Add RtcpRttStats to AudioStream

webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.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.
  */
 
-#ifndef WEBRTC_TEST_FAKE_VIDEORENDERER_H_
-#define WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+#define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
 
-#include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/video_frame.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
-namespace test {
 
-class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+class MockRtcpRttStats : public RtcpRttStats {
  public:
-  void OnFrame(const webrtc::VideoFrame& frame) override {}
+  MOCK_METHOD1(OnRttUpdate, void(int64_t rtt));
+
+  MOCK_CONST_METHOD0(LastProcessedRtt, int64_t());
 };
-
-}  // namespace test
 }  // namespace webrtc
-
-#endif  // WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_