Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/audio/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 153 matching lines @@
     stats.rtt_ms = call_stats.rttMs;
   }
   // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
   //                  implementation.
   stats.aec_quality_min = -1;
 
   webrtc::CodecInst codec_inst = {0};
   if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
     RTC_DCHECK_NE(codec_inst.pltype, -1);
     stats.codec_name = codec_inst.plname;
+    stats.codec_payload_type = rtc::Optional<int>(codec_inst.pltype);
 
     // Get data from the last remote RTCP report.
     for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
       // Lookup report for send ssrc only.
       if (block.source_SSRC == stats.local_ssrc) {
         stats.packets_lost = block.cumulative_num_packets_lost;
         stats.fraction_lost = Q8ToFloat(block.fraction_lost);
         stats.ext_seqnum = block.extended_highest_sequence_number;
         // Convert samples to milliseconds.
         if (codec_inst.plfreq / 1000 > 0) {
@@ skipping 193 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc