mirror of
https://github.com/LizardByte/Sunshine.git
synced 2024-11-18 02:09:49 +00:00
271 lines
6.6 KiB
C++
271 lines
6.6 KiB
C++
#include <thread>
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#include <opus/opus_multistream.h>
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#include "platform/common.h"
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#include "audio.h"
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#include "config.h"
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#include "main.h"
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#include "thread_safe.h"
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#include "utility.h"
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namespace audio {
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using namespace std::literals;
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using opus_t = util::safe_ptr<OpusMSEncoder, opus_multistream_encoder_destroy>;
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using sample_queue_t = std::shared_ptr<safe::queue_t<std::vector<std::int16_t>>>;
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struct audio_ctx_t {
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// We want to change the sink for the first stream only
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std::unique_ptr<std::atomic_bool> sink_flag;
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std::unique_ptr<platf::audio_control_t> control;
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bool restore_sink;
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platf::sink_t sink;
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};
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static int start_audio_control(audio_ctx_t &ctx);
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static void stop_audio_control(audio_ctx_t &);
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int map_stream(int channels, bool quality);
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constexpr auto SAMPLE_RATE = 48000;
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opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
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{
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SAMPLE_RATE,
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2,
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1,
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1,
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platf::speaker::map_stereo,
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},
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{
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SAMPLE_RATE,
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6,
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4,
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2,
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platf::speaker::map_surround51,
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},
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{
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SAMPLE_RATE,
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6,
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6,
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0,
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platf::speaker::map_surround51,
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},
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{
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SAMPLE_RATE,
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8,
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5,
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3,
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platf::speaker::map_surround71,
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},
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{
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SAMPLE_RATE,
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8,
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8,
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0,
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platf::speaker::map_surround71,
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},
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};
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auto control_shared = safe::make_shared<audio_ctx_t>(start_audio_control, stop_audio_control);
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void encodeThread(sample_queue_t samples, config_t config, void *channel_data) {
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auto packets = mail::man->queue<packet_t>(mail::audio_packets);
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//FIXME: Pick correct opus_stream_config_t based on config.channels
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auto stream = &stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
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opus_t opus { opus_multistream_encoder_create(
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stream->sampleRate,
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stream->channelCount,
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stream->streams,
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stream->coupledStreams,
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stream->mapping,
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OPUS_APPLICATION_AUDIO,
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nullptr) };
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// For some reason, audio is crackling when the encoder is set to constant bitstream.
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// We simulate a constant bitstream with OPUS_SET_BITRATE(OPUS_BITRATE_MAX) -->
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// which tries to occupy as much space as possible in the packet
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opus_multistream_encoder_ctl(opus.get(), OPUS_SET_BITRATE(OPUS_BITRATE_MAX));
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auto frame_size = config.packetDuration * stream->sampleRate / 1000;
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while(auto sample = samples->pop()) {
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buffer_t packet { 1400 }; // 1KB
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int bytes = opus_multistream_encode(opus.get(), sample->data(), frame_size, std::begin(packet), packet.size());
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if(bytes < 0) {
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BOOST_LOG(error) << "Couldn't encode audio: "sv << opus_strerror(bytes);
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packets->stop();
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return;
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}
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// Even with OPUS_SET_BITRATE(OPUS_BITRATE_MAX), silent packets are smaller than the rest
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// Drop silent packets to ensure Moonlight won't complain
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// A packet size of 128 seems a reasonable enough threshold
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if(bytes < 128) {
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BOOST_LOG(verbose) << "Dropped silent packet"sv;
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continue;
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}
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packet.fake_resize(bytes);
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packets->raise(channel_data, std::move(packet));
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}
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}
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void capture(safe::mail_t mail, config_t config, void *channel_data) {
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auto shutdown_event = mail->event<bool>(mail::shutdown);
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//FIXME: Pick correct opus_stream_config_t based on config.channels
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auto stream = &stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
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auto ref = control_shared.ref();
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if(!ref) {
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return;
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}
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auto &control = ref->control;
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if(!control) {
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shutdown_event->view();
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return;
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}
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// Order of priorty:
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// 1. Config
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// 2. Virtual if available
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// 3. Host
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std::string *sink = &ref->sink.host;
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if(!config::audio.sink.empty()) {
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sink = &config::audio.sink;
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}
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else if(ref->sink.null) {
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auto &null = *ref->sink.null;
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switch(stream->channelCount) {
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case 2:
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sink = &null.stereo;
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break;
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case 6:
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sink = &null.surround51;
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break;
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case 8:
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sink = &null.surround71;
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break;
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}
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}
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// Only the first to start a session may change the default sink
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if(!ref->sink_flag->exchange(true, std::memory_order_acquire)) {
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ref->restore_sink = !config.flags[config_t::HOST_AUDIO];
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// If the client requests audio on the host, don't change the default sink
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if(!config.flags[config_t::HOST_AUDIO] && control->set_sink(*sink)) {
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return;
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}
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}
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auto samples = std::make_shared<sample_queue_t::element_type>(30);
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std::thread thread { encodeThread, samples, config, channel_data };
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auto fg = util::fail_guard([&]() {
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samples->stop();
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thread.join();
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shutdown_event->view();
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});
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auto frame_size = config.packetDuration * stream->sampleRate / 1000;
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int samples_per_frame = frame_size * stream->channelCount;
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auto mic = control->microphone(stream->mapping, stream->channelCount, stream->sampleRate, frame_size);
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if(!mic) {
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BOOST_LOG(error) << "Couldn't create audio input"sv;
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return;
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}
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while(!shutdown_event->peek()) {
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std::vector<std::int16_t> sample_buffer;
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sample_buffer.resize(samples_per_frame);
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auto status = mic->sample(sample_buffer);
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switch(status) {
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case platf::capture_e::ok:
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break;
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case platf::capture_e::timeout:
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continue;
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case platf::capture_e::reinit:
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mic.reset();
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mic = control->microphone(stream->mapping, stream->channelCount, stream->sampleRate, frame_size);
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if(!mic) {
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BOOST_LOG(error) << "Couldn't re-initialize audio input"sv;
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return;
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}
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return;
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default:
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return;
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}
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samples->raise(std::move(sample_buffer));
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}
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}
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int map_stream(int channels, bool quality) {
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int shift = quality ? 1 : 0;
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switch(channels) {
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case 2:
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return STEREO;
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case 6:
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return SURROUND51 + shift;
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case 8:
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return SURROUND71 + shift;
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}
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return STEREO;
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}
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int start_audio_control(audio_ctx_t &ctx) {
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auto fg = util::fail_guard([]() {
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BOOST_LOG(warning) << "There will be no audio"sv;
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});
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ctx.sink_flag = std::make_unique<std::atomic_bool>(false);
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// The default sink has not been replaced yet.
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ctx.restore_sink = false;
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if(!(ctx.control = platf::audio_control())) {
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return 0;
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}
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auto sink = ctx.control->sink_info();
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if(!sink) {
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// Let the calling code know it failed
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ctx.control.reset();
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return 0;
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}
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ctx.sink = std::move(*sink);
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fg.disable();
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return 0;
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}
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void stop_audio_control(audio_ctx_t &ctx) {
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// restore audio-sink if applicable
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if(!ctx.restore_sink) {
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return;
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}
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const std::string &sink = config::audio.sink.empty() ? ctx.sink.host : config::audio.sink;
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if(!sink.empty()) {
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// Best effort, it's allowed to fail
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ctx.control->set_sink(sink);
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}
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}
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} // namespace audio
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