Configure VoE NACK through AudioSendStream::Config.

webrtc/config.cc

 /*
  *  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.
  */
 #include "webrtc/config.h"
 
 #include <sstream>
 #include <string>
 
 namespace webrtc {
+std::string NackConfig::ToString() const {
+  std::stringstream ss;
+  ss << "{rtp_history_ms: " << rtp_history_ms;
+  ss << '}';
+  return ss.str();
+}
+
 std::string FecConfig::ToString() const {
   std::stringstream ss;
   ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
   ss << '}';
   return ss.str();
 }
 
 std::string RtpExtension::ToString() const {
@@ skipping 107 matching lines @@
   }
   ss << ", encoder_specific_settings: ";
   ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
 
   ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
   ss << '}';
   return ss.str();
 }
 
 }  // namespace webrtc