Add AudioOption for residual echo detector, and enable it by default on non-mobile platforms.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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 140 matching lines @@
             change.audio_jitter_buffer_fast_accelerate);
     SetFrom(&typing_detection, change.typing_detection);
     SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
     SetFrom(&adjust_agc_delta, change.adjust_agc_delta);
     SetFrom(&experimental_agc, change.experimental_agc);
     SetFrom(&extended_filter_aec, change.extended_filter_aec);
     SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
     SetFrom(&experimental_ns, change.experimental_ns);
     SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
     SetFrom(&level_control, change.level_control);
+    SetFrom(&residual_echo_detector, change.residual_echo_detector);
     SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
     SetFrom(&tx_agc_digital_compression_gain,
             change.tx_agc_digital_compression_gain);
     SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
     SetFrom(&recording_sample_rate, change.recording_sample_rate);
     SetFrom(&playout_sample_rate, change.playout_sample_rate);
     SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
     SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
     SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
     SetFrom(&level_control_initial_peak_level_dbfs,
@@ skipping 11 matching lines @@
            audio_jitter_buffer_fast_accelerate ==
                o.audio_jitter_buffer_fast_accelerate &&
            typing_detection == o.typing_detection &&
            aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
            experimental_agc == o.experimental_agc &&
            extended_filter_aec == o.extended_filter_aec &&
            delay_agnostic_aec == o.delay_agnostic_aec &&
            experimental_ns == o.experimental_ns &&
            intelligibility_enhancer == o.intelligibility_enhancer &&
            level_control == o.level_control &&
+           residual_echo_detector == o.residual_echo_detector &&
            adjust_agc_delta == o.adjust_agc_delta &&
            tx_agc_target_dbov == o.tx_agc_target_dbov &&
            tx_agc_digital_compression_gain ==
                o.tx_agc_digital_compression_gain &&
            tx_agc_limiter == o.tx_agc_limiter &&
            recording_sample_rate == o.recording_sample_rate &&
            playout_sample_rate == o.playout_sample_rate &&
            combined_audio_video_bwe == o.combined_audio_video_bwe &&
            audio_network_adaptor == o.audio_network_adaptor &&
            audio_network_adaptor_config == o.audio_network_adaptor_config &&
@@ skipping 18 matching lines @@
     ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
     ost << ToStringIfSet("agc_delta", adjust_agc_delta);
     ost << ToStringIfSet("experimental_agc", experimental_agc);
     ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
     ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
     ost << ToStringIfSet("experimental_ns", experimental_ns);
     ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
     ost << ToStringIfSet("level_control", level_control);
     ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
                          level_control_initial_peak_level_dbfs);
+    ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
     ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
     ost << ToStringIfSet("tx_agc_digital_compression_gain",
         tx_agc_digital_compression_gain);
     ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
     ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
     ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
     ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
     ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
     // The adaptor config is a serialized proto buffer and therefore not human
     // readable. So we comment out the following line.
@@ skipping 24 matching lines @@
   rtc::Optional<int> adjust_agc_delta;
   rtc::Optional<bool> experimental_agc;
   rtc::Optional<bool> extended_filter_aec;
   rtc::Optional<bool> delay_agnostic_aec;
   rtc::Optional<bool> experimental_ns;
   rtc::Optional<bool> intelligibility_enhancer;
   rtc::Optional<bool> level_control;
   // Specifies an optional initialization value for the level controller.
   rtc::Optional<float> level_control_initial_peak_level_dbfs;
   // Note that tx_agc_* only applies to non-experimental AGC.
+  rtc::Optional<bool> residual_echo_detector;
   rtc::Optional<uint16_t> tx_agc_target_dbov;
   rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
   rtc::Optional<bool> tx_agc_limiter;
   rtc::Optional<uint32_t> recording_sample_rate;
   rtc::Optional<uint32_t> playout_sample_rate;
   // Enable combined audio+bandwidth BWE.
   // TODO(pthatcher): This flag is set from the
   // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
   // and check if any other AudioOptions members are unused.
   rtc::Optional<bool> combined_audio_video_bwe;
@@ skipping 877 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_