Finalization of the first version of EchoCanceller 3

webrtc/modules/audio_processing/aec3/echo_canceller3.cc

 /*
  *  Copyright (c) 2016 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.
  */
 #include "webrtc/modules/audio_processing/aec3/echo_canceller3.h"
 
 #include <sstream>
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 
 namespace webrtc {
 
 namespace {
 
 bool DetectSaturation(rtc::ArrayView<const float> y) {
   for (auto y_k : y) {
-    if (y_k >= 32767.0f || y_k <= -32768.0f) {
+    if (y_k >= 32700.0f || y_k <= -32700.0f) {
       return true;
     }
   }
   return false;
 }
 
 void FillSubFrameView(AudioBuffer* frame,
                       size_t sub_frame_index,
                       std::vector<rtc::ArrayView<float>>* sub_frame_view) {
   RTC_DCHECK_GE(1, sub_frame_index);
@@ skipping 70 matching lines @@
     return true;
   }
   render_blocker->ExtractBlock(block);
   return block_processor->BufferRender(block);
 }
 
 void CopyAudioBufferIntoFrame(AudioBuffer* buffer,
                               size_t num_bands,
                               size_t frame_length,
                               std::vector<std::vector<float>>* frame) {
   RTC_DCHECK_EQ(num_bands, frame->size());

[aleloi, 2017/02/23 10:56:45]

Suggest renaming into 'CopyLowestBandIntoFrame' or similar. Also, please add a
DCHECK for (*frame)[0].size().

[peah-webrtc, 2017/02/23 11:18:33]

Done.

-  for (size_t i = 0; i < num_bands; ++i) {
-    rtc::ArrayView<float> buffer_view(&buffer->split_bands_f(0)[i][0],
-                                      frame_length);
-    std::copy(buffer_view.begin(), buffer_view.end(), (*frame)[i].begin());
-  }
+  rtc::ArrayView<float> buffer_view(&buffer->channels_f()[0][0], frame_length);
+  std::copy(buffer_view.begin(), buffer_view.end(), (*frame)[0].begin());
 }
 
 // [B,A] = butter(2,100/4000,'high')
 const CascadedBiQuadFilter::BiQuadCoefficients
     kHighPassFilterCoefficients_8kHz = {{0.94598f, -1.89195f, 0.94598f},
                                         {-1.88903f, 0.89487f}};
 const int kNumberOfHighPassBiQuads_8kHz = 1;
 
 // [B,A] = butter(2,100/8000,'high')
 const CascadedBiQuadFilter::BiQuadCoefficients
@@ skipping 45 matching lines @@
       render_queue_input_frame_(num_bands_,
                                 std::vector<float>(frame_length_, 0.f)),
       render_transfer_queue_(render_transfer_queue) {
   RTC_DCHECK(data_dumper);
 }
 
 EchoCanceller3::RenderWriter::~RenderWriter() = default;
 
 bool EchoCanceller3::RenderWriter::Insert(AudioBuffer* input) {
   RTC_DCHECK_EQ(1, input->num_channels());
-  RTC_DCHECK_EQ(num_bands_, input->num_bands());
   RTC_DCHECK_EQ(frame_length_, input->num_frames_per_band());
   data_dumper_->DumpWav("aec3_render_input", frame_length_,
-                        &input->split_bands_f(0)[0][0],
+                        &input->channels_f()[0][0],
                         LowestBandRate(sample_rate_hz_), 1);
 
   CopyAudioBufferIntoFrame(input, num_bands_, frame_length_,
                            &render_queue_input_frame_);
 
   if (render_highpass_filter_) {
     render_highpass_filter_->Process(render_queue_input_frame_[0]);
   }
 
   return render_transfer_queue_->Insert(&render_queue_input_frame_);
@@ skipping 156 matching lines @@
                                           block_processor_.get(), &block_) &&
         successful_buffering;
 
     frame_to_buffer =
         render_transfer_queue_.Remove(&render_queue_output_frame_);
   }
   return successful_buffering;
 }
 
 }  // namespace webrtc