Prepare the AudioSendStream to be hooked up to send-side BWE.

webrtc/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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 46 matching lines @@
 
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Sender SSRC.
       uint32_t ssrc = 0;
 
-      // RTP header extensions used for the received stream.
+      // RTP header extensions used for the sent stream.
       std::vector<RtpExtension> extensions;
 
       // RTCP CNAME, see RFC 3550.
       std::string c_name;
     } rtp;
 
     // Transport for outgoing packets. The transport is expected to exist for
     // the entire life of the AudioSendStream and is owned by the API client.
     Transport* send_transport = nullptr;
 
@@ skipping 12 matching lines @@
   };
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
                                   uint32_t duration_ms) = 0;
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_