Configure VoE NACK through AudioSendStream::Config.

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 38 matching lines @@
     bool typing_noise_detected = false;
   };
 
   struct Config {
     Config() = delete;
     explicit Config(Transport* send_transport)
         : send_transport(send_transport) {}
 
     std::string ToString() const;
 
-    // Receive-stream specific RTP settings.
+    // Send-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Sender SSRC.
       uint32_t ssrc = 0;
 
       // RTP header extensions used for the sent stream.
       std::vector<RtpExtension> extensions;
 
+      // See NackConfig for description.
+      NackConfig nack;
+
       // 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;
 
     // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
     // components.
@@ skipping 20 matching lines @@
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_