Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/audio/audio_receive_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 168 matching lines @@
   if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
     return stats;
   }
 
   stats.bytes_rcvd = call_stats.bytesReceived;
   stats.packets_rcvd = call_stats.packetsReceived;
   stats.packets_lost = call_stats.cumulativeLost;
   stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
   stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
   if (codec_inst.pltype != -1) {
+    RTC_DCHECK_GE(codec_inst.pltype, 0);
     stats.codec_name = codec_inst.plname;
+    stats.codec_payload_type = rtc::Optional<uint32_t>(codec_inst.pltype);
   }
   stats.ext_seqnum = call_stats.extendedMax;
   if (codec_inst.plfreq / 1000 > 0) {
     stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
   }
   stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
   stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
 
   // Get jitter buffer and total delay (alg + jitter + playout) stats.
   auto ns = channel_proxy_->GetNetworkStatistics();
@@ skipping 100 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc