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audio_pa.c
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audio_pa.c
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/*
* Asynchronous PulseAudio Backend. This file is part of Shairport Sync.
* Copyright (c) Mike Brady 2017
* 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.
*/
// Based (distantly, with thanks) on
// http://stackoverflow.com/questions/29977651/how-can-the-pulseaudio-asynchronous-library-be-used-to-play-raw-pcm-data
#include "audio.h"
#include "common.h"
#include <errno.h>
#include <pthread.h>
#include <pulse/pulseaudio.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
// note -- these are hacked and hardwired into this code.
#define FORMAT PA_SAMPLE_S16NE
#define RATE 44100
// Four seconds buffer -- should be plenty
#define buffer_allocation 44100 * 4 * 2 * 2
static pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER;
/*
static struct {
char *server;
char *sink;
char *service_name;
} pulse_options = {.server = NULL, .sink = NULL, .service_name = NULL};
*/
pa_threaded_mainloop *mainloop;
pa_mainloop_api *mainloop_api;
pa_context *context;
pa_stream *stream;
char *audio_lmb, *audio_umb, *audio_toq, *audio_eoq;
size_t audio_size = buffer_allocation;
size_t audio_occupancy;
void context_state_cb(pa_context *context, void *mainloop);
void stream_state_cb(pa_stream *s, void *mainloop);
void stream_success_cb(pa_stream *stream, int success, void *userdata);
void stream_write_cb(pa_stream *stream, size_t requested_bytes, void *userdata);
static int init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) {
// set up default values first
config.audio_backend_buffer_desired_length = 0.35;
config.audio_backend_latency_offset = 0;
// get settings from settings file
// do the "general" audio options. Note, these options are in the "general" stanza!
parse_general_audio_options();
// now the specific options
if (config.cfg != NULL) {
const char *str;
/* Get the Application Name. */
if (config_lookup_string(config.cfg, "pa.application_name", &str)) {
config.pa_application_name = (char *)str;
}
}
// finish collecting settings
// allocate space for the audio buffer
audio_lmb = malloc(audio_size);
if (audio_lmb == NULL)
die("Can't allocate %d bytes for pulseaudio buffer.", audio_size);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + audio_size;
audio_occupancy = 0;
// Get a mainloop and its context
mainloop = pa_threaded_mainloop_new();
if (mainloop == NULL)
die("could not create a pa_threaded_mainloop.");
mainloop_api = pa_threaded_mainloop_get_api(mainloop);
if (config.pa_application_name)
context = pa_context_new(mainloop_api, config.pa_application_name);
else
context = pa_context_new(mainloop_api, "Shairport Sync");
if (context == NULL)
die("could not create a new context for pulseaudio.");
// Set a callback so we can wait for the context to be ready
pa_context_set_state_callback(context, &context_state_cb, mainloop);
// Lock the mainloop so that it does not run and crash before the context is ready
pa_threaded_mainloop_lock(mainloop);
// Start the mainloop
if (pa_threaded_mainloop_start(mainloop) != 0)
die("could not start the pulseaudio threaded mainloop");
if (pa_context_connect(context, NULL, 0, NULL) != 0)
die("failed to connect to the pulseaudio context -- the error message is \"%s\".",
pa_strerror(pa_context_errno(context)));
// Wait for the context to be ready
for (;;) {
pa_context_state_t context_state = pa_context_get_state(context);
if (!PA_CONTEXT_IS_GOOD(context_state))
die("pa context is not good -- the error message \"%s\".",
pa_strerror(pa_context_errno(context)));
if (context_state == PA_CONTEXT_READY)
break;
pa_threaded_mainloop_wait(mainloop);
}
pa_threaded_mainloop_unlock(mainloop);
return 0;
}
static void deinit(void) {
pa_threaded_mainloop_stop(mainloop);
pa_threaded_mainloop_free(mainloop);
// debug(1, "pa deinit done");
}
static void start(__attribute__((unused)) int sample_rate,
__attribute__((unused)) int sample_format) {
uint32_t buffer_size_in_bytes = (uint32_t)2 * 2 * RATE * 0.1; // hard wired in here
// debug(1, "pa_buffer size is %u bytes.", buffer_size_in_bytes);
pa_threaded_mainloop_lock(mainloop);
// Create a playback stream
pa_sample_spec sample_specifications;
sample_specifications.format = FORMAT;
sample_specifications.rate = RATE;
sample_specifications.channels = 2;
pa_channel_map map;
pa_channel_map_init_stereo(&map);
stream = pa_stream_new(context, "Playback", &sample_specifications, &map);
pa_stream_set_state_callback(stream, stream_state_cb, mainloop);
pa_stream_set_write_callback(stream, stream_write_cb, mainloop);
// pa_stream_set_latency_update_callback(stream, stream_latency_cb, mainloop);
// recommended settings, i.e. server uses sensible values
pa_buffer_attr buffer_attr;
buffer_attr.maxlength = (uint32_t)-1;
buffer_attr.tlength = buffer_size_in_bytes;
buffer_attr.prebuf = (uint32_t)0;
buffer_attr.minreq = (uint32_t)-1;
// Settings copied as per the chromium browser source
pa_stream_flags_t stream_flags;
stream_flags = PA_STREAM_START_CORKED | PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_NOT_MONOTONIC |
// PA_STREAM_AUTO_TIMING_UPDATE;
PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_ADJUST_LATENCY;
// Connect stream to the default audio output sink
if (pa_stream_connect_playback(stream, NULL, &buffer_attr, stream_flags, NULL, NULL) != 0)
die("could not connect to the pulseaudio playback stream -- the error message is \"%s\".",
pa_strerror(pa_context_errno(context)));
// Wait for the stream to be ready
for (;;) {
pa_stream_state_t stream_state = pa_stream_get_state(stream);
if (!PA_STREAM_IS_GOOD(stream_state))
die("stream state is no longer good while waiting for stream to become ready -- the error "
"message is \"%s\".",
pa_strerror(pa_context_errno(context)));
if (stream_state == PA_STREAM_READY)
break;
pa_threaded_mainloop_wait(mainloop);
}
pa_threaded_mainloop_unlock(mainloop);
}
static void play(void *buf, int samples) {
// debug(1,"pa_play of %d samples.",samples);
// copy the samples into the queue
size_t bytes_to_transfer = samples * 2 * 2;
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_occupancy += bytes_to_transfer;
pthread_mutex_lock(&buffer_mutex);
audio_eoq += bytes_to_transfer;
pthread_mutex_unlock(&buffer_mutex);
} 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);
pthread_mutex_lock(&buffer_mutex);
audio_occupancy += bytes_to_transfer;
pthread_mutex_unlock(&buffer_mutex);
audio_eoq = audio_lmb + bytes_to_transfer - space_to_end_of_buffer;
}
if ((audio_occupancy >= 11025 * 2 * 2) && (pa_stream_is_corked(stream))) {
// debug(1,"Uncorked");
pa_threaded_mainloop_lock(mainloop);
pa_stream_cork(stream, 0, stream_success_cb, mainloop);
pa_threaded_mainloop_unlock(mainloop);
}
}
int pa_delay(long *the_delay) {
long result = 0;
int reply = -ENODEV;
pa_usec_t latency;
int negative;
pa_threaded_mainloop_lock(mainloop);
int gl = pa_stream_get_latency(stream, &latency, &negative);
pa_threaded_mainloop_unlock(mainloop);
if (gl == PA_ERR_NODATA) {
// debug(1, "No latency data yet.");
reply = -ENODEV;
} else if (gl != 0) {
// debug(1,"Error %d getting latency.",gl);
reply = -EIO;
} else {
result = (audio_occupancy / (2 * 2)) + (latency * 44100) / 1000000;
reply = 0;
}
*the_delay = result;
return reply;
}
void flush(void) {
// Cork the stream so it will stop playing
pa_threaded_mainloop_lock(mainloop);
if (pa_stream_is_corked(stream) == 0) {
// debug(1,"Flush and cork for flush.");
pa_stream_flush(stream, stream_success_cb, NULL);
pa_stream_cork(stream, 1, stream_success_cb, mainloop);
}
pa_threaded_mainloop_unlock(mainloop);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + audio_size;
audio_occupancy = 0;
}
static void stop(void) {
// Cork the stream so it will stop playing
pa_threaded_mainloop_lock(mainloop);
if (pa_stream_is_corked(stream) == 0) {
// debug(1,"Flush and cork for stop.");
pa_stream_flush(stream, stream_success_cb, NULL);
pa_stream_cork(stream, 1, stream_success_cb, mainloop);
}
pa_threaded_mainloop_unlock(mainloop);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + audio_size;
audio_occupancy = 0;
// debug(1,"pa stop");
pa_stream_disconnect(stream);
}
static void help(void) { printf(" no settings.\n"); }
audio_output audio_pa = {.name = "pa",
.help = &help,
.init = &init,
.deinit = &deinit,
.start = &start,
.stop = &stop,
.flush = &flush,
.delay = &pa_delay,
.play = &play,
.volume = NULL,
.parameters = NULL,
.mute = NULL};
void context_state_cb(__attribute__((unused)) pa_context *context, void *mainloop) {
pa_threaded_mainloop_signal(mainloop, 0);
}
void stream_state_cb(__attribute__((unused)) pa_stream *s, void *mainloop) {
pa_threaded_mainloop_signal(mainloop, 0);
}
void stream_write_cb(pa_stream *stream, size_t requested_bytes,
__attribute__((unused)) void *userdata) {
/*
// play with timing information
const struct pa_timing_info *ti = pa_stream_get_timing_info(stream);
if ((ti == NULL) || (ti->write_index_corrupt)) {
debug(2, "Timing info invalid");
} else {
struct timeval time_now;
pa_gettimeofday(&time_now);
uint64_t time_now_fp = ((uint64_t)time_now.tv_sec << 32) +
((uint64_t)time_now.tv_usec << 32) / 1000000; // types okay
uint64_t time_of_ti_fp = ((uint64_t)(ti->timestamp.tv_sec) << 32) +
((uint64_t)(ti->timestamp.tv_usec) << 32) / 1000000; // types okay
if (time_now_fp >= time_of_ti_fp) {
uint64_t estimate_age = ((time_now_fp - time_of_ti_fp) * 1000000) >> 32;
uint64_t bytes_in_buffer = ti->write_index - ti->read_index;
pa_usec_t microseconds_to_write_buffer = (bytes_in_buffer * 1000000) / (44100 * 2 * 2);
pa_usec_t ea = (pa_usec_t)estimate_age;
pa_usec_t pa_latency = ti->sink_usec + ti->transport_usec + microseconds_to_write_buffer;
pa_usec_t estimated_latency = pa_latency - estimate_age;
// debug(1,"Estimated latency is %d microseconds.",estimated_latency);
// } else {
// debug(1, "Time now is earlier than time of timing information");
}
}
*/
int bytes_to_transfer = requested_bytes;
int bytes_transferred = 0;
uint8_t *buffer = NULL;
while ((bytes_to_transfer > 0) && (audio_occupancy > 0)) {
size_t bytes_we_can_transfer = bytes_to_transfer;
if (audio_occupancy < bytes_we_can_transfer) {
// debug(1, "Underflow? We have %d bytes but we are asked for %d bytes", audio_occupancy,
// bytes_we_can_transfer);
pa_stream_cork(stream, 1, stream_success_cb, mainloop);
// debug(1, "Corked");
bytes_we_can_transfer = audio_occupancy;
}
// bytes we can transfer will never be greater than the bytes available
pa_stream_begin_write(stream, (void **)&buffer, &bytes_we_can_transfer);
if (bytes_we_can_transfer <= (size_t)(audio_umb - audio_toq)) {
// the bytes are all in a row in the audo buffer
memcpy(buffer, audio_toq, bytes_we_can_transfer);
audio_toq += bytes_we_can_transfer;
// lock
pthread_mutex_lock(&buffer_mutex);
audio_occupancy -= bytes_we_can_transfer;
pthread_mutex_unlock(&buffer_mutex);
// unlock
pa_stream_write(stream, buffer, bytes_we_can_transfer, NULL, 0LL, PA_SEEK_RELATIVE);
bytes_transferred += 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(buffer, audio_toq, first_portion_to_write);
uint8_t *new_buffer = buffer + first_portion_to_write;
memcpy(new_buffer, audio_lmb, bytes_we_can_transfer - first_portion_to_write);
pa_stream_write(stream, buffer, bytes_we_can_transfer, NULL, 0LL, PA_SEEK_RELATIVE);
bytes_transferred += bytes_we_can_transfer;
audio_toq = audio_lmb + bytes_we_can_transfer - first_portion_to_write;
// lock
pthread_mutex_lock(&buffer_mutex);
audio_occupancy -= bytes_we_can_transfer;
pthread_mutex_unlock(&buffer_mutex);
// unlock
}
bytes_to_transfer -= bytes_we_can_transfer;
// debug(1,"audio_toq is %llx",audio_toq);
}
// debug(1,"<<<Frames requested %d, written to pa: %d, corked status:
// %d.",requested_bytes/4,bytes_transferred/4,pa_stream_is_corked(stream));
}
void alt_stream_write_cb(pa_stream *stream, size_t requested_bytes,
__attribute__((unused)) void *userdata) {
// debug(1, "***Bytes requested bytes %d.", requested_bytes);
size_t bytes_remaining = requested_bytes;
while (bytes_remaining > 0) {
uint8_t *buffer = NULL;
size_t bytes_to_fill = 44100;
size_t i;
if (bytes_to_fill > bytes_remaining)
bytes_to_fill = bytes_remaining;
pa_stream_begin_write(stream, (void **)&buffer, &bytes_to_fill);
if (buffer) {
for (i = 0; i < bytes_to_fill; i += 2) {
buffer[i] = (i % 100) * 40 / 100 + 44;
buffer[i + 1] = (i % 100) * 40 / 100 + 44;
}
} else {
die("buffer not allocated in alt_stream_write_cb.");
}
pa_stream_write(stream, buffer, bytes_to_fill, NULL, 0LL, PA_SEEK_RELATIVE);
bytes_remaining -= bytes_to_fill;
}
}
void stream_success_cb(__attribute__((unused)) pa_stream *stream,
__attribute__((unused)) int success,
__attribute__((unused)) void *userdata) {
return;
}