Revert of 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/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
deleted file mode 100644
index aeabafa730ff4bcb01f5def54134b94a9e4e592b..0000000000000000000000000000000000000000
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- *  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_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
-#define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
-
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/test/gmock.h"
-
-namespace webrtc {
-
-class MockRtcpRttStats : public RtcpRttStats {
- public:
-  MOCK_METHOD1(OnRttUpdate, void(int64_t rtt));
-
-  MOCK_CONST_METHOD0(LastProcessedRtt, int64_t());
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_