR";
static char channel_map_6p1[] = "FL FR FC LFE BC SL SR";
static char channel_map_7p1[] = "FL FR FC LFE BL BR SL SR";
typedef struct {
enum spa_audio_format spa_format;
sps_format_t sps_format;
unsigned int bytes_per_sample;
} spa_sps_t;
// these are the only formats that audio_pw will ever allow itself to be configured with
static spa_sps_t format_lookup[] = {{SPA_AUDIO_FORMAT_S16_LE, SPS_FORMAT_S16_LE, 2},
{SPA_AUDIO_FORMAT_S16_BE, SPS_FORMAT_S16_BE, 2},
{SPA_AUDIO_FORMAT_S32_LE, SPS_FORMAT_S32_LE, 4},
{SPA_AUDIO_FORMAT_S32_BE, SPS_FORMAT_S32_BE, 4}};
#define BUFFER_SIZE_IN_SECONDS 1
static uint8_t buffer[1024];
static pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER;
static int32_t current_encoded_output_format = 0;
static char *audio_lmb, *audio_umb, *audio_toq, *audio_eoq;
static size_t audio_size = 0;
static size_t audio_occupancy;
static int enable_fill;
static int stream_is_active;
static int on_process_is_running = 0;
struct data {
struct pw_thread_loop *loop;
struct pw_stream *stream;
unsigned int rate;
unsigned int bytes_per_sample;
unsigned int channels;
};
// the pipewire global data structure
struct data data = {NULL, NULL, 0, 0, 0};
// use an SPS_FORMAT_... to find an entry in the format_lookup table or return NULL
static spa_sps_t *sps_format_lookup(sps_format_t to_find) {
spa_sps_t *response = NULL;
unsigned int i = 0;
while ((response == NULL) && (i < sizeof(format_lookup) / sizeof(spa_sps_t))) {
if (format_lookup[i].sps_format == to_find)
response = &format_lookup[i];
else
i++;
}
return response;
}
static void on_process(void *userdata) {
struct data *local_data = userdata;
int n_frames = 0;
pthread_mutex_lock(&buffer_mutex);
// debug(1, "on_process called.");
if (stream_is_active == 0)
debug(1, "on_process called while stream inactive!");
on_process_is_running = 1;
if ((audio_occupancy > 0) || (enable_fill)) {
// get a buffer to see how big it can be
struct pw_buffer *b = pw_stream_dequeue_buffer(local_data->stream);
if (b == NULL) {
pw_log_warn("out of buffers: %m");
die("PipeWire failure -- out of buffers!");
}
struct spa_buffer *buf = b->buffer;
uint8_t *dest = buf->datas[0].data;
if (dest != NULL) {
int stride = local_data->bytes_per_sample * local_data->channels;
// note: the requested field is the number of frames, not bytes, requested
int max_possible_frames = SPA_MIN(b->requested, buf->datas[0].maxsize / stride);
size_t bytes_we_can_transfer = max_possible_frames * stride;
if (audio_occupancy > 0) {
// if (enable_fill == 1)) {
// debug(1, "got audio -- disable_fill");
// }
enable_fill = 0;
if (bytes_we_can_transfer > audio_occupancy)
bytes_we_can_transfer = audio_occupancy;
n_frames = bytes_we_can_transfer / stride;
size_t bytes_to_end_of_buffer = (size_t)(audio_umb - audio_toq); // must be zero or positive
if (bytes_we_can_transfer <= bytes_to_end_of_buffer) {
// the bytes are all in a row in the audio buffer
memcpy(dest, audio_toq, bytes_we_can_transfer);
audio_toq += bytes_we_can_transfer;
} else {
// the bytes are in two places in the audio buffer
size_t first_portion_to_write = audio_umb - audio_toq;
if (first_portion_to_write != 0)
memcpy(dest, audio_toq, first_portion_to_write);
uint8_t *new_dest = dest + first_portion_to_write;
memcpy(new_dest, audio_lmb, bytes_we_can_transfer - first_portion_to_write);
audio_toq = audio_lmb + bytes_we_can_transfer - first_portion_to_write;
}
audio_occupancy -= bytes_we_can_transfer;
} else {
debug(3, "send silence");
// this should really be dithered silence
memset(dest, 0, bytes_we_can_transfer);
n_frames = max_possible_frames;
}
buf->datas[0].chunk->offset = 0;
buf->datas[0].chunk->stride = stride;
buf->datas[0].chunk->size = n_frames * stride;
pw_stream_queue_buffer(local_data->stream, b);
} // (else the first data block does not contain a data pointer)
}
pthread_mutex_unlock(&buffer_mutex);
}
static const struct pw_stream_events stream_events = {PW_VERSION_STREAM_EVENTS,
.process = on_process};
static void deinit(void) {
pw_thread_loop_stop(data.loop);
if (data.stream != NULL)
pw_stream_destroy(data.stream);
pw_thread_loop_destroy(data.loop);
pw_deinit();
on_process_is_running = 0;
if (audio_lmb != NULL)
free(audio_lmb); // deallocate that buffer
}
static int init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) {
// set up default values first
config.audio_backend_buffer_desired_length = 0.5;
config.audio_backend_buffer_interpolation_threshold_in_seconds =
0.02; // below this, soxr interpolation will not occur -- it'll be basic interpolation
// instead.
config.audio_backend_latency_offset = 0;
// 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("pipewire", SPS_FORMAT_SET, SPS_RATE_SET, SPS_CHANNEL_SET);
#else
parse_audio_options("pipewire", SPS_FORMAT_NON_FFMPEG_SET, SPS_RATE_NON_FFMPEG_SET,
SPS_CHANNNEL_NON_FFMPEG_SET);
#endif
// now any PipeWire-specific options
if (config.cfg != NULL) {
const char *str;
// Get the optional Application Name, if provided.
if (config_lookup_string(config.cfg, "pipewire.application_name", &str)) {
config.pw_application_name = (char *)str;
}
// Get the optional PipeWire node name, if provided.
if (config_lookup_string(config.cfg, "pipewire.node_name", &str)) {
config.pw_node_name = (char *)str;
}
// Get the optional PipeWire sink target name, if provided.
if (config_lookup_string(config.cfg, "pipewire.sink_target", &str)) {
config.pw_sink_target = (char *)str;
}
}
// finished collecting settings
audio_lmb = NULL;
audio_size = 0;
current_encoded_output_format = 0;
enable_fill = 1;
int largc = 0;
pw_init(&largc, NULL);
/* make a threaded loop. */
data.loop = pw_thread_loop_new("shairport-sync", NULL);
pw_thread_loop_lock(data.loop);
char *appname = config.pw_application_name;
if (appname == NULL)
appname = "Shairport Sync";
char *nodename = config.pw_node_name;
if (nodename == NULL)
nodename = "Shairport Sync";
struct pw_properties *props = pw_properties_new(
PW_KEY_MEDIA_TYPE, "Audio", PW_KEY_MEDIA_CATEGORY, "Playback", PW_KEY_MEDIA_ROLE, "Music",
PW_KEY_APP_NAME, appname, PW_KEY_NODE_NAME, nodename, NULL);
if (config.pw_sink_target != NULL) {
debug(3, "setting sink target to \"%s\".", config.pw_sink_target);
pw_properties_set(props, PW_KEY_TARGET_OBJECT, config.pw_sink_target);
}
data.stream = pw_stream_new_simple(pw_thread_loop_get_loop(data.loop), config.appName, props,
&stream_events, &data);
pw_thread_loop_start(data.loop);
on_process_is_running = 0;
pw_thread_loop_unlock(data.loop);
return 0;
}
static int check_settings(sps_format_t sample_format, unsigned int sample_rate,
unsigned int channel_count) {
// we know the formats with be big- or little-ended.
// we will accept only S32_..., S16_...
int response = EINVAL;
if (sps_format_lookup(sample_format) != NULL)
response = 0;
debug(3, "pw: 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) {
return search_for_suitable_configuration(channels, rate, format, &check_configuration);
}
static int configure(int32_t requested_encoded_format, char **resulting_channel_map) {
// debug(2, "pw: configure %s.", short_format_description(requested_encoded_format));
int response = 0;
char *channel_map = NULL;
// if (1) {
if (current_encoded_output_format != requested_encoded_format) {
uint64_t start_time = get_absolute_time_in_ns();
if (current_encoded_output_format == 0)
debug(2, "pw: 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(2, "pw: changing output configuration to %s.",
short_format_description(requested_encoded_format));
current_encoded_output_format = requested_encoded_format;
spa_sps_t *format_info =
sps_format_lookup(FORMAT_FROM_ENCODED_FORMAT(current_encoded_output_format));
if (format_info == NULL)
die("Can't find format information!");
// enum spa_audio_format spa_format = format_info->spa_format;
data.bytes_per_sample = format_info->bytes_per_sample;
data.channels = CHANNELS_FROM_ENCODED_FORMAT(current_encoded_output_format);
data.rate = RATE_FROM_ENCODED_FORMAT(current_encoded_output_format);
pw_thread_loop_lock(data.loop);
enable_fill = 0;
if (pw_stream_get_state(data.stream, NULL) != PW_STREAM_STATE_UNCONNECTED) {
response = pw_stream_disconnect(data.stream);
if (response != 0) {
debug(1, "error %d disconnecting stream.", response);
}
}
if (audio_lmb != NULL) {
// debug(3, "deallocating existing audio_pw.c buffer.");
free(audio_lmb);
}
audio_size = data.rate * BUFFER_SIZE_IN_SECONDS * data.bytes_per_sample * data.channels;
// allocate space for the audio buffer
audio_lmb = malloc(audio_size);
if (audio_lmb == NULL)
die("Can't allocate %d bytes for PipeWire buffer.", audio_size);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + audio_size;
audio_occupancy = 0;
// Make one parameter with the supported formats. The SPA_PARAM_EnumFormat
// id means that this is a format enumeration (of 1 value).
struct spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
const struct spa_pod *params[1];
// create a stream with the default channel layout corresponding to
// the number of channels
switch (CHANNELS_FROM_ENCODED_FORMAT(current_encoded_output_format)) {
case 1:
channel_map = channel_map_mono;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
// we are giving the position of 8 channels here, even if we need less than that.
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate, .position = {SPA_AUDIO_CHANNEL_FC}));
break;
case 2:
channel_map = channel_map_stereo;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
// we are giving the position of 8 channels here, even if we need less than that.
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR}));
break;
case 3:
channel_map = channel_map_2p1;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
// we are giving the position of 8 channels here, even if we need less than that.
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_LFE}));
break;
case 4:
channel_map = channel_map_4p0;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
// we are giving the position of 8 channels here, even if we need less than that.
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_FC, SPA_AUDIO_CHANNEL_BC}));
break;
case 5:
channel_map = channel_map_5p0;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
// we are giving the position of 8 channels here, even if we need less than that.
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_FC, SPA_AUDIO_CHANNEL_RL,
SPA_AUDIO_CHANNEL_RR}));
break;
case 6:
channel_map = channel_map_5p1;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_FC, SPA_AUDIO_CHANNEL_LFE,
SPA_AUDIO_CHANNEL_RL, SPA_AUDIO_CHANNEL_RR}));
break;
case 7:
channel_map = channel_map_6p1;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_FC, SPA_AUDIO_CHANNEL_LFE,
SPA_AUDIO_CHANNEL_BC, SPA_AUDIO_CHANNEL_SL,
SPA_AUDIO_CHANNEL_SR}));
break;
case 8:
channel_map = channel_map_7p1;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_FC, SPA_AUDIO_CHANNEL_LFE,
SPA_AUDIO_CHANNEL_RL, SPA_AUDIO_CHANNEL_RR,
SPA_AUDIO_CHANNEL_SL, SPA_AUDIO_CHANNEL_SR}));
break;
default:
channel_map = NULL;
params[0] = spa_format_audio_raw_build(
&b, SPA_PARAM_EnumFormat,
// we are giving the position of 8 channels here, even if we need less than that.
&SPA_AUDIO_INFO_RAW_INIT(.format = format_info->spa_format, .channels = data.channels,
.rate = data.rate,
.position = {SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR,
SPA_AUDIO_CHANNEL_FC, SPA_AUDIO_CHANNEL_LFE,
SPA_AUDIO_CHANNEL_RL, SPA_AUDIO_CHANNEL_RR,
SPA_AUDIO_CHANNEL_SL, SPA_AUDIO_CHANNEL_SR}));
break;
}
// Now connect this stream. We ask that our process function is
// called in a realtime thread.
pw_stream_connect(data.stream, PW_DIRECTION_OUTPUT, PW_ID_ANY,
PW_STREAM_FLAG_AUTOCONNECT | PW_STREAM_FLAG_MAP_BUFFERS |
PW_STREAM_FLAG_RT_PROCESS,
params, 1);
stream_is_active = 0;
enable_fill = 1;
pw_thread_loop_unlock(data.loop);
int64_t elapsed_time = get_absolute_time_in_ns() - start_time;
debug(3, "pw: configuration took %0.3f mS.", elapsed_time * 0.000001);
} else {
debug(2, "pw: setting output configuration -- configuration unchanged, so nothing done.");
}
if ((response == 0) && (resulting_channel_map != NULL)) {
*resulting_channel_map = channel_map;
}
return response;
}
static int play(__attribute__((unused)) void *buf, int samples,
__attribute__((unused)) int sample_type, __attribute__((unused)) uint32_t timestamp,
__attribute__((unused)) uint64_t playtime) {
if (stream_is_active == 0) {
pw_thread_loop_lock(data.loop);
on_process_is_running = 0;
pw_stream_set_active(data.stream, true);
pw_thread_loop_unlock(data.loop);
stream_is_active = 1;
// debug(1, "set stream active");
}
// copy the samples into the queue
// debug(3, "play %u samples; %u samples already in the buffer.", samples, audio_occupancy /
// (data.bytes_per_sample * data.channels));
size_t bytes_to_transfer = samples * data.channels * data.bytes_per_sample;
pthread_mutex_lock(&buffer_mutex);
size_t bytes_available = audio_size - audio_occupancy;
if (bytes_available < bytes_to_transfer)
bytes_to_transfer = bytes_available;
if (bytes_to_transfer > 0) {
size_t space_to_end_of_buffer = audio_umb - audio_eoq;
if (space_to_end_of_buffer >= bytes_to_transfer) {
memcpy(audio_eoq, buf, bytes_to_transfer);
audio_eoq += bytes_to_transfer;
} else {
memcpy(audio_eoq, buf, space_to_end_of_buffer);
buf += space_to_end_of_buffer;
memcpy(audio_lmb, buf, bytes_to_transfer - space_to_end_of_buffer);
audio_eoq = audio_lmb + bytes_to_transfer - space_to_end_of_buffer;
}
audio_occupancy += bytes_to_transfer;
}
pthread_mutex_unlock(&buffer_mutex);
return 0;
}
static int delay(long *the_delay) {
long result = 0;
int reply = -ENODEV; // ENODATA is not defined in FreeBSD
if (on_process_is_running == 0) {
debug(3, "pw_processor not running");
}
if (stream_is_active == 0) {
debug(1, "stream not active but on_process_is_running is 1!");
}
if (on_process_is_running != 0) {
struct pw_time stream_time_info_1, stream_time_info_2;
ssize_t audio_occupancy_now;
// get stable pw_time info to ensure we get an audio occupancy figure
// that relates to the pw_time we have.
// we do this by getting a pw_time before and after getting the occupancy
// and accepting the information if they are both the same
int loop_count = 1;
int non_matching;
int stream_time_valid_if_zero;
do {
stream_time_valid_if_zero =
pw_stream_get_time_n(data.stream, &stream_time_info_1, sizeof(struct pw_time));
audio_occupancy_now = audio_occupancy;
pw_stream_get_time_n(data.stream, &stream_time_info_2, sizeof(struct pw_time));
non_matching = memcmp(&stream_time_info_1, &stream_time_info_2, sizeof(struct pw_time));
if (non_matching != 0) {
loop_count++;
}
} while (((non_matching != 0) || (stream_time_valid_if_zero != 0)) && (loop_count < 10));
if (non_matching != 0) {
debug(1, "can't get a stable pw_time!");
}
if (stream_time_valid_if_zero != 0) {
debug(1, "can't get valid stream info");
}
if (stream_time_info_1.rate.denom == 0) {
debug(2, "non valid stream_time_info_1");
}
if ((non_matching == 0) && (stream_time_valid_if_zero == 0) &&
(stream_time_info_1.rate.denom != 0)) {
int64_t interval_from_pw_time_to_now_ns =
pw_stream_get_nsec(data.stream) - stream_time_info_1.now;
uint64_t frames_possibly_played_since_measurement =
((interval_from_pw_time_to_now_ns * data.rate) + 500000000L) / 1000000000L;
uint64_t net_delay_in_frames = stream_time_info_1.queued + stream_time_info_1.buffered;
uint64_t fixed_delay_ns =
(stream_time_info_1.delay * stream_time_info_1.rate.num * 1000000000L) /
stream_time_info_1.rate.denom; // ns;
uint64_t fixed_delay_in_frames = ((fixed_delay_ns * data.rate) + 500000000L) / 1000000000L;
net_delay_in_frames = net_delay_in_frames + fixed_delay_in_frames +
audio_occupancy_now / (data.bytes_per_sample * data.channels) -
frames_possibly_played_since_measurement;
result = net_delay_in_frames;
reply = 0;
}
}
*the_delay = result;
return reply;
}
static void flush(void) {
pthread_mutex_lock(&buffer_mutex);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + audio_size;
audio_occupancy = 0;
// if (enable_fill == 0) {
// debug(1, "flush enable_fill");
// }
enable_fill = 1;
pthread_mutex_unlock(&buffer_mutex);
}
static void stop(void) {
pthread_mutex_lock(&buffer_mutex);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + audio_size;
audio_occupancy = 0;
// if (enable_fill == 0) {
// debug(1, "stop enable_fill");
// }
pthread_mutex_unlock(&buffer_mutex);
if (stream_is_active == 1) {
pw_thread_loop_lock(data.loop);
// pw_stream_flush(data.stream, true);
pw_stream_set_active(data.stream, false);
pw_thread_loop_unlock(data.loop);
stream_is_active = 0;
// debug(1, "set stream inactive");
}
}
audio_output audio_pw = {.name = "pipewire",
.help = NULL,
.init = &init,
.deinit = &deinit,
.start = NULL,
.get_configuration = &get_configuration,
.configure = &configure,
.stop = &stop,
.is_running = NULL,
.flush = &flush,
.delay = &delay,
.stats = NULL,
.play = &play,
.volume = NULL,
.parameters = NULL,
.mute = NULL};
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