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-rw-r--r--src/audio_core/sink/sink_stream.cpp279
1 files changed, 279 insertions, 0 deletions
diff --git a/src/audio_core/sink/sink_stream.cpp b/src/audio_core/sink/sink_stream.cpp
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index 000000000..37fe725e4
--- /dev/null
+++ b/src/audio_core/sink/sink_stream.cpp
@@ -0,0 +1,279 @@
+// 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(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()};
+ if (!nvdec_active && 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, 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(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 (stalled && queued_buffers <= max_queue_size) {
+ Unstall();
+ }
+}
+
+void SinkStream::Stall() {
+ if (stalled) {
+ return;
+ }
+ stalled = true;
+ system.StallProcesses();
+}
+
+void SinkStream::Unstall() {
+ if (!stalled) {
+ return;
+ }
+ system.UnstallProcesses();
+ stalled = false;
+}
+
+} // namespace AudioCore::Sink