Add aec-refined-adaptive-filter command line switch

content/renderer/media/media_stream_audio_processor.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/media_stream_audio_processor.h"
 
 #include <stddef.h>
 #include <stdint.h>
 #include <utility>
 
@@ skipping 99 matching lines @@
 // on the command line.
 bool GetStartupMinVolumeForAgc(int* startup_min_volume) {
   DCHECK(startup_min_volume);
   std::string min_volume_str(
       base::CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
           switches::kAgcStartupMinVolume));
   return !min_volume_str.empty() &&
          base::StringToInt(min_volume_str, startup_min_volume);
 }
 
+// Checks if the AEC's refined adaptive filter tuning was enabled on the command
+// line.
+bool UseAecRefinedAdaptiveFilter() {
+  return base::CommandLine::ForCurrentProcess()->HasSwitch(
+      switches::kAecRefinedAdaptiveFilter);
+}
+
 }  // namespace
 
 // Wraps AudioBus to provide access to the array of channel pointers, since this
 // is the type webrtc::AudioProcessing deals in. The array is refreshed on every
 // channel_ptrs() call, and will be valid until the underlying AudioBus pointers
 // are changed, e.g. through calls to SetChannelData() or SwapChannels().
 //
 // All methods are called on one of the capture or render audio threads
 // exclusively.
 class MediaStreamAudioBus {
@@ skipping 392 matching lines @@
   }
 
   // Experimental options provided at creation.
   webrtc::Config config;
   config.Set<webrtc::ExtendedFilter>(
       new webrtc::ExtendedFilter(goog_experimental_aec));
   config.Set<webrtc::ExperimentalNs>(
       new webrtc::ExperimentalNs(goog_experimental_ns));
   if (IsDelayAgnosticAecEnabled())
     config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(true));
+  if (UseAecRefinedAdaptiveFilter()) {
+    config.Set<webrtc::RefinedAdaptiveFilter>(
+        new webrtc::RefinedAdaptiveFilter(true));
+  }
   if (goog_beamforming) {
     const auto& geometry =
         GetArrayGeometryPreferringConstraints(audio_constraints, input_params);
 
     // Only enable beamforming if we have at least two mics.
     config.Set<webrtc::Beamforming>(
         new webrtc::Beamforming(geometry.size() > 1, geometry));
   }
 
   // If the experimental AGC is enabled, check for overridden config params.
@@ skipping 199 matching lines @@
   if (echo_information_) {
     echo_information_.get()->UpdateAecDelayStats(ap->echo_cancellation());
   }
 
   // Return 0 if the volume hasn't been changed, and otherwise the new volume.
   return (agc->stream_analog_level() == volume) ?
       0 : agc->stream_analog_level();
 }
 
 }  // namespace content