Move RTP keep-alive config from VideoSendStream::Config to Call::Config

webrtc/video/video_send_stream.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.
  */
 
@@ skipping 39 matching lines @@
   VideoSendStream(int num_cpu_cores,
                   ProcessThread* module_process_thread,
                   rtc::TaskQueue* worker_queue,
                   CallStats* call_stats,
                   RtpTransportControllerSendInterface* transport,
                   BitrateAllocator* bitrate_allocator,
                   SendDelayStats* send_delay_stats,
                   RtcEventLog* event_log,
                   VideoSendStream::Config config,
                   VideoEncoderConfig encoder_config,
-                  const std::map<uint32_t, RtpState>& suspended_ssrcs);
+                  const std::map<uint32_t, RtpState>& suspended_ssrcs,
+                  const RtpKeepAliveConfig& keepalive_config);
 
   ~VideoSendStream() override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // webrtc::VideoSendStream implementation.
   void Start() override;
   void Stop() override;
 
@@ skipping 30 matching lines @@
   const VideoSendStream::Config config_;
   const VideoEncoderConfig::ContentType content_type_;
   std::unique_ptr<VideoSendStreamImpl> send_stream_;
   std::unique_ptr<ViEEncoder> vie_encoder_;
 };
 
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_