Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 9f544c18262d13e812186b46e9a8f79277077bf3..2f12bba4d8d3d56814900bc1a7bbf0f04c9d4e28 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2475,7 +2475,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   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;
@@ -2485,6 +2485,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     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;
@@ -2500,7 +2501,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   }
 
   // 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;
@@ -2510,6 +2511,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     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;
@@ -2533,6 +2535,22 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     info->receivers.push_back(rinfo);
   }
 
+  // Get codec info
+  for (const AudioCodec& codec : send_codecs_) {
+    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
+    RTC_DCHECK_GE(codec_params.payload_type, 0);

[the sun, 2016/11/16 19:41:07]

Same comment as for video engine - can we avoid the DCHECKs and casting here?

[hbos, 2016/11/17 14:14:48]

Done.

+    info->send_codecs.insert(
+        std::make_pair(static_cast<uint32_t>(codec_params.payload_type),
+                       std::move(codec_params)));
+  }
+  for (const AudioCodec& codec : recv_codecs_) {
+    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
+    RTC_DCHECK_GE(codec_params.payload_type, 0);
+    info->receive_codecs.insert(
+        std::make_pair(static_cast<uint32_t>(codec_params.payload_type),
+                       std::move(codec_params)));
+  }
+
   return true;
 }