Integrate FlexfecReceiveStream with Call.

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/api/call/audio_receive_stream.h"
 #include "webrtc/api/call/audio_send_stream.h"
 #include "webrtc/api/call/audio_state.h"
+#include "webrtc/api/call/flexfec_receive_stream.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
@@ skipping 95 matching lines @@
   virtual VideoSendStream* CreateVideoSendStream(
       VideoSendStream::Config config,
       VideoEncoderConfig encoder_config) = 0;
   virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
 
   virtual VideoReceiveStream* CreateVideoReceiveStream(
       VideoReceiveStream::Config configuration) = 0;
   virtual void DestroyVideoReceiveStream(
       VideoReceiveStream* receive_stream) = 0;
 
+  virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      FlexfecReceiveStream::Config configuration) = 0;
+  virtual void DestroyFlexfecReceiveStream(
+      FlexfecReceiveStream* receive_stream) = 0;
+
   // All received RTP and RTCP packets for the call should be inserted to this
   // PacketReceiver. The PacketReceiver pointer is valid as long as the
   // Call instance exists.
   virtual PacketReceiver* Receiver() = 0;
 
   // Returns the call statistics, such as estimated send and receive bandwidth,
   // pacing delay, etc.
   virtual Stats GetStats() const = 0;
 
   // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
@@ skipping 15 matching lines @@
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_