Moved the AEC render sample queue into the audio processing module

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 101 matching lines @@
       return uppermost_native_rate;
     }
     if (rate >= minimum_rate) {
       return rate;
     }
   }
   RTC_NOTREACHED();
   return uppermost_native_rate;
 }
 
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
+
 }  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
 static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
 
 AudioProcessingImpl::ApmSubmoduleStates::ApmSubmoduleStates() {}
 
 bool AudioProcessingImpl::ApmSubmoduleStates::Update(
     bool high_pass_filter_enabled,
     bool echo_canceller_enabled,
@@ skipping 291 matching lines @@
   }
   capture_.capture_audio.reset(
       new AudioBuffer(formats_.api_format.input_stream().num_frames(),
                       formats_.api_format.input_stream().num_channels(),
                       capture_nonlocked_.capture_processing_format.num_frames(),
                       capture_audiobuffer_num_channels,
                       formats_.api_format.output_stream().num_frames()));
 
   public_submodules_->gain_control->Initialize(num_proc_channels(),
                                                proc_sample_rate_hz());
+
   public_submodules_->echo_cancellation->Initialize(
       proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
       num_proc_channels());
+  AllocateRenderQueue();
+
   public_submodules_->echo_control_mobile->Initialize(
       proc_split_sample_rate_hz(), num_reverse_channels(),
       num_output_channels());
   if (constants_.use_experimental_agc) {
     if (!private_submodules_->agc_manager.get()) {
       private_submodules_->agc_manager.reset(new AgcManagerDirect(
           public_submodules_->gain_control.get(),
           public_submodules_->gain_control_for_experimental_agc.get(),
           constants_.agc_startup_min_volume));
     }
@@ skipping 244 matching lines @@
                                        const StreamConfig& output_config,
                                        float* const* dest) {
   TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
   ProcessingConfig processing_config;
   bool reinitialization_required = false;
   {
     // Acquire the capture lock in order to safely call the function
     // that retrieves the render side data. This function accesses apm
     // getters that need the capture lock held when being called.
     rtc::CritScope cs_capture(&crit_capture_);
-    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    EmptyQueuedRenderAudio();
     public_submodules_->echo_control_mobile->ReadQueuedRenderData();
     public_submodules_->gain_control->ReadQueuedRenderData();
 
     if (!src || !dest) {
       return kNullPointerError;
     }
 
     processing_config = formats_.api_format;
     reinitialization_required = UpdateActiveSubmoduleStates();
   }
@@ skipping 39 matching lines @@
       msg->add_output_channel(dest[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
+void AudioProcessingImpl::QueueRenderAudio(const AudioBuffer* audio) {
+  EchoCancellationImpl::PackRenderAudioBuffer(
+      audio, num_output_channels(), num_proc_channels(), &render_queue_buffer_);
+
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());

[ivoc, 2016/10/17 14:11:36]

I think this should actually be RTC_DCHECK_LE(audio->num_frames_per_band(),
160u), to get nicer error messages when the DCHECK fails.

[peah-webrtc, 2016/10/19 05:19:44]

That makes a lot of sense! But I think that currently in the code, the
convention is to write it like this. I have no idea why but I'll keep it like it
is for consistency.

+
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    EmptyQueuedRenderAudio();
+
+    // Retry the insert (should always work).
+    bool result = render_signal_queue_->Insert(&render_queue_buffer_);
+    RTC_DCHECK(result);
+  }
+}
+
+void AudioProcessingImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size =
+      std::max<size_t>(static_cast<size_t>(1),

[ivoc, 2016/10/17 14:11:36]

I think you can remove the <size_t> after std::max.

[peah-webrtc, 2016/10/19 05:19:44]

Done.

+                       kMaxAllowedValuesOfSamplesPerFrame *
+                           EchoCancellationImpl::NumCancellersRequired(
+                               num_output_channels(), num_reverse_channels()));
+
+  // Reallocate the queue if the queue item size is too small to fit the
+  // data to put in the queue.
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+
+    std::vector<float> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<float>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
+}
+
+void AudioProcessingImpl::EmptyQueuedRenderAudio() {
+  rtc::CritScope cs_capture(&crit_capture_);
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    public_submodules_->echo_cancellation->ProcessRenderAudio(
+        capture_queue_buffer_);
+  }
+}
+
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
   {
     // Acquire the capture lock in order to safely call the function
     // that retrieves the render side data. This function accesses apm
     // getters that need the capture lock held when being called.
     // The lock needs to be released as
     // public_submodules_->echo_control_mobile->is_enabled() aquires this lock
     // as well.
     rtc::CritScope cs_capture(&crit_capture_);
-    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    EmptyQueuedRenderAudio();
     public_submodules_->echo_control_mobile->ReadQueuedRenderData();
     public_submodules_->gain_control->ReadQueuedRenderData();
   }
 
   if (!frame) {
     return kNullPointerError;
   }
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
@@ skipping 342 matching lines @@
     render_buffer->SplitIntoFrequencyBands();
   }
 
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
         render_buffer);
   }
 #endif
 
-  RETURN_ON_ERR(
-      public_submodules_->echo_cancellation->ProcessRenderAudio(render_buffer));
+  QueueRenderAudio(render_buffer);
   RETURN_ON_ERR(public_submodules_->echo_control_mobile->ProcessRenderAudio(
       render_buffer));
   if (!constants_.use_experimental_agc) {
     RETURN_ON_ERR(
         public_submodules_->gain_control->ProcessRenderAudio(render_buffer));
   }
 
   if (submodule_states_.RenderMultiBandProcessingActive() &&
       SampleRateSupportsMultiBand(
           formats_.render_processing_format.sample_rate_hz())) {
@@ skipping 449 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc