sunshine-sdk/src/audio.cpp
ReenigneArcher c2420427b1
style: adjust clang-format rules (#2186)
Co-authored-by: Vithorio Polten <reach@vithor.io>
2025-01-19 22:34:47 -05:00

322 lines
8.7 KiB
C++

/**
* @file src/audio.cpp
* @brief Definitions for audio capture and encoding.
*/
// standard includes
#include <thread>
// lib includes
#include <opus/opus_multistream.h>
// local includes
#include "audio.h"
#include "config.h"
#include "globals.h"
#include "logging.h"
#include "platform/common.h"
#include "thread_safe.h"
#include "utility.h"
namespace audio {
using namespace std::literals;
using opus_t = util::safe_ptr<OpusMSEncoder, opus_multistream_encoder_destroy>;
using sample_queue_t = std::shared_ptr<safe::queue_t<std::vector<float>>>;
static int start_audio_control(audio_ctx_t &ctx);
static void stop_audio_control(audio_ctx_t &);
static void apply_surround_params(opus_stream_config_t &stream, const stream_params_t &params);
int map_stream(int channels, bool quality);
constexpr auto SAMPLE_RATE = 48000;
// NOTE: If you adjust the bitrates listed here, make sure to update the
// corresponding bitrate adjustment logic in rtsp_stream::cmd_announce()
opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
{
SAMPLE_RATE,
2,
1,
1,
platf::speaker::map_stereo,
96000,
},
{
SAMPLE_RATE,
2,
1,
1,
platf::speaker::map_stereo,
512000,
},
{
SAMPLE_RATE,
6,
4,
2,
platf::speaker::map_surround51,
256000,
},
{
SAMPLE_RATE,
6,
6,
0,
platf::speaker::map_surround51,
1536000,
},
{
SAMPLE_RATE,
8,
5,
3,
platf::speaker::map_surround71,
450000,
},
{
SAMPLE_RATE,
8,
8,
0,
platf::speaker::map_surround71,
2048000,
},
};
void encodeThread(sample_queue_t samples, config_t config, void *channel_data) {
auto packets = mail::man->queue<packet_t>(mail::audio_packets);
auto stream = stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
if (config.flags[config_t::CUSTOM_SURROUND_PARAMS]) {
apply_surround_params(stream, config.customStreamParams);
}
// Encoding takes place on this thread
platf::adjust_thread_priority(platf::thread_priority_e::high);
opus_t opus {opus_multistream_encoder_create(
stream.sampleRate,
stream.channelCount,
stream.streams,
stream.coupledStreams,
stream.mapping,
OPUS_APPLICATION_RESTRICTED_LOWDELAY,
nullptr
)};
opus_multistream_encoder_ctl(opus.get(), OPUS_SET_BITRATE(stream.bitrate));
opus_multistream_encoder_ctl(opus.get(), OPUS_SET_VBR(0));
BOOST_LOG(info) << "Opus initialized: "sv << stream.sampleRate / 1000 << " kHz, "sv
<< stream.channelCount << " channels, "sv
<< stream.bitrate / 1000 << " kbps (total), LOWDELAY"sv;
auto frame_size = config.packetDuration * stream.sampleRate / 1000;
while (auto sample = samples->pop()) {
buffer_t packet {1400};
int bytes = opus_multistream_encode_float(opus.get(), sample->data(), frame_size, std::begin(packet), packet.size());
if (bytes < 0) {
BOOST_LOG(error) << "Couldn't encode audio: "sv << opus_strerror(bytes);
packets->stop();
return;
}
packet.fake_resize(bytes);
packets->raise(channel_data, std::move(packet));
}
}
void capture(safe::mail_t mail, config_t config, void *channel_data) {
auto shutdown_event = mail->event<bool>(mail::shutdown);
auto stream = stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
if (config.flags[config_t::CUSTOM_SURROUND_PARAMS]) {
apply_surround_params(stream, config.customStreamParams);
}
auto ref = get_audio_ctx_ref();
if (!ref) {
return;
}
auto init_failure_fg = util::fail_guard([&shutdown_event]() {
BOOST_LOG(error) << "Unable to initialize audio capture. The stream will not have audio."sv;
// Wait for shutdown to be signalled if we fail init.
// This allows streaming to continue without audio.
shutdown_event->view();
});
auto &control = ref->control;
if (!control) {
return;
}
// Order of priority:
// 1. Virtual sink
// 2. Audio sink
// 3. Host
std::string *sink = &ref->sink.host;
if (!config::audio.sink.empty()) {
sink = &config::audio.sink;
}
// Prefer the virtual sink if host playback is disabled or there's no other sink
if (ref->sink.null && (!config.flags[config_t::HOST_AUDIO] || sink->empty())) {
auto &null = *ref->sink.null;
switch (stream.channelCount) {
case 2:
sink = &null.stereo;
break;
case 6:
sink = &null.surround51;
break;
case 8:
sink = &null.surround71;
break;
}
}
// Only the first to start a session may change the default sink
if (!ref->sink_flag->exchange(true, std::memory_order_acquire)) {
// If the selected sink is different than the current one, change sinks.
ref->restore_sink = ref->sink.host != *sink;
if (ref->restore_sink) {
if (control->set_sink(*sink)) {
return;
}
}
}
auto frame_size = config.packetDuration * stream.sampleRate / 1000;
auto mic = control->microphone(stream.mapping, stream.channelCount, stream.sampleRate, frame_size);
if (!mic) {
return;
}
// Audio is initialized, so we don't want to print the failure message
init_failure_fg.disable();
// Capture takes place on this thread
platf::adjust_thread_priority(platf::thread_priority_e::critical);
auto samples = std::make_shared<sample_queue_t::element_type>(30);
std::thread thread {encodeThread, samples, config, channel_data};
auto fg = util::fail_guard([&]() {
samples->stop();
thread.join();
shutdown_event->view();
});
int samples_per_frame = frame_size * stream.channelCount;
while (!shutdown_event->peek()) {
std::vector<float> sample_buffer;
sample_buffer.resize(samples_per_frame);
auto status = mic->sample(sample_buffer);
switch (status) {
case platf::capture_e::ok:
break;
case platf::capture_e::timeout:
continue;
case platf::capture_e::reinit:
BOOST_LOG(info) << "Reinitializing audio capture"sv;
mic.reset();
do {
mic = control->microphone(stream.mapping, stream.channelCount, stream.sampleRate, frame_size);
if (!mic) {
BOOST_LOG(warning) << "Couldn't re-initialize audio input"sv;
}
} while (!mic && !shutdown_event->view(5s));
continue;
default:
return;
}
samples->raise(std::move(sample_buffer));
}
}
audio_ctx_ref_t get_audio_ctx_ref() {
static auto control_shared {safe::make_shared<audio_ctx_t>(start_audio_control, stop_audio_control)};
return control_shared.ref();
}
bool is_audio_ctx_sink_available(const audio_ctx_t &ctx) {
if (!ctx.control) {
return false;
}
const std::string &sink = ctx.sink.host.empty() ? config::audio.sink : ctx.sink.host;
if (sink.empty()) {
return false;
}
return ctx.control->is_sink_available(sink);
}
int map_stream(int channels, bool quality) {
int shift = quality ? 1 : 0;
switch (channels) {
case 2:
return STEREO + shift;
case 6:
return SURROUND51 + shift;
case 8:
return SURROUND71 + shift;
}
return STEREO;
}
int start_audio_control(audio_ctx_t &ctx) {
auto fg = util::fail_guard([]() {
BOOST_LOG(warning) << "There will be no audio"sv;
});
ctx.sink_flag = std::make_unique<std::atomic_bool>(false);
// The default sink has not been replaced yet.
ctx.restore_sink = false;
if (!(ctx.control = platf::audio_control())) {
return 0;
}
auto sink = ctx.control->sink_info();
if (!sink) {
// Let the calling code know it failed
ctx.control.reset();
return 0;
}
ctx.sink = std::move(*sink);
fg.disable();
return 0;
}
void stop_audio_control(audio_ctx_t &ctx) {
// restore audio-sink if applicable
if (!ctx.restore_sink) {
return;
}
// Change back to the host sink, unless there was none
const std::string &sink = ctx.sink.host.empty() ? config::audio.sink : ctx.sink.host;
if (!sink.empty()) {
// Best effort, it's allowed to fail
ctx.control->set_sink(sink);
}
}
void apply_surround_params(opus_stream_config_t &stream, const stream_params_t &params) {
stream.channelCount = params.channelCount;
stream.streams = params.streams;
stream.coupledStreams = params.coupledStreams;
stream.mapping = params.mapping;
}
} // namespace audio