Wire up transport sequence number / send time callbacks to webrtc via libjingle.

webrtc/call.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef WEBRTC_CALL_H_
 #define WEBRTC_CALL_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
+#include "webrtc/base/socket.h"

[mflodman, 2015/10/15 13:58:22]

Is this really needed in this header?

It would be good to not include these "itnernal" files in an API file like this.

[stefan-webrtc, 2015/10/15 14:08:51]

Keeping as is for now. Chrome already depends on socket.h, so it's not really
internal. This will however, as you pointed out, force other users of Call to
include socket.h, which may be bad. In that case we will follow-up with a CL to
move it into a different or its own .h file.

 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioDeviceModule;
 class AudioProcessing;
 class VoiceEngine;
 class VoiceEngineObserver;
 
@@ skipping 101 matching lines @@
 
   // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
   // of maximum for entire Call. This should be fixed along with the above.
   // Specifying a start bitrate (>0) will currently reset the current bitrate
   // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
   virtual void SignalNetworkState(NetworkState state) = 0;
 
+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
+
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_