Allow mocking SendSideCongestionController for Call tests.

webrtc/call/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_CALL_H_
 #define WEBRTC_CALL_CALL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/call/audio_receive_stream.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/flexfec_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 class RtcEventLog;
 
 enum class MediaType {
@@ skipping 58 matching lines @@
 
     int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
     int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
     int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
     int64_t pacer_delay_ms = 0;
     int64_t rtt_ms = -1;
   };
 
   static Call* Create(const Call::Config& config);
 
+  // Allows mocking |transport_send| for testing.
+  static Call* Create(
+      const Call::Config& config,
+      std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
+
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
 
   virtual AudioReceiveStream* CreateAudioReceiveStream(
       const AudioReceiveStream::Config& config) = 0;
   virtual void DestroyAudioReceiveStream(
       AudioReceiveStream* receive_stream) = 0;
 
   virtual VideoSendStream* CreateVideoSendStream(
@@ skipping 46 matching lines @@
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_