// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include <atomic>
#include "audio_core/audio_core.h"
#include "audio_core/audio_event.h"
#include "audio_core/audio_manager.h"
#include "audio_core/sink/sdl2_sink.h"
#include "audio_core/sink/sink_stream.h"
#include "common/assert.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
#include "common/reader_writer_queue.h"
#include "common/ring_buffer.h"
#include "common/settings.h"
#include "core/core.h"
// Ignore -Wimplicit-fallthrough due to https://github.com/libsdl-org/SDL/issues/4307
#ifdef __clang__
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wimplicit-fallthrough"
#endif
#include <SDL.h>
#ifdef __clang__
#pragma clang diagnostic pop
#endif
namespace AudioCore::Sink {
/**
* SDL sink stream, responsible for sinking samples to hardware.
*/
class SDLSinkStream final : public SinkStream {
public:
/**
* Create a new sink stream.
*
* @param device_channels_ - Number of channels supported by the hardware.
* @param system_channels_ - Number of channels the audio systems expect.
* @param output_device - Name of the output device to use for this stream.
* @param input_device - Name of the input device to use for this stream.
* @param type_ - Type of this stream.
* @param system_ - Core system.
* @param event - Event used only for audio renderer, signalled on buffer consume.
*/
SDLSinkStream(u32 device_channels_, const u32 system_channels_,
const std::string& output_device, const std::string& input_device,
const StreamType type_, Core::System& system_)
: type{type_}, system{system_} {
system_channels = system_channels_;
device_channels = device_channels_;
SDL_AudioSpec spec;
spec.freq = TargetSampleRate;
spec.channels = static_cast<u8>(device_channels);
spec.format = AUDIO_S16SYS;
if (type == StreamType::Render) {
spec.samples = TargetSampleCount;
} else {
spec.samples = 1024;
}
spec.callback = &SDLSinkStream::DataCallback;
spec.userdata = this;
playing_buffer.consumed = true;
std::string device_name{output_device};
bool capture{false};
if (type == StreamType::In) {
device_name = input_device;
capture = true;
}
SDL_AudioSpec obtained;
if (device_name.empty()) {
device = SDL_OpenAudioDevice(nullptr, capture, &spec, &obtained, false);
} else {
device = SDL_OpenAudioDevice(device_name.c_str(), capture, &spec, &obtained, false);
}
if (device == 0) {
LOG_CRITICAL(Audio_Sink, "Error opening SDL audio device: {}", SDL_GetError());
return;
}
LOG_DEBUG(Service_Audio,
"Opening sdl stream {} with: rate {} channels {} (system channels {}) "
" samples {}",
device, obtained.freq, obtained.channels, system_channels, obtained.samples);
}
/**
* Destroy the sink stream.
*/
~SDLSinkStream() override {
if (device == 0) {
return;
}
SDL_CloseAudioDevice(device);
}
/**
* Finalize the sink stream.
*/
void Finalize() override {
if (device == 0) {
return;
}
SDL_CloseAudioDevice(device);
}
/**
* Start the sink stream.
*
* @param resume - Set to true if this is resuming the stream a previously-active stream.
* Default false.
*/
void Start(const bool resume = false) override {
if (device == 0) {
return;
}
if (resume && was_playing) {
SDL_PauseAudioDevice(device, 0);
paused = false;
} else if (!resume) {
SDL_PauseAudioDevice(device, 0);
paused = false;
}
}
/**
* Stop the sink stream.
*/
void Stop() {
if (device == 0) {
return;
}
SDL_PauseAudioDevice(device, 1);
paused = true;
}
/**
* Append a new buffer and its samples to a waiting queue to play.
*
* @param buffer - Audio buffer information to be queued.
* @param samples - The s16 samples to be queue for playback.
*/
void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override {
if (type == StreamType::In) {
queue.enqueue(buffer);
queued_buffers++;
} else {
constexpr s32 min = std::numeric_limits<s16>::min();
constexpr s32 max = std::numeric_limits<s16>::max();
auto yuzu_volume{Settings::Volume()};
auto volume{system_volume * device_volume * yuzu_volume};
if (system_channels == 6 && device_channels == 2) {
// We're given 6 channels, but our device only outputs 2, so downmix.
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] *
down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] *
down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackLeft)] *
down_mix_coeff[3]) *
volume)
.to_int()};
const auto right_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] *
down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] *
down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackRight)] *
down_mix_coeff[3]) *
volume)
.to_int()};
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
static_cast<s16>(std::clamp(left_sample, min, max));
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
static_cast<s16>(std::clamp(right_sample, min, max));
}
samples.resize(samples.size() / system_channels * device_channels);
} else if (system_channels == 2 && device_channels == 6) {
// We need moar samples! Not all games will provide 6 channel audio.
// TODO: Implement some upmixing here. Currently just passthrough, with other
// channels left as silence.
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
const auto right_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] =
right_sample;
}
samples = std::move(new_samples);
} else if (volume != 1.0f) {
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(std::clamp(
static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
}
samples_buffer.Push(samples);
queue.enqueue(buffer);
queued_buffers++;
}
}
/**
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
*
* @param num_samples - Maximum number of samples to receive.
* @return Vector of recorded samples. May have fewer than num_samples.
*/
std::vector<s16> ReleaseBuffer(const u64 num_samples) override {
static constexpr s32 min = std::numeric_limits<s16>::min();
static constexpr s32 max = std::numeric_limits<s16>::max();
auto samples{samples_buffer.Pop(num_samples)};
// TODO: Up-mix to 6 channels if the game expects it.
// For audio input this is unlikely to ever be the case though.
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
// TODO: Play with this and find something that works better.
auto volume{system_volume * device_volume * 8};
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
if (samples.size() < num_samples) {
samples.resize(num_samples, 0);
}
return samples;
}
/**
* Check if a certain buffer has been consumed (fully played).
*
* @param tag - Unique tag of a buffer to check for.
* @return True if the buffer has been played, otherwise false.
*/
bool IsBufferConsumed(const u64 tag) override {
if (released_buffer.tag == 0) {
if (!released_buffers.try_dequeue(released_buffer)) {
return false;
}
}
if (released_buffer.tag == tag) {
released_buffer.tag = 0;
return true;
}
return false;
}
/**
* Empty out the buffer queue.
*/
void ClearQueue() override {
samples_buffer.Pop();
while (queue.pop()) {
}
while (released_buffers.pop()) {
}
released_buffer = {};
playing_buffer = {};
playing_buffer.consumed = true;
queued_buffers = 0;
}
private:
/**
* Signal events back to the audio system that a buffer was played/can be filled.
*
* @param buffer - Consumed audio buffer to be released.
*/
void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) {
auto& manager{system.AudioCore().GetAudioManager()};
switch (type) {
case StreamType::Out:
released_buffers.enqueue(buffer);
manager.SetEvent(Event::Type::AudioOutManager, true);
break;
case StreamType::In:
released_buffers.enqueue(buffer);
manager.SetEvent(Event::Type::AudioInManager, true);
break;
case StreamType::Render:
break;
}
}
/**
* Main callback from SDL. Either expects samples from us (audio render/audio out), or will
* provide samples to be copied (audio in).
*
* @param userdata - Custom data pointer passed along, points to a SDLSinkStream.
* @param stream - Buffer of samples to be filled or read.
* @param len - Length of the stream in bytes.
*/
static void DataCallback(void* userdata, Uint8* stream, int len) {
auto* impl = static_cast<SDLSinkStream*>(userdata);
if (!impl) {
return;
}
const std::size_t num_channels = impl->GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
const std::size_t num_frames{len / num_channels / sizeof(s16)};
size_t frames_written{0};
[[maybe_unused]] bool underrun{false};
if (impl->type == StreamType::In) {
std::span<s16> input_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size};
while (frames_written < num_frames) {
auto& playing_buffer{impl->playing_buffer};
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
// If no buffer was available we've underrun, just push the samples and
// continue.
underrun = true;
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
(num_frames - frames_written) * frame_size);
frames_written = num_frames;
continue;
} else {
impl->queued_buffers--;
impl->SignalEvent(impl->playing_buffer);
}
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{
std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
impl->playing_buffer.consumed = true;
}
}
std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
} else {
std::span<s16> output_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size};
while (frames_written < num_frames) {
auto& playing_buffer{impl->playing_buffer};
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
// If no buffer was available we've underrun, fill the remaining buffer with
// the last written frame and continue.
underrun = true;
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0],
frame_size_bytes);
}
frames_written = num_frames;
continue;
} else {
impl->queued_buffers--;
impl->SignalEvent(impl->playing_buffer);
}
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{
std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
impl->playing_buffer.consumed = true;
}
}
std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
}
}
/// SDL device id of the opened input/output device
SDL_AudioDeviceID device{};
/// Type of this stream
StreamType type;
/// Core system
Core::System& system;
/// Ring buffer of the samples waiting to be played or consumed
Common::RingBuffer<s16, 0x10000> samples_buffer;
/// Audio buffers queued and waiting to play
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue;
/// The currently-playing audio buffer
::AudioCore::Sink::SinkBuffer playing_buffer{};
/// Audio buffers which have been played and are in queue to be released by the audio system
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{};
/// Currently released buffer waiting to be taken by the audio system
::AudioCore::Sink::SinkBuffer released_buffer{};
/// The last played (or received) frame of audio, used when the callback underruns
std::array<s16, MaxChannels> last_frame{};
};
SDLSink::SDLSink(std::string_view target_device_name) {
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
LOG_CRITICAL(Audio_Sink, "SDL_InitSubSystem audio failed: {}", SDL_GetError());
return;
}
}
if (target_device_name != auto_device_name && !target_device_name.empty()) {
output_device = target_device_name;
} else {
output_device.clear();
}
device_channels = 2;
}
SDLSink::~SDLSink() = default;
SinkStream* SDLSink::AcquireSinkStream(Core::System& system, const u32 system_channels,
const std::string&, const StreamType type) {
SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<SDLSinkStream>(
device_channels, system_channels, output_device, input_device, type, system));
return stream.get();
}
void SDLSink::CloseStream(const SinkStream* stream) {
for (size_t i = 0; i < sink_streams.size(); i++) {
if (sink_streams[i].get() == stream) {
sink_streams[i].reset();
sink_streams.erase(sink_streams.begin() + i);
break;
}
}
}
void SDLSink::CloseStreams() {
sink_streams.clear();
}
void SDLSink::PauseStreams() {
for (auto& stream : sink_streams) {
stream->Stop();
}
}
void SDLSink::UnpauseStreams() {
for (auto& stream : sink_streams) {
stream->Start();
}
}
f32 SDLSink::GetDeviceVolume() const {
if (sink_streams.empty()) {
return 1.0f;
}
return sink_streams[0]->GetDeviceVolume();
}
void SDLSink::SetDeviceVolume(const f32 volume) {
for (auto& stream : sink_streams) {
stream->SetDeviceVolume(volume);
}
}
void SDLSink::SetSystemVolume(const f32 volume) {
for (auto& stream : sink_streams) {
stream->SetSystemVolume(volume);
}
}
std::vector<std::string> ListSDLSinkDevices(const bool capture) {
std::vector<std::string> device_list;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
LOG_CRITICAL(Audio_Sink, "SDL_InitSubSystem audio failed: {}", SDL_GetError());
return {};
}
}
const int device_count = SDL_GetNumAudioDevices(capture);
for (int i = 0; i < device_count; ++i) {
device_list.emplace_back(SDL_GetAudioDeviceName(i, 0));
}
return device_list;
}
} // namespace AudioCore::Sink
