Moved the AECM 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 690 matching lines @@
                                        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_);
     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 41 matching lines @@
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 void AudioProcessingImpl::QueueRenderAudio(const AudioBuffer* audio) {
   EchoCancellationImpl::PackRenderAudioBuffer(audio, num_output_channels(),
                                               num_reverse_channels(),
-                                              &render_queue_buffer_);
+                                              &float_render_queue_buffer_);
 
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
 
   // Insert the samples into the queue.
-  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+  if (!float_render_signal_queue_->Insert(&float_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_);
+    bool result =
+        float_render_signal_queue_->Insert(&float_render_queue_buffer_);
+    RTC_DCHECK(result);
+  }
+
+  EchoControlMobileImpl::PackRenderAudioBuffer(audio, num_output_channels(),
+                                               num_reverse_channels(),
+                                               &int16_render_queue_buffer_);
+
+  // Insert the samples into the queue.
+  if (!int16_render_signal_queue_->Insert(&int16_render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    EmptyQueuedRenderAudio();
+
+    // Retry the insert (should always work).
+    bool result =
+        int16_render_signal_queue_->Insert(&int16_render_queue_buffer_);
     RTC_DCHECK(result);
   }
 }
 
 void AudioProcessingImpl::AllocateRenderQueue() {
-  const size_t new_render_queue_element_max_size =
+  const size_t new_float_render_queue_element_max_size =
       std::max(static_cast<size_t>(1),
                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;
+  const size_t new_int16_render_queue_element_max_size =
+      std::max(static_cast<size_t>(1),
+               kMaxAllowedValuesOfSamplesPerFrame *
+                   EchoControlMobileImpl::NumCancellersRequired(
+                       num_output_channels(), num_reverse_channels()));
 
-    std::vector<float> template_queue_element(render_queue_element_max_size_);
+  // Reallocate the queues if the queue item sizes are too small to fit the
+  // data to put in the queues.
+  if (float_render_queue_element_max_size_ <
+      new_float_render_queue_element_max_size) {
+    float_render_queue_element_max_size_ =
+        new_float_render_queue_element_max_size;
 
-    render_signal_queue_.reset(
+    std::vector<float> template_queue_element(
+        float_render_queue_element_max_size_);
+
+    float_render_signal_queue_.reset(
         new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
             kMaxNumFramesToBuffer, template_queue_element,
-            RenderQueueItemVerifier<float>(render_queue_element_max_size_)));
+            RenderQueueItemVerifier<float>(
+                float_render_queue_element_max_size_)));
 
-    render_queue_buffer_.resize(render_queue_element_max_size_);
-    capture_queue_buffer_.resize(render_queue_element_max_size_);
+    float_render_queue_buffer_.resize(float_render_queue_element_max_size_);
+    float_capture_queue_buffer_.resize(float_render_queue_element_max_size_);
   } else {
-    render_signal_queue_->Clear();
+    float_render_signal_queue_->Clear();
+  }
+
+  if (int16_render_queue_element_max_size_ <
+      new_int16_render_queue_element_max_size) {
+    int16_render_queue_element_max_size_ =
+        new_int16_render_queue_element_max_size;
+
+    std::vector<int16_t> template_queue_element(
+        int16_render_queue_element_max_size_);
+
+    int16_render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(
+                int16_render_queue_element_max_size_)));
+
+    int16_render_queue_buffer_.resize(int16_render_queue_element_max_size_);
+    int16_capture_queue_buffer_.resize(int16_render_queue_element_max_size_);
+  } else {
+    int16_render_signal_queue_->Clear();
   }
 }
 
 void AudioProcessingImpl::EmptyQueuedRenderAudio() {
   rtc::CritScope cs_capture(&crit_capture_);
-  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+  while (float_render_signal_queue_->Remove(&float_capture_queue_buffer_)) {
     public_submodules_->echo_cancellation->ProcessRenderAudio(
-        capture_queue_buffer_);
+        float_capture_queue_buffer_);
+  }
+
+  while (int16_render_signal_queue_->Remove(&int16_capture_queue_buffer_)) {
+    public_submodules_->echo_control_mobile->ProcessRenderAudio(
+        int16_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_);
     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 &&
       frame->sample_rate_hz_ != kSampleRate32kHz &&
@@ skipping 342 matching lines @@
   }
 
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
         render_buffer);
   }
 #endif
 
   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())) {
     render_buffer->MergeFrequencyBands();
   }
@@ skipping 447 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc