Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/api/call/audio_send_stream.cc

 /*
  *  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 12 matching lines @@
   ss << ", channels: " << codec_inst.channels;
   ss << ", rate: " << codec_inst.rate;
   ss << '}';
   return ss.str();
 }
 }  // namespace
 
 namespace webrtc {
 
 AudioSendStream::Stats::Stats() = default;
+AudioSendStream::Stats::~Stats() = default;
 
 AudioSendStream::Config::Config(Transport* send_transport)
     : send_transport(send_transport) {}
 
 AudioSendStream::Config::~Config() = default;
 
 std::string AudioSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
@@ skipping 56 matching lines @@
       enable_opus_dtx == rhs.enable_opus_dtx &&
       opus_max_playback_rate == rhs.opus_max_playback_rate &&
       cng_payload_type == rhs.cng_payload_type &&
       cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
       min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
     return true;
   }
   return false;
 }
 }  // namespace webrtc