// 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/settings.h"
#include "core/core.h"
namespace AudioCore::Sink {
void SinkStream::AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) {
if (type == StreamType::In) {
queue.enqueue(buffer);
queued_buffers++;
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.
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++;
}
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;
}
if (queued_buffers > max_queue_size) {
Stall();
}
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);
if (queued_buffers <= max_queue_size) {
Unstall();
}
}
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};
// 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()) {
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;
}
// Due to many frames being queued up with nvdec (5 frames or so?), a lot of buffers also get
// queued up (30+) but not all at once, which causes constant stalling here, so just let the
// video play out without attempting to stall.
// Can hopefully remove this later with a more complete NVDEC implementation.
const auto nvdec_active{system.AudioCore().IsNVDECActive()};
// Core timing cannot be paused in single-core mode, so Stall ends up being called over and over
// and never recovers to a normal state, so just skip attempting to sync things on single-core.
if (system.IsMulticore() && !nvdec_active && queued_buffers > max_queue_size) {
Stall();
} else if (system.IsMulticore() && queued_buffers <= max_queue_size) {
Unstall();
}
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--;
}
// 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;
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);
if (system.IsMulticore() && queued_buffers <= max_queue_size) {
Unstall();
}
}
void SinkStream::Stall() {
std::scoped_lock lk{stall_guard};
if (stalled_lock) {
return;
}
stalled_lock = system.StallApplication();
}
void SinkStream::Unstall() {
std::scoped_lock lk{stall_guard};
if (!stalled_lock) {
return;
}
system.UnstallApplication();
stalled_lock.unlock();
}
} // namespace AudioCore::Sink
