Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 2457 matching lines @@
   call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
                                     transport_overhead_per_packet);
 }
 
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(info);
 
   // Get SSRC and stats for each sender.
-  RTC_DCHECK(info->senders.size() == 0);
+  RTC_DCHECK_EQ(info->senders.size(), 0U);
   for (const auto& stream : send_streams_) {
     webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
     VoiceSenderInfo sinfo;
     sinfo.add_ssrc(stats.local_ssrc);
     sinfo.bytes_sent = stats.bytes_sent;
     sinfo.packets_sent = stats.packets_sent;
     sinfo.packets_lost = stats.packets_lost;
     sinfo.fraction_lost = stats.fraction_lost;
     sinfo.codec_name = stats.codec_name;
+    sinfo.codec_payload_type = stats.codec_payload_type;
     sinfo.ext_seqnum = stats.ext_seqnum;
     sinfo.jitter_ms = stats.jitter_ms;
     sinfo.rtt_ms = stats.rtt_ms;
     sinfo.audio_level = stats.audio_level;
     sinfo.aec_quality_min = stats.aec_quality_min;
     sinfo.echo_delay_median_ms = stats.echo_delay_median_ms;
     sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
     sinfo.echo_return_loss = stats.echo_return_loss;
     sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
     sinfo.residual_echo_likelihood = stats.residual_echo_likelihood;
     sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
     info->senders.push_back(sinfo);
   }
 
   // Get SSRC and stats for each receiver.
-  RTC_DCHECK(info->receivers.size() == 0);
+  RTC_DCHECK_EQ(info->receivers.size(), 0U);
   for (const auto& stream : recv_streams_) {
     webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
     VoiceReceiverInfo rinfo;
     rinfo.add_ssrc(stats.remote_ssrc);
     rinfo.bytes_rcvd = stats.bytes_rcvd;
     rinfo.packets_rcvd = stats.packets_rcvd;
     rinfo.packets_lost = stats.packets_lost;
     rinfo.fraction_lost = stats.fraction_lost;
     rinfo.codec_name = stats.codec_name;
+    rinfo.codec_payload_type = stats.codec_payload_type;
     rinfo.ext_seqnum = stats.ext_seqnum;
     rinfo.jitter_ms = stats.jitter_ms;
     rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
     rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
     rinfo.delay_estimate_ms = stats.delay_estimate_ms;
     rinfo.audio_level = stats.audio_level;
     rinfo.expand_rate = stats.expand_rate;
     rinfo.speech_expand_rate = stats.speech_expand_rate;
     rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
     rinfo.accelerate_rate = stats.accelerate_rate;
     rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
     rinfo.decoding_calls_to_silence_generator =
         stats.decoding_calls_to_silence_generator;
     rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
     rinfo.decoding_normal = stats.decoding_normal;
     rinfo.decoding_plc = stats.decoding_plc;
     rinfo.decoding_cng = stats.decoding_cng;
     rinfo.decoding_plc_cng = stats.decoding_plc_cng;
     rinfo.decoding_muted_output = stats.decoding_muted_output;
     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
     info->receivers.push_back(rinfo);
   }
 
+  // Get codec info
+  for (const AudioCodec& codec : send_codecs_) {
+    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
+    info->send_codecs.insert(
+        std::make_pair(codec_params.payload_type, std::move(codec_params)));
+  }
+  for (const AudioCodec& codec : recv_codecs_) {
+    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
+    info->receive_codecs.insert(
+        std::make_pair(codec_params.payload_type, std::move(codec_params)));
+  }
+
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetRawAudioSink(
     uint32_t ssrc,
     std::unique_ptr<webrtc::AudioSinkInterface> sink) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
                   << " " << (sink ? "(ptr)" : "NULL");
   if (ssrc == 0) {
@@ skipping 32 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE