285 lines
11 KiB
C++
285 lines
11 KiB
C++
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
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// SPDX-License-Identifier: GPL-2.0-or-later
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#include <array>
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#include <atomic>
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#include <memory>
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#include <span>
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#include <vector>
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#include "audio_core/audio_core.h"
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#include "audio_core/common/common.h"
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#include "audio_core/sink/sink_stream.h"
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#include "common/common_types.h"
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#include "common/fixed_point.h"
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#include "common/settings.h"
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#include "core/core.h"
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namespace AudioCore::Sink {
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void SinkStream::AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) {
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if (type == StreamType::In) {
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queue.enqueue(buffer);
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queued_buffers++;
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return;
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}
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constexpr s32 min{std::numeric_limits<s16>::min()};
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constexpr s32 max{std::numeric_limits<s16>::max()};
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auto yuzu_volume{Settings::Volume()};
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if (yuzu_volume > 1.0f) {
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yuzu_volume = 0.6f + 20 * std::log10(yuzu_volume);
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}
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auto volume{system_volume * device_volume * yuzu_volume};
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if (system_channels == 6 && device_channels == 2) {
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// We're given 6 channels, but our device only outputs 2, so downmix.
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constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
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for (u32 read_index = 0, write_index = 0; read_index < samples.size();
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read_index += system_channels, write_index += device_channels) {
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const auto left_sample{
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((Common::FixedPoint<49, 15>(
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samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
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down_mix_coeff[0] +
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samples[read_index + static_cast<u32>(Channels::Center)] * down_mix_coeff[1] +
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samples[read_index + static_cast<u32>(Channels::LFE)] * down_mix_coeff[2] +
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samples[read_index + static_cast<u32>(Channels::BackLeft)] * down_mix_coeff[3]) *
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volume)
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.to_int()};
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const auto right_sample{
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((Common::FixedPoint<49, 15>(
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samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
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down_mix_coeff[0] +
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samples[read_index + static_cast<u32>(Channels::Center)] * down_mix_coeff[1] +
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samples[read_index + static_cast<u32>(Channels::LFE)] * down_mix_coeff[2] +
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samples[read_index + static_cast<u32>(Channels::BackRight)] * down_mix_coeff[3]) *
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volume)
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.to_int()};
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samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
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static_cast<s16>(std::clamp(left_sample, min, max));
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samples[write_index + static_cast<u32>(Channels::FrontRight)] =
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static_cast<s16>(std::clamp(right_sample, min, max));
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}
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samples.resize(samples.size() / system_channels * device_channels);
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} else if (system_channels == 2 && device_channels == 6) {
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// We need moar samples! Not all games will provide 6 channel audio.
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// TODO: Implement some upmixing here. Currently just passthrough, with other
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// channels left as silence.
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std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
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for (u32 read_index = 0, write_index = 0; read_index < samples.size();
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read_index += system_channels, write_index += device_channels) {
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const auto left_sample{static_cast<s16>(std::clamp(
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static_cast<s32>(
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static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
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volume),
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min, max))};
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new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
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const auto right_sample{static_cast<s16>(std::clamp(
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static_cast<s32>(
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static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
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volume),
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min, max))};
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new_samples[write_index + static_cast<u32>(Channels::FrontRight)] = right_sample;
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}
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samples = std::move(new_samples);
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} else if (volume != 1.0f) {
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for (u32 i = 0; i < samples.size(); i++) {
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samples[i] = static_cast<s16>(
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std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
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}
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}
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samples_buffer.Push(samples);
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queue.enqueue(buffer);
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queued_buffers++;
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}
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std::vector<s16> SinkStream::ReleaseBuffer(u64 num_samples) {
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constexpr s32 min = std::numeric_limits<s16>::min();
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constexpr s32 max = std::numeric_limits<s16>::max();
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auto samples{samples_buffer.Pop(num_samples)};
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// TODO: Up-mix to 6 channels if the game expects it.
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// For audio input this is unlikely to ever be the case though.
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// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
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// TODO: Play with this and find something that works better.
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auto volume{system_volume * device_volume * 8};
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for (u32 i = 0; i < samples.size(); i++) {
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samples[i] = static_cast<s16>(
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std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
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}
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if (samples.size() < num_samples) {
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samples.resize(num_samples, 0);
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}
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return samples;
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}
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void SinkStream::ClearQueue() {
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samples_buffer.Pop();
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while (queue.pop()) {
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}
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queued_buffers = 0;
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playing_buffer = {};
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playing_buffer.consumed = true;
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}
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void SinkStream::ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames) {
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const std::size_t num_channels = GetDeviceChannels();
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const std::size_t frame_size = num_channels;
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const std::size_t frame_size_bytes = frame_size * sizeof(s16);
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size_t frames_written{0};
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// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
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// paused and we'll desync, so just return.
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if (system.IsPaused() || system.IsShuttingDown()) {
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return;
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}
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if (queued_buffers > max_queue_size) {
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Stall();
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}
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while (frames_written < num_frames) {
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// If the playing buffer has been consumed or has no frames, we need a new one
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if (playing_buffer.consumed || playing_buffer.frames == 0) {
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if (!queue.try_dequeue(playing_buffer)) {
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// If no buffer was available we've underrun, just push the samples and
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// continue.
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samples_buffer.Push(&input_buffer[frames_written * frame_size],
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(num_frames - frames_written) * frame_size);
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frames_written = num_frames;
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continue;
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}
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// Successfully dequeued a new buffer.
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queued_buffers--;
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}
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// Get the minimum frames available between the currently playing buffer, and the
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// amount we have left to fill
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size_t frames_available{std::min<u64>(playing_buffer.frames - playing_buffer.frames_played,
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num_frames - frames_written)};
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samples_buffer.Push(&input_buffer[frames_written * frame_size],
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frames_available * frame_size);
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frames_written += frames_available;
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playing_buffer.frames_played += frames_available;
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// If that's all the frames in the current buffer, add its samples and mark it as
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// consumed
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if (playing_buffer.frames_played >= playing_buffer.frames) {
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playing_buffer.consumed = true;
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}
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}
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std::memcpy(&last_frame[0], &input_buffer[(frames_written - 1) * frame_size], frame_size_bytes);
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if (queued_buffers <= max_queue_size) {
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Unstall();
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}
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}
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void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames) {
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const std::size_t num_channels = GetDeviceChannels();
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const std::size_t frame_size = num_channels;
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const std::size_t frame_size_bytes = frame_size * sizeof(s16);
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size_t frames_written{0};
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// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
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// paused and we'll desync, so just play silence.
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if (system.IsPaused() || system.IsShuttingDown()) {
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constexpr std::array<s16, 6> silence{};
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for (size_t i = frames_written; i < num_frames; i++) {
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std::memcpy(&output_buffer[i * frame_size], &silence[0], frame_size_bytes);
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}
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return;
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}
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// Due to many frames being queued up with nvdec (5 frames or so?), a lot of buffers also get
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// queued up (30+) but not all at once, which causes constant stalling here, so just let the
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// video play out without attempting to stall.
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// Can hopefully remove this later with a more complete NVDEC implementation.
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const auto nvdec_active{system.AudioCore().IsNVDECActive()};
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// Core timing cannot be paused in single-core mode, so Stall ends up being called over and over
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// and never recovers to a normal state, so just skip attempting to sync things on single-core.
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if (system.IsMulticore() && !nvdec_active && queued_buffers > max_queue_size) {
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Stall();
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} else if (system.IsMulticore() && queued_buffers <= max_queue_size) {
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Unstall();
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}
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while (frames_written < num_frames) {
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// If the playing buffer has been consumed or has no frames, we need a new one
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if (playing_buffer.consumed || playing_buffer.frames == 0) {
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if (!queue.try_dequeue(playing_buffer)) {
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// If no buffer was available we've underrun, fill the remaining buffer with
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// the last written frame and continue.
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for (size_t i = frames_written; i < num_frames; i++) {
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std::memcpy(&output_buffer[i * frame_size], &last_frame[0], frame_size_bytes);
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}
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frames_written = num_frames;
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continue;
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}
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// Successfully dequeued a new buffer.
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queued_buffers--;
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}
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// Get the minimum frames available between the currently playing buffer, and the
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// amount we have left to fill
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size_t frames_available{std::min<u64>(playing_buffer.frames - playing_buffer.frames_played,
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num_frames - frames_written)};
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samples_buffer.Pop(&output_buffer[frames_written * frame_size],
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frames_available * frame_size);
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frames_written += frames_available;
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playing_buffer.frames_played += frames_available;
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// If that's all the frames in the current buffer, add its samples and mark it as
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// consumed
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if (playing_buffer.frames_played >= playing_buffer.frames) {
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playing_buffer.consumed = true;
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}
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}
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std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
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frame_size_bytes);
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if (system.IsMulticore() && queued_buffers <= max_queue_size) {
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Unstall();
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}
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}
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void SinkStream::Stall() {
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std::scoped_lock lk{stall_guard};
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if (stalled_lock) {
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return;
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}
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stalled_lock = system.StallProcesses();
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}
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void SinkStream::Unstall() {
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std::scoped_lock lk{stall_guard};
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if (!stalled_lock) {
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return;
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}
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system.UnstallProcesses();
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stalled_lock.unlock();
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}
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} // namespace AudioCore::Sink
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