Refactoring audio code

- Add AudioBus
- Move resampling to AudioBus
- Move SincResampler to engine/audio

build/android/app/CMakeLists.txt

@@ -49,7 +49,6 @@ project(kaliber)
 add_library(kaliber SHARED
     ../../../src/base/collusion_test.cc
     ../../../src/base/log.cc
-    ../../../src/base/sinc_resampler.cc
     ../../../src/base/task_runner.cc
     ../../../src/base/thread_pool.cc
     ../../../src/base/timer.cc
@@ -62,8 +61,10 @@ add_library(kaliber SHARED
     ../../../src/demo/sky_quad.cc
     ../../../src/engine/animatable.cc
     ../../../src/engine/animator.cc
+    ../../../src/engine/audio/audio_bus.cc
     ../../../src/engine/audio/audio_mixer.cc
     ../../../src/engine/audio/audio_sink_oboe.cc
+    ../../../src/engine/audio/sinc_resampler.cc
     ../../../src/engine/drawable.cc
     ../../../src/engine/engine.cc
     ../../../src/engine/font.cc

build/linux/Makefile

@@ -78,7 +78,6 @@ objs_from_src_in = $(call objs_from_src, $(shell find $(1) -name "*.cc" -o -name
 BASE_SRC := \
   $(SRC_ROOT)/base/collusion_test.cc \
   $(SRC_ROOT)/base/log.cc \
-  $(SRC_ROOT)/base/sinc_resampler.cc \
   $(SRC_ROOT)/base/task_runner.cc \
   $(SRC_ROOT)/base/thread_pool.cc \
   $(SRC_ROOT)/base/timer.cc
@@ -95,8 +94,10 @@ $(BASE_LIB): $(BASE_OBJS)
 ENGINE_SRC := \
   $(SRC_ROOT)/engine/animatable.cc \
   $(SRC_ROOT)/engine/animator.cc \
+  $(SRC_ROOT)/engine/audio/audio_bus.cc \
   $(SRC_ROOT)/engine/audio/audio_mixer.cc \
   $(SRC_ROOT)/engine/audio/audio_sink_alsa.cc \
+  $(SRC_ROOT)/engine/audio/sinc_resampler.cc \
   $(SRC_ROOT)/engine/drawable.cc \
   $(SRC_ROOT)/engine/engine.cc \
   $(SRC_ROOT)/engine/font.cc \

src/engine/audio/audio_bus.cc

@@ -0,0 +1,112 @@
+#include "engine/audio/audio_bus.h"
+
+#include "base/log.h"
+#include "engine/audio/sinc_resampler.h"
+#include "engine/engine.h"
+
+using namespace base;
+
+namespace eng {
+
+namespace {
+
+template <typename T>
+std::array<std::unique_ptr<T[]>, 2> Deinterleave(size_t num_channels,
+                                                 size_t num_samples,
+                                                 float* input_buffer) {
+  std::array<std::unique_ptr<T[]>, 2> channels;
+
+  if (num_channels == 1) {
+    // Single channel.
+    if constexpr (std::is_same<float, T>::value) {
+      channels[0] = std::make_unique<T[]>(num_samples);
+      memcpy(channels[0].get(), input_buffer, num_samples * sizeof(float));
+    } else {
+      channels[0] = std::make_unique<T[]>(num_samples);
+      for (int i = 0; i < num_samples; ++i)
+        channels[0].get()[i] = static_cast<T>(input_buffer[i]);
+    }
+  } else {
+    // Deinterleave into separate channels.
+    channels[0] = std::make_unique<T[]>(num_samples);
+    channels[1] = std::make_unique<T[]>(num_samples);
+    for (size_t i = 0, j = 0; i < num_samples * 2; i += 2) {
+      channels[0].get()[j] = static_cast<T>(input_buffer[i]);
+      channels[1].get()[j++] = static_cast<T>(input_buffer[i + 1]);
+    }
+  }
+
+  return channels;
+}
+
+std::unique_ptr<SincResampler> CreateResampler(int src_samle_rate,
+                                               int dst_sample_rate,
+                                               size_t num_samples) {
+  const double io_ratio = static_cast<double>(src_samle_rate) /
+                          static_cast<double>(dst_sample_rate);
+  auto resampler = std::make_unique<SincResampler>(io_ratio, num_samples);
+  resampler->PrimeWithSilence();
+  return resampler;
+}
+
+}  // namespace
+
+AudioBus::AudioBus() = default;
+AudioBus::~AudioBus() = default;
+
+void AudioBus::SetAudioConfig(size_t num_channels, size_t sample_rate) {
+  num_channels_ = num_channels;
+  sample_rate_ = sample_rate;
+}
+
+void AudioBus::FromInterleaved(std::unique_ptr<float[]> input_buffer,
+                               size_t samples_per_channel) {
+  auto channels = Deinterleave<float>(num_channels_, samples_per_channel,
+                                      input_buffer.get());
+
+  size_t hw_sample_rate = Engine::Get().GetAudioHardwareSampleRate();
+
+  if (hw_sample_rate == sample_rate_) {
+    // No need for resmapling.
+    channel_data_[0] = std::move(channels[0]);
+    if (num_channels_ == 2)
+      channel_data_[1] = std::move(channels[1]);
+    samples_per_channel_ = samples_per_channel;
+  } else {
+    if (!resampler_[0]) {
+      for (size_t i = 0; i < num_channels_; ++i) {
+        resampler_[i] =
+            CreateResampler(sample_rate_, hw_sample_rate, samples_per_channel);
+      }
+    }
+
+    size_t num_resampled_samples =
+        (size_t)(((float)hw_sample_rate / (float)sample_rate_) *
+                 samples_per_channel);
+    DCHECK(num_resampled_samples <= (size_t)resampler_[0]->ChunkSize());
+
+    if (!channel_data_[0]) {
+      channel_data_[0] = std::make_unique<float[]>(num_resampled_samples);
+      if (num_channels_ == 2)
+        channel_data_[1] = std::make_unique<float[]>(num_resampled_samples);
+    }
+    samples_per_channel_ = num_resampled_samples;
+
+    // Resample to match the system sample rate.
+    for (size_t i = 0; i < num_channels_; ++i) {
+      resampler_[i]->Resample(num_resampled_samples, channel_data_[i].get(),
+                              [&](int frames, float* destination) {
+                                memcpy(destination, channels[i].get(),
+                                       frames * sizeof(float));
+                              });
+    }
+
+    if (IsEndOfStream()) {
+      // We are done with the resampler.
+      for (size_t i = 0; i < num_channels_; ++i)
+        resampler_[i].reset();
+    }
+  }
+}
+
+}  // namespace eng

src/engine/audio/audio_bus.h

@@ -0,0 +1,45 @@
+#ifndef ENGINE_AUDIO_AUDIO_BUS_H
+#define ENGINE_AUDIO_AUDIO_BUS_H
+
+#include <memory>
+
+namespace eng {
+
+class SincResampler;
+
+class AudioBus {
+ public:
+  AudioBus();
+  virtual ~AudioBus();
+
+  virtual void Stream(bool loop) = 0;
+  virtual void SwapBuffers() = 0;
+  virtual void ResetStream() = 0;
+  virtual bool IsEndOfStream() const = 0;
+
+  float* GetChannelData(int channel) const {
+    return channel_data_[channel].get();
+  }
+
+  size_t samples_per_channel() const { return samples_per_channel_; }
+  int sample_rate() const { return sample_rate_; }
+  int num_channels() const { return num_channels_; }
+
+ protected:
+  void SetAudioConfig(size_t num_channels, size_t sample_rate);
+
+  void FromInterleaved(std::unique_ptr<float[]> input_buffer,
+                       size_t samples_per_channel);
+
+ private:
+  std::unique_ptr<float[]> channel_data_[2];
+  size_t samples_per_channel_ = 0;
+  size_t sample_rate_ = 0;
+  size_t num_channels_ = 0;
+
+  std::unique_ptr<SincResampler> resampler_[2];
+};
+
+}  // namespace eng
+
+#endif  // ENGINE_AUDIO_AUDIO_BUS_H

src/engine/audio/audio_mixer.cc

@@ -5,7 +5,7 @@
 #include "base/log.h"
 #include "base/task_runner.h"
 #include "base/thread_pool.h"
-#include "engine/sound.h"
+#include "engine/audio/audio_bus.h"
 
 #if defined(__ANDROID__)
 #include "engine/audio/audio_sink_oboe.h"
@@ -49,7 +49,7 @@ void AudioMixer::DestroyResource(uint64_t resource_id) {
 }
 
 void AudioMixer::Play(uint64_t resource_id,
-                      std::shared_ptr<Sound> sound,
+                      std::shared_ptr<AudioBus> sound,
                       float amplitude,
                       bool reset_pos) {
   if (!audio_enabled_)
@@ -76,6 +76,8 @@ void AudioMixer::Play(uint64_t resource_id,
     it->second->src_index = 0;
     it->second->accumulator = 0;
     sound->ResetStream();
+  } else if (it->second->src_index >= sound->samples_per_channel()) {
+    return;
   }
 
   it->second->active = true;
@@ -161,7 +163,7 @@ void AudioMixer::Resume() {
   audio_sink_->Resume();
 }
 
-int AudioMixer::GetHardwareSampleRate() {
+size_t AudioMixer::GetHardwareSampleRate() {
   return audio_sink_->GetHardwareSampleRate();
 }
 
@@ -182,11 +184,12 @@ void AudioMixer::RenderAudio(float* output_buffer, size_t num_frames) {
     if (flags & kStopped) {
       marked_for_removal = true;
     } else {
-      const float* src[2] = {sound->GetBuffer(0), sound->GetBuffer(1)};
+      const float* src[2] = {sound->GetChannelData(0),
+                             sound->GetChannelData(1)};
       if (!src[1])
         src[1] = src[0];  // mono.
 
-      size_t num_samples = sound->GetNumSamples();
+      size_t num_samples = sound->samples_per_channel();
       size_t src_index = it->get()->src_index;
       size_t step = it->get()->step.load(std::memory_order_relaxed);
       size_t accumulator = it->get()->accumulator;
@@ -195,71 +198,59 @@ void AudioMixer::RenderAudio(float* output_buffer, size_t num_frames) {
           it->get()->amplitude_inc.load(std::memory_order_relaxed);
       float max_amplitude =
           it->get()->max_amplitude.load(std::memory_order_relaxed);
-
       size_t channel_offset =
-          (flags & kSimulateStereo) && !sound->is_streaming_sound()
+          (flags & kSimulateStereo) ? sound->sample_rate() / 10 : 0;
-              ? sound->sample_rate() / 10
-              : 0;
 
-      DCHECK(num_samples || sound->is_streaming_sound());
+      DCHECK(num_samples > 0);
 
       for (size_t i = 0; i < num_frames * kChannelCount;) {
-        if (num_samples) {
+        // Mix the 1st channel.
-          // Mix the 1st channel.
+        output_buffer[i++] += src[0][src_index] * amplitude;
-          output_buffer[i++] += src[0][src_index] * amplitude;
 
-          // Mix the 2nd channel. Offset the source index for stereo simulation.
+        // Mix the 2nd channel. Offset the source index for stereo simulation.
-          size_t ind = channel_offset + src_index;
+        size_t ind = channel_offset + src_index;
-          if (ind < num_samples)
+        if (ind < num_samples)
-            output_buffer[i++] += src[1][ind] * amplitude;
+          output_buffer[i++] += src[1][ind] * amplitude;
-          else if (flags & kLoop)
+        else if (flags & kLoop)
-            output_buffer[i++] += src[1][ind % num_samples] * amplitude;
+          output_buffer[i++] += src[1][ind % num_samples] * amplitude;
-          else
+        else
-            i++;
+          i++;
 
-          // Apply amplitude modification.
+        // Apply amplitude modification.
-          amplitude += amplitude_inc;
+        amplitude += amplitude_inc;
-          if (amplitude <= 0) {
+        if (amplitude <= 0) {
-            marked_for_removal = true;
+          marked_for_removal = true;
-            break;
+          break;
-          } else if (amplitude > max_amplitude) {
+        } else if (amplitude > max_amplitude) {
-            amplitude = max_amplitude;
+          amplitude = max_amplitude;
-          }
-
-          // Advance source index. Apply basic resampling for variations.
-          accumulator += step;
-          src_index += accumulator / 100;
-          accumulator %= 100;
         }
 
+        // Advance source index. Apply basic resampling for variations.
+        accumulator += step;
+        src_index += accumulator / 100;
+        accumulator %= 100;
+
         // Remove, loop or stream if the source data is consumed
         if (src_index >= num_samples) {
-          if (num_samples)
+          src_index %= num_samples;
-            src_index %= num_samples;
 
-          if (!sound->is_streaming_sound()) {
+          if (sound->IsEndOfStream()) {
-            if (!(flags & kLoop)) {
+            marked_for_removal = true;
-              marked_for_removal = true;
+            break;
-              break;
+          }
-            }
-          } else if (!it->get()->streaming_in_progress.load(
-                         std::memory_order_acquire)) {
-            if (sound->eof()) {
-              DCHECK((flags & kLoop) == 0);
-              marked_for_removal = true;
-              break;
-            }
 
+          if (!it->get()->streaming_in_progress.load(
+                  std::memory_order_acquire)) {
             it->get()->streaming_in_progress.store(true,
                                                    std::memory_order_relaxed);
 
             // Swap buffers and start streaming in background.
             sound->SwapBuffers();
-            src[0] = sound->GetBuffer(0);
+            src[0] = sound->GetChannelData(0);
-            src[1] = sound->GetBuffer(1);
+            src[1] = sound->GetChannelData(1);
             if (!src[1])
               src[1] = src[0];  // mono.
-            num_samples = sound->GetNumSamples();
+            num_samples = sound->samples_per_channel();
 
             ThreadPool::Get().PostTask(
                 HERE,
@@ -284,8 +275,7 @@ void AudioMixer::RenderAudio(float* output_buffer, size_t num_frames) {
   }
 
   for (auto it = end_list_.begin(); it != end_list_.end();) {
-    if ((!it->get()->sound->is_streaming_sound() ||
+    if (!it->get()->streaming_in_progress.load(std::memory_order_relaxed)) {
-         !it->get()->streaming_in_progress.load(std::memory_order_relaxed))) {
       main_thread_task_runner_->PostTask(
           HERE, std::bind(&AudioMixer::EndCallback, this, *it));
       it = end_list_.erase(it);

src/engine/audio/audio_mixer.h

@@ -17,7 +17,7 @@ class TaskRunner;
 namespace eng {
 
 class AudioSink;
-class Sound;
+class AudioBus;
 
 // Mix and render audio with low overhead. A platform specific AudioSink
 // implementation is expected to periodically call RenderAudio() in a background
@@ -31,7 +31,7 @@ class AudioMixer : public AudioSinkDelegate {
   void DestroyResource(uint64_t resource_id);
 
   void Play(uint64_t resource_id,
-            std::shared_ptr<Sound> sound,
+            std::shared_ptr<AudioBus> sound,
             float amplitude,
             bool reset_pos);
   void Stop(uint64_t resource_id);
@@ -48,7 +48,7 @@ class AudioMixer : public AudioSinkDelegate {
   void Suspend();
   void Resume();
 
-  int GetHardwareSampleRate();
+  size_t GetHardwareSampleRate();
 
  private:
   enum SampleFlags { kLoop = 1, kStopped = 2, kSimulateStereo = 4 };
@@ -62,7 +62,7 @@ class AudioMixer : public AudioSinkDelegate {
     base::Closure restart_cb;
 
     // Initialized by main thread, used by audio thread.
-    std::shared_ptr<Sound> sound;
+    std::shared_ptr<AudioBus> sound;
     size_t src_index = 0;
     size_t accumulator = 0;
     float amplitude = 1.0f;

src/engine/audio/audio_sink.h

@@ -15,7 +15,7 @@ class AudioSink {
   virtual void Suspend() = 0;
   virtual void Resume() = 0;
 
-  virtual int GetHardwareSampleRate() = 0;
+  virtual size_t GetHardwareSampleRate() = 0;
 
  private:
   AudioSink(const AudioSink&) = delete;

src/engine/audio/audio_sink_alsa.cc

@@ -152,7 +152,7 @@ void AudioSinkAlsa::Resume() {
   suspend_audio_thread_.store(false, std::memory_order_relaxed);
 }
 
-int AudioSinkAlsa::GetHardwareSampleRate() {
+size_t AudioSinkAlsa::GetHardwareSampleRate() {
   return sample_rate_;
 }
 

src/engine/audio/audio_sink_alsa.h

@@ -22,7 +22,7 @@ class AudioSinkAlsa final : public AudioSink {
   void Suspend() final;
   void Resume() final;
 
-  int GetHardwareSampleRate() final;
+  size_t GetHardwareSampleRate() final;
 
  private:
   // Handle for the PCM device.
@@ -33,7 +33,7 @@ class AudioSinkAlsa final : public AudioSink {
   std::atomic<bool> suspend_audio_thread_{false};
 
   size_t num_channels_ = 0;
-  int sample_rate_ = 0;
+  size_t sample_rate_ = 0;
   size_t period_size_ = 0;
 
   AudioSinkDelegate* delegate_ = nullptr;

src/engine/audio/audio_sink_oboe.cc

@@ -29,7 +29,7 @@ void AudioSinkOboe::Resume() {
   stream_->start();
 }
 
-int AudioSinkOboe::GetHardwareSampleRate() {
+size_t AudioSinkOboe::GetHardwareSampleRate() {
   return stream_->getSampleRate();
 }
 

src/engine/audio/audio_sink_oboe.h

@@ -22,7 +22,7 @@ class AudioSinkOboe final : public AudioSink {
   void Suspend() final;
   void Resume() final;
 
-  int GetHardwareSampleRate() final;
+  size_t GetHardwareSampleRate() final;
 
  private:
   class StreamCallback final : public oboe::AudioStreamCallback {

src/engine/audio/sinc_resampler.cc

@@ -74,7 +74,7 @@
 // Note: we're glossing over how the sub-sample handling works with
 // |virtual_source_idx_|, etc.
 
-#include "base/sinc_resampler.h"
+#include "engine/audio/sinc_resampler.h"
 
 #include <cmath>
 #include <cstring>
@@ -94,7 +94,9 @@
 #include <arm_neon.h>
 #endif
 
-namespace base {
+using namespace base;
+
+namespace eng {
 
 namespace {
 
@@ -257,14 +259,14 @@ void SincResampler::InitializeKernel() {
     for (int i = 0; i < kernel_size_; ++i) {
       const int idx = i + offset_idx * kernel_size_;
       const float pre_sinc =
-          base::kPiFloat * (i - kernel_size_ / 2 - subsample_offset);
+          kPiFloat * (i - kernel_size_ / 2 - subsample_offset);
       kernel_pre_sinc_storage_[idx] = pre_sinc;
 
       // Compute Blackman window, matching the offset of the sinc().
       const float x = (i - subsample_offset) / kernel_size_;
       const float window =
-          static_cast<float>(kA0 - kA1 * cos(2.0 * base::kPiDouble * x) +
+          static_cast<float>(kA0 - kA1 * cos(2.0 * kPiDouble * x) +
-                             kA2 * cos(4.0 * base::kPiDouble * x));
+                             kA2 * cos(4.0 * kPiDouble * x));
       kernel_window_storage_[idx] = window;
 
       // Compute the sinc with offset, then window the sinc() function and store
@@ -545,4 +547,4 @@ float SincResampler::Convolve_NEON(const int kernel_size,
 }
 #endif
 
-}  // namespace base
+}  // namespace eng

src/engine/audio/sinc_resampler.h

@@ -1,16 +1,18 @@
 // Copyright 2012 The Chromium Authors
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
+//
+// Modified for Kaliber engine.
 
-#ifndef BASE_SINC_RESAMPLER_H
+#ifndef ENGINE_AUDIO__SINC_RESAMPLER_H
-#define BASE_SINC_RESAMPLER_H
+#define ENGINE_AUDIO__SINC_RESAMPLER_H
 
 #include <functional>
 #include <memory>
 
 #include "base/mem.h"
 
-namespace base {
+namespace eng {
 
 // SincResampler is a high-quality single-channel sample-rate converter.
 class SincResampler {
@@ -163,12 +165,12 @@ class SincResampler {
   // Contains kKernelOffsetCount kernels back-to-back, each of size
   // `kernel_size_`. The kernel offsets are sub-sample shifts of a windowed sinc
   // shifted from 0.0 to 1.0 sample.
-  AlignedMemPtr<float[]> kernel_storage_;
+  base::AlignedMemPtr<float[]> kernel_storage_;
-  AlignedMemPtr<float[]> kernel_pre_sinc_storage_;
+  base::AlignedMemPtr<float[]> kernel_pre_sinc_storage_;
-  AlignedMemPtr<float[]> kernel_window_storage_;
+  base::AlignedMemPtr<float[]> kernel_window_storage_;
 
   // Data from the source is copied into this buffer for each processing pass.
-  AlignedMemPtr<float[]> input_buffer_;
+  base::AlignedMemPtr<float[]> input_buffer_;
 
   // Stores the runtime selection of which Convolve function to use.
   using ConvolveProc =
@@ -184,6 +186,6 @@ class SincResampler {
   float* r4_;
 };
 
-}  // namespace base
+}  // namespace eng
 
-#endif  // BASE_SINC_RESAMPLER_H
+#endif  // ENGINE_AUDIO__SINC_RESAMPLER_H

src/engine/engine.cc

@@ -474,7 +474,7 @@ const std::string& Engine::GetSharedDataPath() const {
   return platform_->GetSharedDataPath();
 }
 
-int Engine::GetAudioHardwareSampleRate() {
+size_t Engine::GetAudioHardwareSampleRate() {
   return audio_mixer_->GetHardwareSampleRate();
 }
 

src/engine/engine.h

@@ -137,7 +137,7 @@ class Engine : public PlatformObserver {
 
   const std::string& GetSharedDataPath() const;
 
-  int GetAudioHardwareSampleRate();
+  size_t GetAudioHardwareSampleRate();
 
   bool IsMobile() const;
 

src/engine/sound.cc

@@ -3,7 +3,6 @@
 #include <array>
 
 #include "base/log.h"
-#include "base/sinc_resampler.h"
 #define MINIMP3_ONLY_MP3
 #define MINIMP3_ONLY_SIMD
 #define MINIMP3_FLOAT_OUTPUT
@@ -15,53 +14,10 @@
 
 using namespace base;
 
-namespace {
-
-constexpr size_t kMaxSamplesPerChunk = MINIMP3_MAX_SAMPLES_PER_FRAME * 10;
-
-template <typename T>
-std::array<std::unique_ptr<T[]>, 2> Deinterleave(size_t num_channels,
-                                                 size_t num_samples,
-                                                 float* input_buffer) {
-  std::array<std::unique_ptr<T[]>, 2> channels;
-
-  if (num_channels == 1) {
-    // Single channel.
-    if constexpr (std::is_same<float, T>::value) {
-      channels[0] = std::make_unique<T[]>(num_samples);
-      memcpy(channels[0].get(), input_buffer, num_samples * sizeof(float));
-    } else {
-      channels[0] = std::make_unique<T[]>(num_samples);
-      for (int i = 0; i < num_samples; ++i)
-        channels[0].get()[i] = static_cast<T>(input_buffer[i]);
-    }
-  } else {
-    // Deinterleave into separate channels.
-    channels[0] = std::make_unique<T[]>(num_samples);
-    channels[1] = std::make_unique<T[]>(num_samples);
-    for (size_t i = 0, j = 0; i < num_samples * 2; i += 2) {
-      channels[0].get()[j] = static_cast<T>(input_buffer[i]);
-      channels[1].get()[j++] = static_cast<T>(input_buffer[i + 1]);
-    }
-  }
-
-  return channels;
-}
-
-std::unique_ptr<SincResampler> CreateResampler(int src_samle_rate,
-                                               int dst_sample_rate,
-                                               size_t num_samples) {
-  const double io_ratio = static_cast<double>(src_samle_rate) /
-                          static_cast<double>(dst_sample_rate);
-  auto resampler = std::make_unique<SincResampler>(io_ratio, num_samples);
-  resampler->PrimeWithSilence();
-  return resampler;
-}
-
-}  // namespace
-
 namespace eng {
 
+constexpr size_t kMaxSamplesPerChunk = MINIMP3_MAX_SAMPLES_PER_FRAME;
+
 Sound::Sound() = default;
 
 Sound::~Sound() {
@@ -92,164 +48,98 @@ bool Sound::Load(const std::string& file_name, bool stream) {
     return false;
   }
 
-  is_streaming_sound_ = stream;
+  LOG << (stream ? "Streaming " : "Loaded ") << file_name << ". "
-
-  LOG << (is_streaming_sound_ ? "Streaming " : "Loaded ") << file_name << ". "
       << mp3_dec_->samples << " samples, " << mp3_dec_->info.channels
       << " channels, " << mp3_dec_->info.hz << " hz, "
       << "layer " << mp3_dec_->info.layer << ", "
       << "avg_bitrate_kbps " << mp3_dec_->info.bitrate_kbps;
 
-  num_channels_ = mp3_dec_->info.channels;
+  SetAudioConfig(mp3_dec_->info.channels, mp3_dec_->info.hz);
-  sample_rate_ = mp3_dec_->info.hz;
-  num_samples_back_ = cur_sample_back_ = 0;
-  eof_ = false;
 
-  DCHECK(num_channels_ > 0 && num_channels_ <= 2);
+  samples_per_channel_ = 0;
+  eos_ = false;
 
-  hw_sample_rate_ = Engine::Get().GetAudioHardwareSampleRate();
+  DCHECK(mp3_dec_->info.channels > 0 && mp3_dec_->info.channels <= 2);
 
-  if (is_streaming_sound_) {
+  if (stream) {
-    if (sample_rate_ != hw_sample_rate_) {
+    // Fill up the buffer.
-      for (int i = 0; i < mp3_dec_->info.channels; ++i) {
+    StreamInternal(kMaxSamplesPerChunk * mp3_dec_->info.channels, false);
-        resampler_[i] =
-            CreateResampler(sample_rate_, hw_sample_rate_,
-                            (int)kMaxSamplesPerChunk / mp3_dec_->info.channels);
-      }
-    }
-
-    // Fill up buffers.
-    StreamInternal(kMaxSamplesPerChunk, false);
     SwapBuffers();
-    StreamInternal(kMaxSamplesPerChunk, false);
+    StreamInternal(kMaxSamplesPerChunk * mp3_dec_->info.channels, false);
-
     // No need to stream if sample fits into the buffer.
-    if (eof_)
+    if (eos_)
-      is_streaming_sound_ = false;
+      stream = false;
   } else {
-    if (sample_rate_ != hw_sample_rate_) {
-      for (int i = 0; i < mp3_dec_->info.channels; ++i) {
-        resampler_[i] =
-            CreateResampler(sample_rate_, hw_sample_rate_,
-                            mp3_dec_->samples / mp3_dec_->info.channels);
-      }
-    }
-
     // Decode entire file.
     StreamInternal(mp3_dec_->samples, false);
     SwapBuffers();
-    eof_ = true;
+    eos_ = true;
   }
 
-  if (!is_streaming_sound_) {
+  if (!stream) {
     // We are done with decoding.
     encoded_data_.reset();
-    for (int i = 0; i < mp3_dec_->info.channels; ++i)
-      resampler_[i].reset();
     mp3dec_ex_close(mp3_dec_.get());
     mp3_dec_.reset();
   }
 
+  read_pos_ = 0;
+
   return true;
 }
 
-bool Sound::Stream(bool loop) {
+void Sound::Stream(bool loop) {
-  DCHECK(is_streaming_sound_);
+  DCHECK(mp3_dec_);
-
+  StreamInternal(kMaxSamplesPerChunk * mp3_dec_->info.channels, loop);
-  return StreamInternal(kMaxSamplesPerChunk, loop);
 }
 
 void Sound::SwapBuffers() {
-  front_buffer_[0].swap(back_buffer_[0]);
+  FromInterleaved(std::move(interleaved_data_), samples_per_channel_);
-  front_buffer_[1].swap(back_buffer_[1]);
+  samples_per_channel_ = 0;
-
-  cur_sample_front_ = cur_sample_back_;
-  num_samples_front_ = num_samples_back_;
-  num_samples_back_ = 0;
 }
 
 void Sound::ResetStream() {
-  if (is_streaming_sound_ && cur_sample_front_ != 0) {
+  if (mp3_dec_ && read_pos_ != 0) {
-    // Seek to 0 and ivalidate decoded data.
+    // Seek to 0 and stream.
     mp3dec_ex_seek(mp3_dec_.get(), 0);
-    eof_ = false;
+    eos_ = false;
-    num_samples_back_ = num_samples_front_ = 0;
+    StreamInternal(kMaxSamplesPerChunk * mp3_dec_->info.channels, false);
-    cur_sample_front_ = cur_sample_back_ = 0;
+    SwapBuffers();
+    StreamInternal(kMaxSamplesPerChunk * mp3_dec_->info.channels, false);
+    read_pos_ = 0;
   }
 }
 
-float* Sound::GetBuffer(int channel) const {
+void Sound::StreamInternal(size_t num_samples, bool loop) {
-  return front_buffer_[channel].get();
-}
-
-bool Sound::StreamInternal(size_t num_samples, bool loop) {
   auto buffer = std::make_unique<float[]>(num_samples);
   size_t samples_read_per_channel = 0;
 
-  cur_sample_back_ = mp3_dec_->cur_sample;
+  read_pos_ = mp3_dec_->cur_sample;
 
   for (;;) {
     size_t samples_read =
         mp3dec_ex_read(mp3_dec_.get(), buffer.get(), num_samples);
     if (samples_read != num_samples && mp3_dec_->last_error) {
       LOG << "mp3 decode error: " << mp3_dec_->last_error;
-      eof_ = true;
+      break;
-      return false;
     }
 
-    samples_read_per_channel = samples_read / mp3_dec_->info.channels;
+    if (samples_read == 0 && loop) {
-    if (!samples_read_per_channel && loop) {
       mp3dec_ex_seek(mp3_dec_.get(), 0);
       loop = false;
       continue;
     }
+
+    samples_read_per_channel = samples_read / mp3_dec_->info.channels;
     break;
   }
 
-  if (samples_read_per_channel) {
+  if (samples_read_per_channel > 0) {
-    Preprocess(std::move(buffer), samples_read_per_channel);
+    interleaved_data_ = std::move(buffer);
+    samples_per_channel_ = samples_read_per_channel;
   } else {
-    num_samples_back_ = 0;
+    samples_per_channel_ = 0;
-    eof_ = true;
+    eos_ = true;
-  }
-
-  return true;
-}
-
-void Sound::Preprocess(std::unique_ptr<float[]> input_buffer,
-                       size_t samples_per_channel) {
-  auto channels = Deinterleave<float>(num_channels_, samples_per_channel,
-                                      input_buffer.get());
-
-  if (hw_sample_rate_ == sample_rate_) {
-    // No need for resmapling.
-    back_buffer_[0] = std::move(channels[0]);
-    if (num_channels_ == 2)
-      back_buffer_[1] = std::move(channels[1]);
-    num_samples_back_ = samples_per_channel;
-  } else {
-    size_t num_resampled_samples = resampler_[0]->ChunkSize();
-    DCHECK(num_resampled_samples ==
-           (size_t)(((float)hw_sample_rate_ / (float)sample_rate_) *
-                    samples_per_channel));
-
-    if (!back_buffer_[0]) {
-      if (max_samples_ < num_resampled_samples)
-        max_samples_ = num_resampled_samples;
-      back_buffer_[0] = std::make_unique<float[]>(max_samples_);
-      if (num_channels_ == 2)
-        back_buffer_[1] = std::make_unique<float[]>(max_samples_);
-    }
-    num_samples_back_ = num_resampled_samples;
-
-    // Resample to match the system sample rate.
-    for (size_t i = 0; i < num_channels_; ++i) {
-      resampler_[i]->Resample(num_resampled_samples, back_buffer_[i].get(),
-                              [&](int frames, float* destination) {
-                                memcpy(destination, channels[i].get(),
-                                       frames * sizeof(float));
-                              });
-    }
   }
 }
 

src/engine/sound.h

@@ -5,11 +5,9 @@
 #include <memory>
 #include <string>
 
-typedef struct mp3dec_ex mp3dec_ex_t;
+#include "engine/audio/audio_bus.h"
 
-namespace base {
+typedef struct mp3dec_ex mp3dec_ex_t;
-class SincResampler;
-}  // namespace base
 
 namespace eng {
 
@@ -17,61 +15,30 @@ namespace eng {
 // files. Resamples the decoded audio to match the system sample rate if
 // necessary. Non-streaming sounds Can be shared between multiple audio
 // resources and played simultaneously.
-class Sound {
+class Sound final : public AudioBus {
  public:
   Sound();
-  ~Sound();
+  ~Sound() final;
 
   bool Load(const std::string& file_name, bool stream);
 
-  bool Stream(bool loop);
+  // AudioBus interface
-
+  void Stream(bool loop) final;
-  void SwapBuffers();
+  void SwapBuffers() final;
-
+  void ResetStream() final;
-  void ResetStream();
+  bool IsEndOfStream() const final { return eos_; }
-
-  float* GetBuffer(int channel) const;
-
-  size_t GetNumSamples() const { return num_samples_front_; }
-
-  size_t num_channels() const { return num_channels_; }
-  int sample_rate() const { return sample_rate_; }
-
-  bool is_streaming_sound() const { return is_streaming_sound_; }
-
-  bool eof() const { return eof_; }
 
  private:
   // Buffer holding decoded audio.
-  std::unique_ptr<float[]> back_buffer_[2];
+  std::unique_ptr<float[]> interleaved_data_;
-  std::unique_ptr<float[]> front_buffer_[2];
+  size_t samples_per_channel_ = 0;
-
-  size_t num_samples_back_ = 0;
-  size_t num_samples_front_ = 0;
-  size_t max_samples_ = 0;
-
-  size_t cur_sample_front_ = 0;
-  size_t cur_sample_back_ = 0;
-
-  size_t num_channels_ = 0;
-  int sample_rate_ = 0;
-
-  int hw_sample_rate_ = 0;
 
   std::unique_ptr<char[]> encoded_data_;
-
   std::unique_ptr<mp3dec_ex_t> mp3_dec_;
+  uint64_t read_pos_ = 0;
+  bool eos_ = false;
 
-  std::unique_ptr<base::SincResampler> resampler_[2];
+  void StreamInternal(size_t num_samples, bool loop);
-
-  bool eof_ = false;
-
-  bool is_streaming_sound_ = false;
-
-  bool StreamInternal(size_t num_samples, bool loop);
-
-  void Preprocess(std::unique_ptr<float[]> input_buffer,
-                  size_t samples_per_channel);
 };
 
 }  // namespace eng

src/engine/sound_player.cc

@@ -2,9 +2,9 @@
 
 #include "base/interpolation.h"
 #include "base/log.h"
+#include "engine/audio/audio_bus.h"
 #include "engine/audio/audio_mixer.h"
 #include "engine/engine.h"
-#include "engine/sound.h"
 
 using namespace base;
 
@@ -17,16 +17,10 @@ SoundPlayer::~SoundPlayer() {
   Engine::Get().GetAudioMixer()->DestroyResource(resource_id_);
 }
 
-void SoundPlayer::SetSound(std::shared_ptr<Sound> sound) {
+void SoundPlayer::SetSound(std::shared_ptr<AudioBus> sound) {
-  CHECK(!sound->is_streaming_sound()) << "Streaming sound cannot be shared.";
-
   sound_ = sound;
 }
 
-void SoundPlayer::SetSound(std::unique_ptr<Sound> sound) {
-  sound_ = std::move(sound);
-}
-
 void SoundPlayer::Play(bool loop, float fade_in_duration) {
   if (!sound_)
     return;

src/engine/sound_player.h

@@ -7,15 +7,14 @@
 
 namespace eng {
 
-class Sound;
+class AudioBus;
 
 class SoundPlayer {
  public:
   SoundPlayer();
   ~SoundPlayer();
 
-  void SetSound(std::shared_ptr<Sound> sound);
+  void SetSound(std::shared_ptr<AudioBus> sound);
-  void SetSound(std::unique_ptr<Sound> sound);
 
   void Play(bool loop, float fade_in_duration = 0);
 
@@ -37,7 +36,7 @@ class SoundPlayer {
 
  private:
   uint64_t resource_id_ = 0;
-  std::shared_ptr<Sound> sound_;
+  std::shared_ptr<AudioBus> sound_;
 
   float max_amplitude_ = 1.0f;