Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/media/base/mediachannel.h

 /*
  *  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 535 matching lines @@
     } else {
       return 0;
     }
   }
   int64_t bytes_sent;
   int packets_sent;
   int packets_lost;
   float fraction_lost;
   int64_t rtt_ms;
   std::string codec_name;
+  rtc::Optional<int> codec_payload_type;
   std::vector<SsrcSenderInfo> local_stats;
   std::vector<SsrcReceiverInfo> remote_stats;
 };
 
 struct MediaReceiverInfo {
   MediaReceiverInfo()
       : bytes_rcvd(0),
         packets_rcvd(0),
         packets_lost(0),
         fraction_lost(0.0) {
@@ skipping 25 matching lines @@
     } else {
       return 0;
     }
   }
 
   int64_t bytes_rcvd;
   int packets_rcvd;
   int packets_lost;
   float fraction_lost;
   std::string codec_name;
+  rtc::Optional<int> codec_payload_type;
   std::vector<SsrcReceiverInfo> local_stats;
   std::vector<SsrcSenderInfo> remote_stats;
 };
 
 struct VoiceSenderInfo : public MediaSenderInfo {
   VoiceSenderInfo()
       : ext_seqnum(0),
         jitter_ms(0),
         audio_level(0),
         aec_quality_min(0.0),
@@ skipping 80 matching lines @@
         preferred_bitrate(0),
         adapt_reason(0),
         adapt_changes(0),
         avg_encode_ms(0),
         encode_usage_percent(0),
         frames_encoded(0) {}
 
   std::vector<SsrcGroup> ssrc_groups;
   // TODO(hbos): Move this to |VideoMediaInfo::send_codecs|?
   std::string encoder_implementation_name;
-  // TODO(hbos): Move this to |MediaSenderInfo| when supported by
-  // |VoiceSenderInfo| as well (which also extends that class).
-  rtc::Optional<uint32_t> codec_payload_type;
   int packets_cached;
   int firs_rcvd;
   int plis_rcvd;
   int nacks_rcvd;
   int send_frame_width;
   int send_frame_height;
   int framerate_input;
   int framerate_sent;
   int nominal_bitrate;
   int preferred_bitrate;
@@ skipping 25 matching lines @@
         min_playout_delay_ms(0),
         render_delay_ms(0),
         target_delay_ms(0),
         current_delay_ms(0),
         capture_start_ntp_time_ms(-1) {
   }
 
   std::vector<SsrcGroup> ssrc_groups;
   // TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|?
   std::string decoder_implementation_name;
-  // TODO(hbos): Move this to |MediaReceiverInfo| when supported by
-  // |VoiceReceiverInfo| as well (which also extends that class).
-  rtc::Optional<uint32_t> codec_payload_type;
   int packets_concealed;
   int firs_sent;
   int plis_sent;
   int nacks_sent;
   int frame_width;
   int frame_height;
   int framerate_rcvd;
   int framerate_decoded;
   int framerate_output;
   // Framerate as sent to the renderer.
@@ skipping 62 matching lines @@
   int64_t bucket_delay;
 };
 
 // Maps from payload type to |RtpCodecParameters|.
 typedef std::map<int, webrtc::RtpCodecParameters> RtpCodecParametersMap;
 
 struct VoiceMediaInfo {
   void Clear() {
     senders.clear();
     receivers.clear();
+    send_codecs.clear();
+    receive_codecs.clear();
   }
   std::vector<VoiceSenderInfo> senders;
   std::vector<VoiceReceiverInfo> receivers;
+  RtpCodecParametersMap send_codecs;
+  RtpCodecParametersMap receive_codecs;
 };
 
 struct VideoMediaInfo {
   void Clear() {
     senders.clear();
     receivers.clear();
     bw_estimations.clear();
     send_codecs.clear();
     receive_codecs.clear();
   }
@@ skipping 331 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_