Merge pull request #1163 from FearlessTobi/add-audio-stretching
audio_core: Add audio stretching support
This commit is contained in:
commit
926dd41587
21 changed files with 463 additions and 49 deletions
3
.gitmodules
vendored
3
.gitmodules
vendored
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@ -31,3 +31,6 @@
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[submodule "opus"]
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path = externals/opus
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url = https://github.com/ogniK5377/opus.git
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[submodule "soundtouch"]
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path = externals/soundtouch
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url = https://github.com/citra-emu/ext-soundtouch.git
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3
externals/CMakeLists.txt
vendored
3
externals/CMakeLists.txt
vendored
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@ -50,6 +50,9 @@ add_subdirectory(open_source_archives EXCLUDE_FROM_ALL)
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add_library(unicorn-headers INTERFACE)
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target_include_directories(unicorn-headers INTERFACE ./unicorn/include)
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# SoundTouch
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add_subdirectory(soundtouch)
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# Xbyak
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if (ARCHITECTURE_x86_64)
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# Defined before "dynarmic" above
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1
externals/soundtouch
vendored
Submodule
1
externals/soundtouch
vendored
Submodule
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@ -0,0 +1 @@
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Subproject commit 060181eaf273180d3a7e87349895bd0cb6ccbf4a
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@ -17,6 +17,8 @@ add_library(audio_core STATIC
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sink_stream.h
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stream.cpp
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stream.h
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time_stretch.cpp
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time_stretch.h
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$<$<BOOL:${ENABLE_CUBEB}>:cubeb_sink.cpp cubeb_sink.h>
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)
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@ -24,6 +26,7 @@ add_library(audio_core STATIC
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create_target_directory_groups(audio_core)
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target_link_libraries(audio_core PUBLIC common core)
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target_link_libraries(audio_core PRIVATE SoundTouch)
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if(ENABLE_CUBEB)
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target_link_libraries(audio_core PRIVATE cubeb)
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@ -3,27 +3,23 @@
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// Refer to the license.txt file included.
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#include <algorithm>
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#include <atomic>
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#include <cstring>
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#include <mutex>
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#include "audio_core/cubeb_sink.h"
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#include "audio_core/stream.h"
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#include "audio_core/time_stretch.h"
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#include "common/logging/log.h"
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#include "common/ring_buffer.h"
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#include "core/settings.h"
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namespace AudioCore {
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class SinkStreamImpl final : public SinkStream {
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class CubebSinkStream final : public SinkStream {
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public:
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SinkStreamImpl(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
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const std::string& name)
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: ctx{ctx}, num_channels{num_channels_} {
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if (num_channels == 6) {
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// 6-channel audio does not seem to work with cubeb + SDL, so we downsample this to 2
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// channel for now
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is_6_channel = true;
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num_channels = 2;
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}
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CubebSinkStream(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
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const std::string& name)
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: ctx{ctx}, num_channels{std::min(num_channels_, 2u)}, time_stretch{sample_rate,
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num_channels} {
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cubeb_stream_params params{};
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params.rate = sample_rate;
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@ -38,7 +34,7 @@ public:
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if (cubeb_stream_init(ctx, &stream_backend, name.c_str(), nullptr, nullptr, output_device,
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¶ms, std::max(512u, minimum_latency),
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&SinkStreamImpl::DataCallback, &SinkStreamImpl::StateCallback,
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&CubebSinkStream::DataCallback, &CubebSinkStream::StateCallback,
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this) != CUBEB_OK) {
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LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream");
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return;
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@ -50,7 +46,7 @@ public:
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}
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}
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~SinkStreamImpl() {
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~CubebSinkStream() {
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if (!ctx) {
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return;
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}
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@ -62,27 +58,32 @@ public:
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cubeb_stream_destroy(stream_backend);
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}
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void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) override {
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if (!ctx) {
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void EnqueueSamples(u32 source_num_channels, const std::vector<s16>& samples) override {
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if (source_num_channels > num_channels) {
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// Downsample 6 channels to 2
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std::vector<s16> buf;
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buf.reserve(samples.size() * num_channels / source_num_channels);
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for (size_t i = 0; i < samples.size(); i += source_num_channels) {
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for (size_t ch = 0; ch < num_channels; ch++) {
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buf.push_back(samples[i + ch]);
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}
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}
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queue.Push(buf);
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return;
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}
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std::lock_guard lock{queue_mutex};
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queue.Push(samples);
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}
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queue.reserve(queue.size() + samples.size() * GetNumChannels());
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size_t SamplesInQueue(u32 num_channels) const override {
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if (!ctx)
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return 0;
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if (is_6_channel) {
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// Downsample 6 channels to 2
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const size_t sample_count_copy_size = samples.size() * 2;
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queue.reserve(sample_count_copy_size);
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for (size_t i = 0; i < samples.size(); i += num_channels) {
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queue.push_back(samples[i]);
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queue.push_back(samples[i + 1]);
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}
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} else {
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// Copy as-is
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std::copy(samples.begin(), samples.end(), std::back_inserter(queue));
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}
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return queue.Size() / num_channels;
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}
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void Flush() override {
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should_flush = true;
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}
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u32 GetNumChannels() const {
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@ -95,10 +96,11 @@ private:
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cubeb* ctx{};
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cubeb_stream* stream_backend{};
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u32 num_channels{};
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bool is_6_channel{};
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std::mutex queue_mutex;
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std::vector<s16> queue;
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Common::RingBuffer<s16, 0x10000> queue;
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std::array<s16, 2> last_frame;
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std::atomic<bool> should_flush{};
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TimeStretcher time_stretch;
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static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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void* output_buffer, long num_frames);
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@ -144,38 +146,52 @@ CubebSink::~CubebSink() {
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SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
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const std::string& name) {
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sink_streams.push_back(
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std::make_unique<SinkStreamImpl>(ctx, sample_rate, num_channels, output_device, name));
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std::make_unique<CubebSinkStream>(ctx, sample_rate, num_channels, output_device, name));
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return *sink_streams.back();
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}
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long SinkStreamImpl::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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void* output_buffer, long num_frames) {
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SinkStreamImpl* impl = static_cast<SinkStreamImpl*>(user_data);
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long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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void* output_buffer, long num_frames) {
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CubebSinkStream* impl = static_cast<CubebSinkStream*>(user_data);
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u8* buffer = reinterpret_cast<u8*>(output_buffer);
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if (!impl) {
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return {};
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}
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std::lock_guard lock{impl->queue_mutex};
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const size_t num_channels = impl->GetNumChannels();
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const size_t samples_to_write = num_channels * num_frames;
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size_t samples_written;
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const size_t frames_to_write{
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std::min(impl->queue.size() / impl->GetNumChannels(), static_cast<size_t>(num_frames))};
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if (Settings::values.enable_audio_stretching) {
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const std::vector<s16> in{impl->queue.Pop()};
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const size_t num_in{in.size() / num_channels};
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s16* const out{reinterpret_cast<s16*>(buffer)};
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const size_t out_frames = impl->time_stretch.Process(in.data(), num_in, out, num_frames);
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samples_written = out_frames * num_channels;
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memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * impl->GetNumChannels());
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impl->queue.erase(impl->queue.begin(),
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impl->queue.begin() + frames_to_write * impl->GetNumChannels());
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if (impl->should_flush) {
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impl->time_stretch.Flush();
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impl->should_flush = false;
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}
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} else {
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samples_written = impl->queue.Pop(buffer, samples_to_write);
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}
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if (frames_to_write < num_frames) {
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// Fill the rest of the frames with silence
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memset(buffer + frames_to_write * sizeof(s16) * impl->GetNumChannels(), 0,
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(num_frames - frames_to_write) * sizeof(s16) * impl->GetNumChannels());
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if (samples_written >= num_channels) {
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std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),
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num_channels * sizeof(s16));
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}
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// Fill the rest of the frames with last_frame
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for (size_t i = samples_written; i < samples_to_write; i += num_channels) {
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std::memcpy(buffer + i * sizeof(s16), &impl->last_frame[0], num_channels * sizeof(s16));
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}
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return num_frames;
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}
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void SinkStreamImpl::StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state) {}
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void CubebSinkStream::StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state) {}
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std::vector<std::string> ListCubebSinkDevices() {
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std::vector<std::string> device_list;
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@ -21,6 +21,12 @@ public:
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private:
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struct NullSinkStreamImpl final : SinkStream {
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void EnqueueSamples(u32 /*num_channels*/, const std::vector<s16>& /*samples*/) override {}
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size_t SamplesInQueue(u32 /*num_channels*/) const override {
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return 0;
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}
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void Flush() override {}
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} null_sink_stream;
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};
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@ -25,6 +25,10 @@ public:
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* @param samples Samples in interleaved stereo PCM16 format.
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*/
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virtual void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) = 0;
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virtual std::size_t SamplesInQueue(u32 num_channels) const = 0;
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virtual void Flush() = 0;
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};
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using SinkStreamPtr = std::unique_ptr<SinkStream>;
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@ -73,6 +73,7 @@ static void VolumeAdjustSamples(std::vector<s16>& samples) {
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void Stream::PlayNextBuffer() {
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if (!IsPlaying()) {
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// Ensure we are in playing state before playing the next buffer
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sink_stream.Flush();
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return;
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}
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@ -83,6 +84,7 @@ void Stream::PlayNextBuffer() {
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if (queued_buffers.empty()) {
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// No queued buffers - we are effectively paused
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sink_stream.Flush();
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return;
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}
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@ -90,6 +92,7 @@ void Stream::PlayNextBuffer() {
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queued_buffers.pop();
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VolumeAdjustSamples(active_buffer->Samples());
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sink_stream.EnqueueSamples(GetNumChannels(), active_buffer->GetSamples());
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CoreTiming::ScheduleEventThreadsafe(GetBufferReleaseCycles(*active_buffer), release_event, {});
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68
src/audio_core/time_stretch.cpp
Normal file
68
src/audio_core/time_stretch.cpp
Normal file
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@ -0,0 +1,68 @@
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// Copyright 2018 yuzu Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <algorithm>
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#include <cmath>
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#include <cstddef>
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#include "audio_core/time_stretch.h"
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#include "common/logging/log.h"
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namespace AudioCore {
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TimeStretcher::TimeStretcher(u32 sample_rate, u32 channel_count)
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: m_sample_rate(sample_rate), m_channel_count(channel_count) {
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m_sound_touch.setChannels(channel_count);
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m_sound_touch.setSampleRate(sample_rate);
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m_sound_touch.setPitch(1.0);
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m_sound_touch.setTempo(1.0);
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}
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void TimeStretcher::Clear() {
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m_sound_touch.clear();
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}
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void TimeStretcher::Flush() {
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m_sound_touch.flush();
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}
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size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t num_out) {
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const double time_delta = static_cast<double>(num_out) / m_sample_rate; // seconds
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// We were given actual_samples number of samples, and num_samples were requested from us.
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double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
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const double max_latency = 1.0; // seconds
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const double max_backlog = m_sample_rate * max_latency;
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const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
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if (backlog_fullness > 5.0) {
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// Too many samples in backlog: Don't push anymore on
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num_in = 0;
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}
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// We ideally want the backlog to be about 50% full.
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// This gives some headroom both ways to prevent underflow and overflow.
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// We tweak current_ratio to encourage this.
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constexpr double tweak_time_scale = 0.05; // seconds
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const double tweak_correction = (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
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current_ratio *= std::pow(1.0 + 2.0 * tweak_correction, tweak_correction < 0 ? 3.0 : 1.0);
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// This low-pass filter smoothes out variance in the calculated stretch ratio.
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// The time-scale determines how responsive this filter is.
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constexpr double lpf_time_scale = 2.0; // seconds
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const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
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m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
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// Place a lower limit of 5% speed. When a game boots up, there will be
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// many silence samples. These do not need to be timestretched.
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m_stretch_ratio = std::max(m_stretch_ratio, 0.05);
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m_sound_touch.setTempo(m_stretch_ratio);
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LOG_DEBUG(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, m_stretch_ratio,
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backlog_fullness);
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m_sound_touch.putSamples(in, num_in);
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return m_sound_touch.receiveSamples(out, num_out);
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}
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} // namespace AudioCore
|
36
src/audio_core/time_stretch.h
Normal file
36
src/audio_core/time_stretch.h
Normal file
|
@ -0,0 +1,36 @@
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// Copyright 2018 yuzu Emulator Project
|
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#pragma once
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#include <array>
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#include <cstddef>
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#include <SoundTouch.h>
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#include "common/common_types.h"
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namespace AudioCore {
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class TimeStretcher {
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public:
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TimeStretcher(u32 sample_rate, u32 channel_count);
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/// @param in Input sample buffer
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/// @param num_in Number of input frames in `in`
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/// @param out Output sample buffer
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/// @param num_out Desired number of output frames in `out`
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/// @returns Actual number of frames written to `out`
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size_t Process(const s16* in, size_t num_in, s16* out, size_t num_out);
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void Clear();
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void Flush();
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private:
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u32 m_sample_rate;
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u32 m_channel_count;
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soundtouch::SoundTouch m_sound_touch;
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double m_stretch_ratio = 1.0;
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};
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||||
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} // namespace AudioCore
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|
@ -71,6 +71,7 @@ add_library(common STATIC
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param_package.cpp
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||||
param_package.h
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||||
quaternion.h
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||||
ring_buffer.h
|
||||
scm_rev.cpp
|
||||
scm_rev.h
|
||||
scope_exit.h
|
||||
|
|
111
src/common/ring_buffer.h
Normal file
111
src/common/ring_buffer.h
Normal file
|
@ -0,0 +1,111 @@
|
|||
// Copyright 2018 yuzu emulator team
|
||||
// Licensed under GPLv2 or any later version
|
||||
// Refer to the license.txt file included.
|
||||
|
||||
#pragma once
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#include <algorithm>
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#include <array>
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#include <atomic>
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#include <cstddef>
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#include <cstring>
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#include <type_traits>
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#include <vector>
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#include "common/common_types.h"
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namespace Common {
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/// SPSC ring buffer
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/// @tparam T Element type
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||||
/// @tparam capacity Number of slots in ring buffer
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||||
/// @tparam granularity Slot size in terms of number of elements
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||||
template <typename T, size_t capacity, size_t granularity = 1>
|
||||
class RingBuffer {
|
||||
/// A "slot" is made of `granularity` elements of `T`.
|
||||
static constexpr size_t slot_size = granularity * sizeof(T);
|
||||
// T must be safely memcpy-able and have a trivial default constructor.
|
||||
static_assert(std::is_trivial_v<T>);
|
||||
// Ensure capacity is sensible.
|
||||
static_assert(capacity < std::numeric_limits<size_t>::max() / 2 / granularity);
|
||||
static_assert((capacity & (capacity - 1)) == 0, "capacity must be a power of two");
|
||||
// Ensure lock-free.
|
||||
static_assert(std::atomic<size_t>::is_always_lock_free);
|
||||
|
||||
public:
|
||||
/// Pushes slots into the ring buffer
|
||||
/// @param new_slots Pointer to the slots to push
|
||||
/// @param slot_count Number of slots to push
|
||||
/// @returns The number of slots actually pushed
|
||||
size_t Push(const void* new_slots, size_t slot_count) {
|
||||
const size_t write_index = m_write_index.load();
|
||||
const size_t slots_free = capacity + m_read_index.load() - write_index;
|
||||
const size_t push_count = std::min(slot_count, slots_free);
|
||||
|
||||
const size_t pos = write_index % capacity;
|
||||
const size_t first_copy = std::min(capacity - pos, push_count);
|
||||
const size_t second_copy = push_count - first_copy;
|
||||
|
||||
const char* in = static_cast<const char*>(new_slots);
|
||||
std::memcpy(m_data.data() + pos * granularity, in, first_copy * slot_size);
|
||||
in += first_copy * slot_size;
|
||||
std::memcpy(m_data.data(), in, second_copy * slot_size);
|
||||
|
||||
m_write_index.store(write_index + push_count);
|
||||
|
||||
return push_count;
|
||||
}
|
||||
|
||||
size_t Push(const std::vector<T>& input) {
|
||||
return Push(input.data(), input.size());
|
||||
}
|
||||
|
||||
/// Pops slots from the ring buffer
|
||||
/// @param output Where to store the popped slots
|
||||
/// @param max_slots Maximum number of slots to pop
|
||||
/// @returns The number of slots actually popped
|
||||
size_t Pop(void* output, size_t max_slots = ~size_t(0)) {
|
||||
const size_t read_index = m_read_index.load();
|
||||
const size_t slots_filled = m_write_index.load() - read_index;
|
||||
const size_t pop_count = std::min(slots_filled, max_slots);
|
||||
|
||||
const size_t pos = read_index % capacity;
|
||||
const size_t first_copy = std::min(capacity - pos, pop_count);
|
||||
const size_t second_copy = pop_count - first_copy;
|
||||
|
||||
char* out = static_cast<char*>(output);
|
||||
std::memcpy(out, m_data.data() + pos * granularity, first_copy * slot_size);
|
||||
out += first_copy * slot_size;
|
||||
std::memcpy(out, m_data.data(), second_copy * slot_size);
|
||||
|
||||
m_read_index.store(read_index + pop_count);
|
||||
|
||||
return pop_count;
|
||||
}
|
||||
|
||||
std::vector<T> Pop(size_t max_slots = ~size_t(0)) {
|
||||
std::vector<T> out(std::min(max_slots, capacity) * granularity);
|
||||
const size_t count = Pop(out.data(), out.size() / granularity);
|
||||
out.resize(count * granularity);
|
||||
return out;
|
||||
}
|
||||
|
||||
/// @returns Number of slots used
|
||||
size_t Size() const {
|
||||
return m_write_index.load() - m_read_index.load();
|
||||
}
|
||||
|
||||
/// @returns Maximum size of ring buffer
|
||||
constexpr size_t Capacity() const {
|
||||
return capacity;
|
||||
}
|
||||
|
||||
private:
|
||||
// It is important to align the below variables for performance reasons:
|
||||
// Having them on the same cache-line would result in false-sharing between them.
|
||||
alignas(128) std::atomic<size_t> m_read_index{0};
|
||||
alignas(128) std::atomic<size_t> m_write_index{0};
|
||||
|
||||
std::array<T, granularity * capacity> m_data;
|
||||
};
|
||||
|
||||
} // namespace Common
|
|
@ -148,6 +148,7 @@ struct Values {
|
|||
|
||||
// Audio
|
||||
std::string sink_id;
|
||||
bool enable_audio_stretching;
|
||||
std::string audio_device_id;
|
||||
float volume;
|
||||
|
||||
|
|
|
@ -120,6 +120,9 @@ TelemetrySession::TelemetrySession() {
|
|||
Telemetry::AppendOSInfo(field_collection);
|
||||
|
||||
// Log user configuration information
|
||||
AddField(Telemetry::FieldType::UserConfig, "Audio_SinkId", Settings::values.sink_id);
|
||||
AddField(Telemetry::FieldType::UserConfig, "Audio_EnableAudioStretching",
|
||||
Settings::values.enable_audio_stretching);
|
||||
AddField(Telemetry::FieldType::UserConfig, "Core_UseCpuJit", Settings::values.use_cpu_jit);
|
||||
AddField(Telemetry::FieldType::UserConfig, "Core_UseMultiCore",
|
||||
Settings::values.use_multi_core);
|
||||
|
|
|
@ -1,5 +1,6 @@
|
|||
add_executable(tests
|
||||
common/param_package.cpp
|
||||
common/ring_buffer.cpp
|
||||
core/arm/arm_test_common.cpp
|
||||
core/arm/arm_test_common.h
|
||||
core/core_timing.cpp
|
||||
|
|
130
src/tests/common/ring_buffer.cpp
Normal file
130
src/tests/common/ring_buffer.cpp
Normal file
|
@ -0,0 +1,130 @@
|
|||
// Copyright 2018 yuzu emulator team
|
||||
// Licensed under GPLv2 or any later version
|
||||
// Refer to the license.txt file included.
|
||||
|
||||
#include <algorithm>
|
||||
#include <array>
|
||||
#include <cstddef>
|
||||
#include <numeric>
|
||||
#include <thread>
|
||||
#include <vector>
|
||||
#include <catch2/catch.hpp>
|
||||
#include "common/ring_buffer.h"
|
||||
|
||||
namespace Common {
|
||||
|
||||
TEST_CASE("RingBuffer: Basic Tests", "[common]") {
|
||||
RingBuffer<char, 4, 1> buf;
|
||||
|
||||
// Pushing values into a ring buffer with space should succeed.
|
||||
for (size_t i = 0; i < 4; i++) {
|
||||
const char elem = static_cast<char>(i);
|
||||
const size_t count = buf.Push(&elem, 1);
|
||||
REQUIRE(count == 1);
|
||||
}
|
||||
|
||||
REQUIRE(buf.Size() == 4);
|
||||
|
||||
// Pushing values into a full ring buffer should fail.
|
||||
{
|
||||
const char elem = static_cast<char>(42);
|
||||
const size_t count = buf.Push(&elem, 1);
|
||||
REQUIRE(count == 0);
|
||||
}
|
||||
|
||||
REQUIRE(buf.Size() == 4);
|
||||
|
||||
// Popping multiple values from a ring buffer with values should succeed.
|
||||
{
|
||||
const std::vector<char> popped = buf.Pop(2);
|
||||
REQUIRE(popped.size() == 2);
|
||||
REQUIRE(popped[0] == 0);
|
||||
REQUIRE(popped[1] == 1);
|
||||
}
|
||||
|
||||
REQUIRE(buf.Size() == 2);
|
||||
|
||||
// Popping a single value from a ring buffer with values should succeed.
|
||||
{
|
||||
const std::vector<char> popped = buf.Pop(1);
|
||||
REQUIRE(popped.size() == 1);
|
||||
REQUIRE(popped[0] == 2);
|
||||
}
|
||||
|
||||
REQUIRE(buf.Size() == 1);
|
||||
|
||||
// Pushing more values than space available should partially suceed.
|
||||
{
|
||||
std::vector<char> to_push(6);
|
||||
std::iota(to_push.begin(), to_push.end(), 88);
|
||||
const size_t count = buf.Push(to_push);
|
||||
REQUIRE(count == 3);
|
||||
}
|
||||
|
||||
REQUIRE(buf.Size() == 4);
|
||||
|
||||
// Doing an unlimited pop should pop all values.
|
||||
{
|
||||
const std::vector<char> popped = buf.Pop();
|
||||
REQUIRE(popped.size() == 4);
|
||||
REQUIRE(popped[0] == 3);
|
||||
REQUIRE(popped[1] == 88);
|
||||
REQUIRE(popped[2] == 89);
|
||||
REQUIRE(popped[3] == 90);
|
||||
}
|
||||
|
||||
REQUIRE(buf.Size() == 0);
|
||||
}
|
||||
|
||||
TEST_CASE("RingBuffer: Threaded Test", "[common]") {
|
||||
RingBuffer<char, 4, 2> buf;
|
||||
const char seed = 42;
|
||||
const size_t count = 1000000;
|
||||
size_t full = 0;
|
||||
size_t empty = 0;
|
||||
|
||||
const auto next_value = [](std::array<char, 2>& value) {
|
||||
value[0] += 1;
|
||||
value[1] += 2;
|
||||
};
|
||||
|
||||
std::thread producer{[&] {
|
||||
std::array<char, 2> value = {seed, seed};
|
||||
size_t i = 0;
|
||||
while (i < count) {
|
||||
if (const size_t c = buf.Push(&value[0], 1); c > 0) {
|
||||
REQUIRE(c == 1);
|
||||
i++;
|
||||
next_value(value);
|
||||
} else {
|
||||
full++;
|
||||
std::this_thread::yield();
|
||||
}
|
||||
}
|
||||
}};
|
||||
|
||||
std::thread consumer{[&] {
|
||||
std::array<char, 2> value = {seed, seed};
|
||||
size_t i = 0;
|
||||
while (i < count) {
|
||||
if (const std::vector<char> v = buf.Pop(1); v.size() > 0) {
|
||||
REQUIRE(v.size() == 2);
|
||||
REQUIRE(v[0] == value[0]);
|
||||
REQUIRE(v[1] == value[1]);
|
||||
i++;
|
||||
next_value(value);
|
||||
} else {
|
||||
empty++;
|
||||
std::this_thread::yield();
|
||||
}
|
||||
}
|
||||
}};
|
||||
|
||||
producer.join();
|
||||
consumer.join();
|
||||
|
||||
REQUIRE(buf.Size() == 0);
|
||||
printf("RingBuffer: Threaded Test: full: %zu, empty: %zu\n", full, empty);
|
||||
}
|
||||
|
||||
} // namespace Common
|
|
@ -95,6 +95,8 @@ void Config::ReadValues() {
|
|||
|
||||
qt_config->beginGroup("Audio");
|
||||
Settings::values.sink_id = qt_config->value("output_engine", "auto").toString().toStdString();
|
||||
Settings::values.enable_audio_stretching =
|
||||
qt_config->value("enable_audio_stretching", true).toBool();
|
||||
Settings::values.audio_device_id =
|
||||
qt_config->value("output_device", "auto").toString().toStdString();
|
||||
Settings::values.volume = qt_config->value("volume", 1).toFloat();
|
||||
|
@ -230,6 +232,7 @@ void Config::SaveValues() {
|
|||
|
||||
qt_config->beginGroup("Audio");
|
||||
qt_config->setValue("output_engine", QString::fromStdString(Settings::values.sink_id));
|
||||
qt_config->setValue("enable_audio_stretching", Settings::values.enable_audio_stretching);
|
||||
qt_config->setValue("output_device", QString::fromStdString(Settings::values.audio_device_id));
|
||||
qt_config->setValue("volume", Settings::values.volume);
|
||||
qt_config->endGroup();
|
||||
|
|
|
@ -46,6 +46,8 @@ void ConfigureAudio::setConfiguration() {
|
|||
}
|
||||
ui->output_sink_combo_box->setCurrentIndex(new_sink_index);
|
||||
|
||||
ui->toggle_audio_stretching->setChecked(Settings::values.enable_audio_stretching);
|
||||
|
||||
// The device list cannot be pre-populated (nor listed) until the output sink is known.
|
||||
updateAudioDevices(new_sink_index);
|
||||
|
||||
|
@ -67,6 +69,7 @@ void ConfigureAudio::applyConfiguration() {
|
|||
Settings::values.sink_id =
|
||||
ui->output_sink_combo_box->itemText(ui->output_sink_combo_box->currentIndex())
|
||||
.toStdString();
|
||||
Settings::values.enable_audio_stretching = ui->toggle_audio_stretching->isChecked();
|
||||
Settings::values.audio_device_id =
|
||||
ui->audio_device_combo_box->itemText(ui->audio_device_combo_box->currentIndex())
|
||||
.toStdString();
|
||||
|
|
|
@ -31,6 +31,16 @@
|
|||
</item>
|
||||
</layout>
|
||||
</item>
|
||||
<item>
|
||||
<widget class="QCheckBox" name="toggle_audio_stretching">
|
||||
<property name="toolTip">
|
||||
<string>This post-processing effect adjusts audio speed to match emulation speed and helps prevent audio stutter. This however increases audio latency.</string>
|
||||
</property>
|
||||
<property name="text">
|
||||
<string>Enable audio stretching</string>
|
||||
</property>
|
||||
</widget>
|
||||
</item>
|
||||
<item>
|
||||
<layout class="QHBoxLayout">
|
||||
<item>
|
||||
|
|
|
@ -108,6 +108,8 @@ void Config::ReadValues() {
|
|||
|
||||
// Audio
|
||||
Settings::values.sink_id = sdl2_config->Get("Audio", "output_engine", "auto");
|
||||
Settings::values.enable_audio_stretching =
|
||||
sdl2_config->GetBoolean("Audio", "enable_audio_stretching", true);
|
||||
Settings::values.audio_device_id = sdl2_config->Get("Audio", "output_device", "auto");
|
||||
Settings::values.volume = sdl2_config->GetReal("Audio", "volume", 1);
|
||||
|
||||
|
|
|
@ -150,6 +150,12 @@ swap_screen =
|
|||
# auto (default): Auto-select, null: No audio output, cubeb: Cubeb audio engine (if available)
|
||||
output_engine =
|
||||
|
||||
# Whether or not to enable the audio-stretching post-processing effect.
|
||||
# This effect adjusts audio speed to match emulation speed and helps prevent audio stutter,
|
||||
# at the cost of increasing audio latency.
|
||||
# 0: No, 1 (default): Yes
|
||||
enable_audio_stretching =
|
||||
|
||||
# Which audio device to use.
|
||||
# auto (default): Auto-select
|
||||
output_device =
|
||||
|
|
Loading…
Reference in a new issue