Configure VoE NACK through AudioSendStream::Config.

webrtc/config.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 // TODO(pbos): Move Config from common.h to here.
 
 #ifndef WEBRTC_CONFIG_H_
 #define WEBRTC_CONFIG_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Settings for NACK, see RFC 4585 for details.
 struct NackConfig {
   NackConfig() : rtp_history_ms(0) {}
+  std::string ToString() const;
   // Send side: the time RTP packets are stored for retransmissions.
   // Receive side: the time the receiver is prepared to wait for
   // retransmissions.
   // Set to '0' to disable.
   int rtp_history_ms;
 };
 
 // Settings for forward error correction, see RFC 5109 for details. Set the
 // payload types to '-1' to disable.
 struct FecConfig {
@@ skipping 131 matching lines @@
 struct VoicePacing {
   VoicePacing() : enabled(false) {}
   explicit VoicePacing(bool value) : enabled(value) {}
   static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
   bool enabled;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CONFIG_H_