audio_core: Simplify sink interface

src/audio_core/cubeb_sink.cpp

@@ -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::function<void(s16*, std::size_t)> cb;
-    std::vector<s16> queue;
 
     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) {
+void CubebSink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
-    if (!impl->ctx)
+    impl->cb = cb;
-        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;
 }
 
 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};
+    impl->cb(buffer, num_frames);
-
-    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);
-    }
 
     return num_frames;
 }

src/audio_core/cubeb_sink.h

@@ -17,9 +17,7 @@ public:
 
     unsigned int GetNativeSampleRate() const override;
 
-    void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
+    void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
-
-    std::size_t SamplesInQueue() const override;
 
 private:
     struct Impl;

src/audio_core/dsp_interface.cpp

@@ -12,16 +12,13 @@
 namespace AudioCore {
 
 DspInterface::DspInterface() = default;
-
+DspInterface::~DspInterface() = default;
-DspInterface::~DspInterface() {
-    if (perform_time_stretching) {
-        FlushResidualStretcherAudio();
-    }
-}
 
 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) {
+    fifo.Push(frame.data(), frame.size());
-        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());
-    }
 }
 
-void DspInterface::FlushResidualStretcherAudio() {
+void DspInterface::FlushResidualStretcherAudio() {}
-    if (!sink)
-        return;
 
-    time_stretcher.Flush();
+void DspInterface::OutputCallback(s16* buffer, size_t num_frames) {
-    while (true) {
+    const size_t frames_written = fifo.Pop(buffer, num_frames);
-        std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
+
-        if (residual_audio.empty())
+    if (frames_written > 0) {
-            break;
+        std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16));
-        sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
+    }
+
+    // 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));
     }
 }
 

src/audio_core/dsp_interface.h

@@ -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;
 };
 

src/audio_core/null_sink.h

@@ -19,11 +19,7 @@ public:
         return native_sample_rate;
     }
 
-    void EnqueueSamples(const s16*, std::size_t) override {}
+    void SetCallback(std::function<void(s16*, std::size_t)>) override {}
-
-    std::size_t SamplesInQueue() const override {
-        return 0;
-    }
 };
 
 } // namespace AudioCore

src/audio_core/sdl2_sink.cpp

@@ -2,8 +2,8 @@
 // Licensed under GPLv2 or any later version
 // Refer to the license.txt file included.
 
-#include <list>
+#include <string>
-#include <numeric>
+#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) {
+void SDL2Sink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
-    if (impl->audio_device_id <= 0)
+    impl->cb = cb;
-        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::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) /
+    const size_t num_frames = buffer_size_in_bytes / (2 * sizeof(s16));
-                                 sizeof(s16); // Keep track of size in 16-bit increments.
 
-    while (remaining_size > 0 && !impl->queue.empty()) {
+    impl->cb(reinterpret_cast<s16*>(buffer), num_frames);
-        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));
-    }
 }
 
 std::vector<std::string> ListSDL2SinkDevices() {

src/audio_core/sdl2_sink.h

@@ -17,9 +17,7 @@ public:
 
     unsigned int GetNativeSampleRate() const override;
 
-    void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
+    void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
-
-    std::size_t SamplesInQueue() const override;
 
 private:
     struct Impl;

src/audio_core/sink.h

@@ -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:
+    /// The native rate of this sink. The sink expects to be fed samples that respect this.
-    /// samples/sec)
+    /// (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;
+    virtual void SetCallback(std::function<void(s16*, std::size_t)> cb) = 0;
-
-    /// Samples enqueued that have not been played yet.
-    virtual std::size_t SamplesInQueue() const = 0;
 };
 
 } // namespace AudioCore