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	audio_core: Simplify sink interface
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					 8 changed files with 42 additions and 131 deletions
				
			
		|  | @ -13,13 +13,11 @@ namespace AudioCore { | |||
| 
 | ||||
| struct CubebSink::Impl { | ||||
|     unsigned int sample_rate = 0; | ||||
|     std::vector<std::string> device_list; | ||||
| 
 | ||||
|     cubeb* ctx = nullptr; | ||||
|     cubeb_stream* stream = nullptr; | ||||
| 
 | ||||
|     std::mutex queue_mutex; | ||||
|     std::vector<s16> queue; | ||||
|     std::function<void(s16*, std::size_t)> cb; | ||||
| 
 | ||||
|     static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer, | ||||
|                              void* output_buffer, long num_frames); | ||||
|  | @ -95,45 +93,19 @@ unsigned int CubebSink::GetNativeSampleRate() const { | |||
|     return impl->sample_rate; | ||||
| } | ||||
| 
 | ||||
| void CubebSink::EnqueueSamples(const s16* samples, std::size_t sample_count) { | ||||
|     if (!impl->ctx) | ||||
|         return; | ||||
| 
 | ||||
|     std::lock_guard lock{impl->queue_mutex}; | ||||
| 
 | ||||
|     impl->queue.reserve(impl->queue.size() + sample_count * 2); | ||||
|     std::copy(samples, samples + sample_count * 2, std::back_inserter(impl->queue)); | ||||
| } | ||||
| 
 | ||||
| size_t CubebSink::SamplesInQueue() const { | ||||
|     if (!impl->ctx) | ||||
|         return 0; | ||||
| 
 | ||||
|     std::lock_guard lock{impl->queue_mutex}; | ||||
|     return impl->queue.size() / 2; | ||||
| void CubebSink::SetCallback(std::function<void(s16*, std::size_t)> cb) { | ||||
|     impl->cb = cb; | ||||
| } | ||||
| 
 | ||||
| long CubebSink::Impl::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer, | ||||
|                                    void* output_buffer, long num_frames) { | ||||
|     Impl* impl = static_cast<Impl*>(user_data); | ||||
|     u8* buffer = reinterpret_cast<u8*>(output_buffer); | ||||
|     s16* buffer = reinterpret_cast<s16*>(output_buffer); | ||||
| 
 | ||||
|     if (!impl) | ||||
|     if (!impl || !impl->cb) | ||||
|         return 0; | ||||
| 
 | ||||
|     std::lock_guard lock{impl->queue_mutex}; | ||||
| 
 | ||||
|     std::size_t frames_to_write = | ||||
|         std::min(impl->queue.size() / 2, static_cast<std::size_t>(num_frames)); | ||||
| 
 | ||||
|     memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * 2); | ||||
|     impl->queue.erase(impl->queue.begin(), impl->queue.begin() + frames_to_write * 2); | ||||
| 
 | ||||
|     if (frames_to_write < num_frames) { | ||||
|         // Fill the rest of the frames with silence
 | ||||
|         memset(buffer + frames_to_write * sizeof(s16) * 2, 0, | ||||
|                (num_frames - frames_to_write) * sizeof(s16) * 2); | ||||
|     } | ||||
|     impl->cb(buffer, num_frames); | ||||
| 
 | ||||
|     return num_frames; | ||||
| } | ||||
|  |  | |||
|  | @ -17,9 +17,7 @@ public: | |||
| 
 | ||||
|     unsigned int GetNativeSampleRate() const override; | ||||
| 
 | ||||
|     void EnqueueSamples(const s16* samples, std::size_t sample_count) override; | ||||
| 
 | ||||
|     std::size_t SamplesInQueue() const override; | ||||
|     void SetCallback(std::function<void(s16*, std::size_t)> cb) override; | ||||
| 
 | ||||
| private: | ||||
|     struct Impl; | ||||
|  |  | |||
|  | @ -12,16 +12,13 @@ | |||
| namespace AudioCore { | ||||
| 
 | ||||
| DspInterface::DspInterface() = default; | ||||
| 
 | ||||
| DspInterface::~DspInterface() { | ||||
|     if (perform_time_stretching) { | ||||
|         FlushResidualStretcherAudio(); | ||||
|     } | ||||
| } | ||||
| DspInterface::~DspInterface() = default; | ||||
| 
 | ||||
| void DspInterface::SetSink(const std::string& sink_id, const std::string& audio_device) { | ||||
|     const SinkDetails& sink_details = GetSinkDetails(sink_id); | ||||
|     sink = sink_details.factory(audio_device); | ||||
|     sink->SetCallback( | ||||
|         [this](s16* buffer, std::size_t num_frames) { OutputCallback(buffer, num_frames); }); | ||||
|     time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate()); | ||||
| } | ||||
| 
 | ||||
|  | @ -51,32 +48,21 @@ void DspInterface::OutputFrame(StereoFrame16& frame) { | |||
|         frame[i][1] = static_cast<s16>(frame[i][1] * volume_scale_factor); | ||||
|     } | ||||
| 
 | ||||
|     if (perform_time_stretching) { | ||||
|         time_stretcher.AddSamples(&frame[0][0], frame.size()); | ||||
|         std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue()); | ||||
|         sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2); | ||||
|     } else { | ||||
|         constexpr std::size_t maximum_sample_latency = 2048; // about 64 miliseconds
 | ||||
|         if (sink->SamplesInQueue() > maximum_sample_latency) { | ||||
|             // This can occur if we're running too fast and samples are starting to back up.
 | ||||
|             // Just drop the samples.
 | ||||
|             return; | ||||
|         } | ||||
| 
 | ||||
|         sink->EnqueueSamples(&frame[0][0], frame.size()); | ||||
|     } | ||||
|     fifo.Push(frame.data(), frame.size()); | ||||
| } | ||||
| 
 | ||||
| void DspInterface::FlushResidualStretcherAudio() { | ||||
|     if (!sink) | ||||
|         return; | ||||
| void DspInterface::FlushResidualStretcherAudio() {} | ||||
| 
 | ||||
|     time_stretcher.Flush(); | ||||
|     while (true) { | ||||
|         std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue()); | ||||
|         if (residual_audio.empty()) | ||||
|             break; | ||||
|         sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2); | ||||
| void DspInterface::OutputCallback(s16* buffer, size_t num_frames) { | ||||
|     const size_t frames_written = fifo.Pop(buffer, num_frames); | ||||
| 
 | ||||
|     if (frames_written > 0) { | ||||
|         std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16)); | ||||
|     } | ||||
| 
 | ||||
|     // Hold last emitted frame; this prevents popping.
 | ||||
|     for (size_t i = frames_written; i < num_frames; i++) { | ||||
|         std::memcpy(buffer + 2 * i, &last_frame[0], 2 * sizeof(s16)); | ||||
|     } | ||||
| } | ||||
| 
 | ||||
|  |  | |||
|  | @ -9,6 +9,7 @@ | |||
| #include "audio_core/audio_types.h" | ||||
| #include "audio_core/time_stretch.h" | ||||
| #include "common/common_types.h" | ||||
| #include "common/ring_buffer.h" | ||||
| #include "core/memory.h" | ||||
| 
 | ||||
| namespace Service { | ||||
|  | @ -81,9 +82,12 @@ protected: | |||
| 
 | ||||
| private: | ||||
|     void FlushResidualStretcherAudio(); | ||||
|     void OutputCallback(s16* buffer, std::size_t num_frames); | ||||
| 
 | ||||
|     std::unique_ptr<Sink> sink; | ||||
|     bool perform_time_stretching = false; | ||||
|     Common::RingBuffer<s16, 0x2000, 2> fifo; | ||||
|     std::array<s16, 2> last_frame{}; | ||||
|     TimeStretcher time_stretcher; | ||||
| }; | ||||
| 
 | ||||
|  |  | |||
|  | @ -19,11 +19,7 @@ public: | |||
|         return native_sample_rate; | ||||
|     } | ||||
| 
 | ||||
|     void EnqueueSamples(const s16*, std::size_t) override {} | ||||
| 
 | ||||
|     std::size_t SamplesInQueue() const override { | ||||
|         return 0; | ||||
|     } | ||||
|     void SetCallback(std::function<void(s16*, std::size_t)>) override {} | ||||
| }; | ||||
| 
 | ||||
| } // namespace AudioCore
 | ||||
|  |  | |||
|  | @ -2,8 +2,8 @@ | |||
| // Licensed under GPLv2 or any later version
 | ||||
| // Refer to the license.txt file included.
 | ||||
| 
 | ||||
| #include <list> | ||||
| #include <numeric> | ||||
| #include <string> | ||||
| #include <vector> | ||||
| #include <SDL.h> | ||||
| #include "audio_core/audio_types.h" | ||||
| #include "audio_core/sdl2_sink.h" | ||||
|  | @ -17,7 +17,7 @@ struct SDL2Sink::Impl { | |||
| 
 | ||||
|     SDL_AudioDeviceID audio_device_id = 0; | ||||
| 
 | ||||
|     std::list<std::vector<s16>> queue; | ||||
|     std::function<void(s16*, std::size_t)> cb; | ||||
| 
 | ||||
|     static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes); | ||||
| }; | ||||
|  | @ -74,58 +74,18 @@ unsigned int SDL2Sink::GetNativeSampleRate() const { | |||
|     return impl->sample_rate; | ||||
| } | ||||
| 
 | ||||
| void SDL2Sink::EnqueueSamples(const s16* samples, std::size_t sample_count) { | ||||
|     if (impl->audio_device_id <= 0) | ||||
|         return; | ||||
| 
 | ||||
|     SDL_LockAudioDevice(impl->audio_device_id); | ||||
|     impl->queue.emplace_back(samples, samples + sample_count * 2); | ||||
|     SDL_UnlockAudioDevice(impl->audio_device_id); | ||||
| } | ||||
| 
 | ||||
| size_t SDL2Sink::SamplesInQueue() const { | ||||
|     if (impl->audio_device_id <= 0) | ||||
|         return 0; | ||||
| 
 | ||||
|     SDL_LockAudioDevice(impl->audio_device_id); | ||||
| 
 | ||||
|     std::size_t total_size = | ||||
|         std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<std::size_t>(0), | ||||
|                         [](std::size_t sum, const auto& buffer) { | ||||
|                             // Division by two because each stereo sample is made of
 | ||||
|                             // two s16.
 | ||||
|                             return sum + buffer.size() / 2; | ||||
|                         }); | ||||
| 
 | ||||
|     SDL_UnlockAudioDevice(impl->audio_device_id); | ||||
| 
 | ||||
|     return total_size; | ||||
| void SDL2Sink::SetCallback(std::function<void(s16*, std::size_t)> cb) { | ||||
|     impl->cb = cb; | ||||
| } | ||||
| 
 | ||||
| void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) { | ||||
|     Impl* impl = reinterpret_cast<Impl*>(impl_); | ||||
|     if (!impl || !impl->cb) | ||||
|         return; | ||||
| 
 | ||||
|     std::size_t remaining_size = static_cast<std::size_t>(buffer_size_in_bytes) / | ||||
|                                  sizeof(s16); // Keep track of size in 16-bit increments.
 | ||||
|     const size_t num_frames = buffer_size_in_bytes / (2 * sizeof(s16)); | ||||
| 
 | ||||
|     while (remaining_size > 0 && !impl->queue.empty()) { | ||||
|         if (impl->queue.front().size() <= remaining_size) { | ||||
|             memcpy(buffer, impl->queue.front().data(), impl->queue.front().size() * sizeof(s16)); | ||||
|             buffer += impl->queue.front().size() * sizeof(s16); | ||||
|             remaining_size -= impl->queue.front().size(); | ||||
|             impl->queue.pop_front(); | ||||
|         } else { | ||||
|             memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16)); | ||||
|             buffer += remaining_size * sizeof(s16); | ||||
|             impl->queue.front().erase(impl->queue.front().begin(), | ||||
|                                       impl->queue.front().begin() + remaining_size); | ||||
|             remaining_size = 0; | ||||
|         } | ||||
|     } | ||||
| 
 | ||||
|     if (remaining_size > 0) { | ||||
|         memset(buffer, 0, remaining_size * sizeof(s16)); | ||||
|     } | ||||
|     impl->cb(reinterpret_cast<s16*>(buffer), num_frames); | ||||
| } | ||||
| 
 | ||||
| std::vector<std::string> ListSDL2SinkDevices() { | ||||
|  |  | |||
|  | @ -17,9 +17,7 @@ public: | |||
| 
 | ||||
|     unsigned int GetNativeSampleRate() const override; | ||||
| 
 | ||||
|     void EnqueueSamples(const s16* samples, std::size_t sample_count) override; | ||||
| 
 | ||||
|     std::size_t SamplesInQueue() const override; | ||||
|     void SetCallback(std::function<void(s16*, std::size_t)> cb) override; | ||||
| 
 | ||||
| private: | ||||
|     struct Impl; | ||||
|  |  | |||
|  | @ -4,7 +4,7 @@ | |||
| 
 | ||||
| #pragma once | ||||
| 
 | ||||
| #include <vector> | ||||
| #include <functional> | ||||
| #include "common/common_types.h" | ||||
| 
 | ||||
| namespace AudioCore { | ||||
|  | @ -20,19 +20,16 @@ class Sink { | |||
| public: | ||||
|     virtual ~Sink() = default; | ||||
| 
 | ||||
|     /// The native rate of this sink. The sink expects to be fed samples that respect this. (Units:
 | ||||
|     /// samples/sec)
 | ||||
|     /// The native rate of this sink. The sink expects to be fed samples that respect this.
 | ||||
|     /// (Units: samples/sec)
 | ||||
|     virtual unsigned int GetNativeSampleRate() const = 0; | ||||
| 
 | ||||
|     /**
 | ||||
|      * Feed stereo samples to sink. | ||||
|      * Set callback for samples | ||||
|      * @param samples Samples in interleaved stereo PCM16 format. | ||||
|      * @param sample_count Number of samples. | ||||
|      */ | ||||
|     virtual void EnqueueSamples(const s16* samples, std::size_t sample_count) = 0; | ||||
| 
 | ||||
|     /// Samples enqueued that have not been played yet.
 | ||||
|     virtual std::size_t SamplesInQueue() const = 0; | ||||
|     virtual void SetCallback(std::function<void(s16*, std::size_t)> cb) = 0; | ||||
| }; | ||||
| 
 | ||||
| } // namespace AudioCore
 | ||||
|  |  | |||
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