Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

webrtc/call/flexfec_receive_stream.h

 /*
  *  Copyright (c) 2016 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 11 matching lines @@
 class FlexfecReceiveStream {
  public:
   struct Stats {
     std::string ToString(int64_t time_ms) const;
 
     // TODO(brandtr): Add appropriate stats here.
     int flexfec_bitrate_bps;
   };
 
   struct Config {
+    explicit Config(Transport* rtcp_send_transport)
+        : rtcp_send_transport(rtcp_send_transport) {}
+
     std::string ToString() const;
 
+    // Returns true if all RTP information is available in order to
+    // enable receiving FlexFEC.
+    bool IsCompleteAndEnabled() const;

[stefan-webrtc, 2016/12/22 09:14:54]

Is this used for anything but tests? If not, I'd prefer to have it part of the
test code.

[brandtr, 2017/01/03 10:01:05]

Yes, it's used here:
https://codereview.webrtc.org/2589713003/diff/280001/webrtc/media/engine/webr...

+
     // Payload type for FlexFEC.
     int payload_type = -1;
 
     // SSRC for FlexFEC stream to be received.
     uint32_t remote_ssrc = 0;
 
     // Vector containing a single element, corresponding to the SSRC of the
     // media stream being protected by this FlexFEC stream. The vector MUST have
     // size 1.
     //
@@ skipping 27 matching lines @@
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~FlexfecReceiveStream() = default;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_