// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <array>
#include <atomic>
#include <memory>
#include <span>
#include <vector>
#include "audio_core/audio_core.h"
#include "audio_core/common/common.h"
#include "audio_core/sink/sink_stream.h"
#include "common/common_types.h"
#include "common/fixed_point.h"
#include "common/scope_exit.h"
#include "common/settings.h"
#include "core/core.h"
#include "core/core_timing.h"
namespace AudioCore::Sink {
void SinkStream::AppendBuffer(SinkBuffer& buffer, std::span<s16> samples) {
SCOPE_EXIT {
queue.enqueue(buffer);
++queued_buffers;
};
if (type == StreamType::In) {
return;
}
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
auto yuzu_volume{Settings::Volume()};
if (yuzu_volume > 1.0f) {
yuzu_volume = 0.6f + 20 * std::log10(yuzu_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.
// Front = 1.0
// Center = 0.596
// LFE = 0.354
// Back = 0.707
static constexpr std::array<f32, 4> down_mix_coeff{1.0, 0.596f, 0.354f, 0.707f};
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto fl =
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontLeft)]);
const auto fr =
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontRight)]);
const auto c =
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::Center)]);
const auto lfe =
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::LFE)]);
const auto bl =
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::BackLeft)]);
const auto br =
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::BackRight)]);
const auto left_sample{
static_cast<s32>((fl * down_mix_coeff[0] + c * down_mix_coeff[1] +
lfe * down_mix_coeff[2] + bl * down_mix_coeff[3]) *
volume)};
const auto right_sample{
static_cast<s32>((fr * down_mix_coeff[0] + c * down_mix_coeff[1] +
lfe * down_mix_coeff[2] + br * down_mix_coeff[3]) *
volume)};
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_buffer.Push(samples.subspan(0, samples.size() / system_channels * device_channels));
return;
}
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);
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_buffer.Push(new_samples);
return;
}
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);
}
std::vector<s16> SinkStream::ReleaseBuffer(u64 num_samples) {
constexpr s32 min = std::numeric_limits<s16>::min();
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;
}
void SinkStream::ClearQueue() {
samples_buffer.Pop();
while (queue.pop()) {
}
queued_buffers = 0;
playing_buffer = {};
playing_buffer.consumed = true;
}
void SinkStream::ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames) {
const std::size_t num_channels = GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
size_t frames_written{0};
// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
// paused and we'll desync, so just return.
if (system.IsPaused() || system.IsShuttingDown()) {
return;
}
while (frames_written < num_frames) {
// 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 (!queue.try_dequeue(playing_buffer)) {
// If no buffer was available we've underrun, just push the samples and
// continue.
samples_buffer.Push(&input_buffer[frames_written * frame_size],
(num_frames - frames_written) * frame_size);
frames_written = num_frames;
continue;
}
// Successfully dequeued a new buffer.
queued_buffers--;
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{std::min<u64>(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
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) {
playing_buffer.consumed = true;
}
}
std::memcpy(&last_frame[0], &input_buffer[(frames_written - 1) * frame_size], frame_size_bytes);
}
void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames) {
const std::size_t num_channels = GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
size_t frames_written{0};
size_t actual_frames_written{0};
// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
// paused and we'll desync, so just play silence.
if (system.IsPaused() || system.IsShuttingDown()) {
if (system.IsShuttingDown()) {
{
std::scoped_lock lk{release_mutex};
queued_buffers.store(0);
}
release_cv.notify_one();
}
static constexpr std::array<s16, 6> silence{};
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(&output_buffer[i * frame_size], &silence[0], frame_size_bytes);
}
return;
}
while (frames_written < num_frames) {
// 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 (!queue.try_dequeue(playing_buffer)) {
// If no buffer was available we've underrun, fill the remaining buffer with
// the last written frame and continue.
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(&output_buffer[i * frame_size], &last_frame[0], frame_size_bytes);
}
frames_written = num_frames;
continue;
}
// Successfully dequeued a new buffer.
queued_buffers--;
{ std::unique_lock lk{release_mutex}; }
release_cv.notify_one();
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{std::min<u64>(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
samples_buffer.Pop(&output_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
actual_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) {
playing_buffer.consumed = true;
}
}
std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
{
std::scoped_lock lk{sample_count_lock};
last_sample_count_update_time = system.CoreTiming().GetGlobalTimeNs();
min_played_sample_count = max_played_sample_count;
max_played_sample_count += actual_frames_written;
}
}
u64 SinkStream::GetExpectedPlayedSampleCount() {
std::scoped_lock lk{sample_count_lock};
auto cur_time{system.CoreTiming().GetGlobalTimeNs()};
auto time_delta{cur_time - last_sample_count_update_time};
auto exp_played_sample_count{min_played_sample_count +
(TargetSampleRate * time_delta) / std::chrono::seconds{1}};
// Add 15ms of latency in sample reporting to allow for some leeway in scheduler timings
return std::min<u64>(exp_played_sample_count, max_played_sample_count) + TargetSampleCount * 3;
}
void SinkStream::WaitFreeSpace(std::stop_token stop_token) {
std::unique_lock lk{release_mutex};
release_cv.wait_for(lk, std::chrono::milliseconds(5),
[this]() { return paused || queued_buffers < max_queue_size; });
if (queued_buffers > max_queue_size + 3) {
Common::CondvarWait(release_cv, lk, stop_token,
[this] { return paused || queued_buffers < max_queue_size; });
}
}
void SinkStream::SignalPause() {
{
std::scoped_lock lk{release_mutex};
paused = true;
}
release_cv.notify_one();
}
} // namespace AudioCore::Sink
