Add RtcpRttStats to AudioStream

webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h

Index: webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
diff --git a/webrtc/test/fake_videorenderer.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
similarity index 52%
copy from webrtc/test/fake_videorenderer.h
copy to webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
index ff43fc09e475a03dcd1ff9dcef8b934dd03e105d..aeabafa730ff4bcb01f5def54134b94a9e4e592b 100644
--- a/webrtc/test/fake_videorenderer.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
@@ -8,21 +8,19 @@
  *  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));
 
-}  // namespace test
+  MOCK_CONST_METHOD0(LastProcessedRtt, int64_t());
+};
 }  // namespace webrtc
-
-#endif  // WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_