Move RtpTransportControllerSend to a new file.

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include <string>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ skipping 450 matching lines @@
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
       helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
   send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
 }
 
 }  // namespace test
 }  // namespace webrtc