Further tuning for AEC3 for initial echo suppression and handling of echo path changes

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"
@@ skipping 242 matching lines @@
 
 bool EchoCanceller3::AnalyzeRender(AudioBuffer* render) {
   RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_);
   RTC_DCHECK(render);
   return render_writer_->Insert(render);
 }
 
 void EchoCanceller3::AnalyzeCapture(AudioBuffer* capture) {
   RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
   RTC_DCHECK(capture);
-  data_dumper_->DumpWav("aec3_capture_analyze_input", frame_length_,
+  data_dumper_->DumpWav("aec3_capture_analyze_input", capture->num_frames(),

[peah-webrtc, 2017/02/27 14:26:03]

The framelength was wrong here before.

                         capture->channels_f()[0], sample_rate_hz_, 1);
 
   saturated_microphone_signal_ = false;
   for (size_t k = 0; k < capture->num_channels(); ++k) {
     saturated_microphone_signal_ |=
         DetectSaturation(rtc::ArrayView<const float>(capture->channels_f()[k],
                                                      capture->num_frames()));
     if (saturated_microphone_signal_) {
       break;
     }
@@ skipping 78 matching lines @@
                                           block_processor_.get(), &block_) &&
         successful_buffering;
 
     frame_to_buffer =
         render_transfer_queue_.Remove(&render_queue_output_frame_);
   }
   return successful_buffering;
 }
 
 }  // namespace webrtc