Using AudioOption to enable audio network adaptor.

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  Copyright (c) 2014 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 167 matching lines @@
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 
   // Provides target audio bitrate to this encoder to allow it to adapt.
   virtual void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps);
 
   // Provides RTT to this encoder to allow it to adapt.
   virtual void OnReceivedRtt(int rtt_ms);
 
   // To allow encoder to adapt its frame length, it must be provided the frame
-  // length range that receives can accept.
+  // length range that receivers can accept.
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_