Added the ability to adjust the AEC3 performance for large rooms

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 182 matching lines @@
 
   if (render_highpass_filter_) {
     render_highpass_filter_->Process(render_queue_input_frame_[0]);
   }
 
   static_cast<void>(render_transfer_queue_->Insert(&render_queue_input_frame_));
 }
 
 int EchoCanceller3::instance_count_ = 0;
 
-EchoCanceller3::EchoCanceller3(int sample_rate_hz, bool use_highpass_filter)
-    : EchoCanceller3(sample_rate_hz,
+EchoCanceller3::EchoCanceller3(
+    const AudioProcessing::Config::EchoCanceller3& config,
+    int sample_rate_hz,
+    bool use_highpass_filter)
+    : EchoCanceller3(config,
+                     sample_rate_hz,
                      use_highpass_filter,
                      std::unique_ptr<BlockProcessor>(
-                         BlockProcessor::Create(sample_rate_hz))) {}
-EchoCanceller3::EchoCanceller3(int sample_rate_hz,
-                               bool use_highpass_filter,
-                               std::unique_ptr<BlockProcessor> block_processor)
+                         BlockProcessor::Create(config, sample_rate_hz))) {}
+EchoCanceller3::EchoCanceller3(
+    const AudioProcessing::Config::EchoCanceller3& config,

[ivoc, 2017/06/30 11:40:03]

This argument is also not used, Is it needed?

[peah-webrtc, 2017/06/30 13:20:16]

No, you are totally right. Good point and find!

Done.

+    int sample_rate_hz,
+    bool use_highpass_filter,
+    std::unique_ptr<BlockProcessor> block_processor)
     : data_dumper_(
           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
       sample_rate_hz_(sample_rate_hz),
       num_bands_(NumBandsForRate(sample_rate_hz_)),
       frame_length_(rtc::CheckedDivExact(LowestBandRate(sample_rate_hz_), 100)),
       output_framer_(num_bands_),
       capture_blocker_(num_bands_),
       render_blocker_(num_bands_),
       render_transfer_queue_(
           kRenderTransferQueueSize,
@@ skipping 103 matching lines @@
 std::string EchoCanceller3::ToString(
     const AudioProcessing::Config::EchoCanceller3& config) {
   std::stringstream ss;
   ss << "{"
      << "enabled: " << (config.enabled ? "true" : "false") << "}";
   return ss.str();
 }
 
 bool EchoCanceller3::Validate(
     const AudioProcessing::Config::EchoCanceller3& config) {
-  return true;
+  return (config.echo_decay >= 0.f && config.echo_decay < 1.f);
 }
 
 void EchoCanceller3::EmptyRenderQueue() {
   RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
   bool frame_to_buffer =
       render_transfer_queue_.Remove(&render_queue_output_frame_);
   while (frame_to_buffer) {
     BufferRenderFrameContent(&render_queue_output_frame_, 0, &render_blocker_,
                              block_processor_.get(), &block_, &sub_frame_view_);
 
     if (sample_rate_hz_ != 8000) {
       BufferRenderFrameContent(&render_queue_output_frame_, 1, &render_blocker_,
                                block_processor_.get(), &block_,
                                &sub_frame_view_);
     }
 
     BufferRemainingRenderFrameContent(&render_blocker_, block_processor_.get(),
                                       &block_);
 
     frame_to_buffer =
         render_transfer_queue_.Remove(&render_queue_output_frame_);
   }
 }
 
 }  // namespace webrtc