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audio_core: Simplify sink interface
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761ef78408
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f34711219a
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@ -13,13 +13,11 @@ namespace AudioCore {
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struct CubebSink::Impl {
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unsigned int sample_rate = 0;
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std::vector<std::string> device_list;
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cubeb* ctx = nullptr;
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cubeb_stream* stream = nullptr;
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std::mutex queue_mutex;
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std::vector<s16> queue;
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std::function<void(s16*, std::size_t)> cb;
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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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@ -95,45 +93,19 @@ unsigned int CubebSink::GetNativeSampleRate() const {
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return impl->sample_rate;
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}
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void CubebSink::EnqueueSamples(const s16* samples, std::size_t sample_count) {
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if (!impl->ctx)
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return;
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std::lock_guard lock{impl->queue_mutex};
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impl->queue.reserve(impl->queue.size() + sample_count * 2);
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std::copy(samples, samples + sample_count * 2, std::back_inserter(impl->queue));
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}
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size_t CubebSink::SamplesInQueue() const {
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if (!impl->ctx)
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return 0;
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std::lock_guard lock{impl->queue_mutex};
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return impl->queue.size() / 2;
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void CubebSink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
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impl->cb = cb;
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}
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long CubebSink::Impl::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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Impl* impl = static_cast<Impl*>(user_data);
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u8* buffer = reinterpret_cast<u8*>(output_buffer);
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s16* buffer = reinterpret_cast<s16*>(output_buffer);
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if (!impl)
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if (!impl || !impl->cb)
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return 0;
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std::lock_guard lock{impl->queue_mutex};
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std::size_t frames_to_write =
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std::min(impl->queue.size() / 2, static_cast<std::size_t>(num_frames));
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memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * 2);
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impl->queue.erase(impl->queue.begin(), impl->queue.begin() + frames_to_write * 2);
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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) * 2, 0,
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(num_frames - frames_to_write) * sizeof(s16) * 2);
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}
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impl->cb(buffer, num_frames);
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return num_frames;
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}
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@ -17,9 +17,7 @@ public:
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unsigned int GetNativeSampleRate() const override;
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void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
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std::size_t SamplesInQueue() const override;
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void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
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private:
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struct Impl;
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@ -12,16 +12,13 @@
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namespace AudioCore {
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DspInterface::DspInterface() = default;
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DspInterface::~DspInterface() {
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if (perform_time_stretching) {
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FlushResidualStretcherAudio();
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}
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}
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DspInterface::~DspInterface() = default;
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void DspInterface::SetSink(const std::string& sink_id, const std::string& audio_device) {
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const SinkDetails& sink_details = GetSinkDetails(sink_id);
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sink = sink_details.factory(audio_device);
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sink->SetCallback(
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[this](s16* buffer, std::size_t num_frames) { OutputCallback(buffer, num_frames); });
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time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
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}
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@ -51,32 +48,21 @@ void DspInterface::OutputFrame(StereoFrame16& frame) {
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frame[i][1] = static_cast<s16>(frame[i][1] * volume_scale_factor);
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}
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if (perform_time_stretching) {
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time_stretcher.AddSamples(&frame[0][0], frame.size());
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std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
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sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
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} else {
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constexpr std::size_t maximum_sample_latency = 2048; // about 64 miliseconds
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if (sink->SamplesInQueue() > maximum_sample_latency) {
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// This can occur if we're running too fast and samples are starting to back up.
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// Just drop the samples.
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return;
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}
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sink->EnqueueSamples(&frame[0][0], frame.size());
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}
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fifo.Push(frame.data(), frame.size());
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}
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void DspInterface::FlushResidualStretcherAudio() {
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if (!sink)
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return;
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void DspInterface::FlushResidualStretcherAudio() {}
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time_stretcher.Flush();
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while (true) {
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std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
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if (residual_audio.empty())
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break;
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sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
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void DspInterface::OutputCallback(s16* buffer, size_t num_frames) {
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const size_t frames_written = fifo.Pop(buffer, num_frames);
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if (frames_written > 0) {
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std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16));
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}
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// Hold last emitted frame; this prevents popping.
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for (size_t i = frames_written; i < num_frames; i++) {
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std::memcpy(buffer + 2 * i, &last_frame[0], 2 * sizeof(s16));
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}
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}
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@ -9,6 +9,7 @@
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#include "audio_core/audio_types.h"
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#include "audio_core/time_stretch.h"
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#include "common/common_types.h"
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#include "common/ring_buffer.h"
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#include "core/memory.h"
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namespace Service {
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@ -81,9 +82,12 @@ protected:
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private:
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void FlushResidualStretcherAudio();
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void OutputCallback(s16* buffer, std::size_t num_frames);
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std::unique_ptr<Sink> sink;
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bool perform_time_stretching = false;
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Common::RingBuffer<s16, 0x2000, 2> fifo;
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std::array<s16, 2> last_frame{};
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TimeStretcher time_stretcher;
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};
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@ -19,11 +19,7 @@ public:
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return native_sample_rate;
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}
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void EnqueueSamples(const s16*, std::size_t) override {}
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std::size_t SamplesInQueue() const override {
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return 0;
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}
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void SetCallback(std::function<void(s16*, std::size_t)>) override {}
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};
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} // namespace AudioCore
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@ -2,8 +2,8 @@
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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 <list>
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#include <numeric>
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#include <string>
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#include <vector>
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#include <SDL.h>
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#include "audio_core/audio_types.h"
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#include "audio_core/sdl2_sink.h"
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@ -17,7 +17,7 @@ struct SDL2Sink::Impl {
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SDL_AudioDeviceID audio_device_id = 0;
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std::list<std::vector<s16>> queue;
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std::function<void(s16*, std::size_t)> cb;
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static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes);
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};
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@ -74,58 +74,18 @@ unsigned int SDL2Sink::GetNativeSampleRate() const {
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return impl->sample_rate;
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}
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void SDL2Sink::EnqueueSamples(const s16* samples, std::size_t sample_count) {
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if (impl->audio_device_id <= 0)
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return;
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SDL_LockAudioDevice(impl->audio_device_id);
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impl->queue.emplace_back(samples, samples + sample_count * 2);
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SDL_UnlockAudioDevice(impl->audio_device_id);
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}
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size_t SDL2Sink::SamplesInQueue() const {
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if (impl->audio_device_id <= 0)
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return 0;
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SDL_LockAudioDevice(impl->audio_device_id);
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std::size_t total_size =
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std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<std::size_t>(0),
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[](std::size_t sum, const auto& buffer) {
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// Division by two because each stereo sample is made of
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// two s16.
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return sum + buffer.size() / 2;
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});
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SDL_UnlockAudioDevice(impl->audio_device_id);
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return total_size;
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void SDL2Sink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
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impl->cb = cb;
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}
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void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) {
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Impl* impl = reinterpret_cast<Impl*>(impl_);
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if (!impl || !impl->cb)
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return;
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std::size_t remaining_size = static_cast<std::size_t>(buffer_size_in_bytes) /
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sizeof(s16); // Keep track of size in 16-bit increments.
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const size_t num_frames = buffer_size_in_bytes / (2 * sizeof(s16));
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while (remaining_size > 0 && !impl->queue.empty()) {
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if (impl->queue.front().size() <= remaining_size) {
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memcpy(buffer, impl->queue.front().data(), impl->queue.front().size() * sizeof(s16));
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buffer += impl->queue.front().size() * sizeof(s16);
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remaining_size -= impl->queue.front().size();
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impl->queue.pop_front();
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} else {
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memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16));
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buffer += remaining_size * sizeof(s16);
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impl->queue.front().erase(impl->queue.front().begin(),
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impl->queue.front().begin() + remaining_size);
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remaining_size = 0;
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}
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}
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if (remaining_size > 0) {
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memset(buffer, 0, remaining_size * sizeof(s16));
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}
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impl->cb(reinterpret_cast<s16*>(buffer), num_frames);
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}
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std::vector<std::string> ListSDL2SinkDevices() {
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@ -17,9 +17,7 @@ public:
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unsigned int GetNativeSampleRate() const override;
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void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
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std::size_t SamplesInQueue() const override;
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void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
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private:
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struct Impl;
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@ -4,7 +4,7 @@
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#pragma once
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#include <vector>
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#include <functional>
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#include "common/common_types.h"
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namespace AudioCore {
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@ -20,19 +20,16 @@ class Sink {
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public:
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virtual ~Sink() = default;
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/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units:
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/// samples/sec)
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/// The native rate of this sink. The sink expects to be fed samples that respect this.
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/// (Units: samples/sec)
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virtual unsigned int GetNativeSampleRate() const = 0;
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/**
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* Feed stereo samples to sink.
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* Set callback for samples
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* @param samples Samples in interleaved stereo PCM16 format.
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* @param sample_count Number of samples.
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*/
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virtual void EnqueueSamples(const s16* samples, std::size_t sample_count) = 0;
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/// Samples enqueued that have not been played yet.
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virtual std::size_t SamplesInQueue() const = 0;
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virtual void SetCallback(std::function<void(s16*, std::size_t)> cb) = 0;
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};
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} // namespace AudioCore
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