Split webrtc/video into webrtc/{audio,call,video}.

webrtc/audio/audio_receive_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 "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/video/audio_receive_stream.h"
 
 namespace webrtc {
 
 const size_t kAbsoluteSendTimeLength = 4;
 
 void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
                                     int id,
                                     uint32_t abs_send_time) {
   const size_t kRtpOneByteHeaderLength = 4;
   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
@@ skipping 42 matching lines @@
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
   EXPECT_CALL(rbe, IncomingPacket(packet_time.timestamp / 1000,
                                   sizeof(rtp_packet) - kExpectedHeaderLength,
                                   testing::_, false))
       .Times(1);
   EXPECT_TRUE(
       recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
 }
 }  // namespace webrtc