Expose RtpCodecParameters to VideoMediaInfo 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 676 matching lines @@
         framerate_sent(0),
         nominal_bitrate(0),
         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 23 matching lines @@
         max_decode_ms(0),
         jitter_buffer_ms(0),
         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 55 matching lines @@
 
   int available_send_bandwidth;
   int available_recv_bandwidth;
   int target_enc_bitrate;
   int actual_enc_bitrate;
   int retransmit_bitrate;
   int transmit_bitrate;
   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();
   }
   std::vector<VoiceSenderInfo> senders;
   std::vector<VoiceReceiverInfo> receivers;
 };
 
 struct VideoMediaInfo {
   void Clear() {
     senders.clear();
     receivers.clear();
     bw_estimations.clear();
+    send_codecs.clear();
+    receive_codecs.clear();
   }
   std::vector<VideoSenderInfo> senders;
   std::vector<VideoReceiverInfo> receivers;
   std::vector<BandwidthEstimationInfo> bw_estimations;
+  RtpCodecParametersMap send_codecs;
+  RtpCodecParametersMap receive_codecs;
 };
 
 struct DataMediaInfo {
   void Clear() {
     senders.clear();
     receivers.clear();
   }
   std::vector<DataSenderInfo> senders;
   std::vector<DataReceiverInfo> receivers;
 };
@@ skipping 316 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_