Adding activation logic of the new APM level control functionality using MediaConstraints

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 @@
     SetFrom(&audio_jitter_buffer_fast_accelerate,
             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(&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);
   }
 
   bool operator==(const AudioOptions& o) const {
     return echo_cancellation == o.echo_cancellation &&
-        auto_gain_control == o.auto_gain_control &&
-        noise_suppression == o.noise_suppression &&
-        highpass_filter == o.highpass_filter &&
-        stereo_swapping == o.stereo_swapping &&
-        audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
-        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 &&
-        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;
+           auto_gain_control == o.auto_gain_control &&
+           noise_suppression == o.noise_suppression &&
+           highpass_filter == o.highpass_filter &&
+           stereo_swapping == o.stereo_swapping &&
+           audio_jitter_buffer_max_packets ==
+               o.audio_jitter_buffer_max_packets &&
+           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 &&
+           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;
   }
   bool operator!=(const AudioOptions& o) const { return !(*this == o); }
 
   std::string ToString() const {
     std::ostringstream ost;
     ost << "AudioOptions {";
     ost << ToStringIfSet("aec", echo_cancellation);
     ost << ToStringIfSet("agc", auto_gain_control);
     ost << ToStringIfSet("ns", noise_suppression);
     ost << ToStringIfSet("hf", highpass_filter);
     ost << ToStringIfSet("swap", stereo_swapping);
     ost << ToStringIfSet("audio_jitter_buffer_max_packets",
                          audio_jitter_buffer_max_packets);
     ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
                          audio_jitter_buffer_fast_accelerate);
     ost << ToStringIfSet("typing", typing_detection);
     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("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 << "}";
     return ost.str();
   }
@@ skipping 15 matching lines @@
   rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
   // Audio processing to detect typing.
   rtc::Optional<bool> typing_detection;
   rtc::Optional<bool> aecm_generate_comfort_noise;
   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;
   // Note that tx_agc_* only applies to non-experimental AGC.
   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.
@@ skipping 863 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_