Reland "Update rtt on audio only calls".

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_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
-#define WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_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/common_audio/smoothing_filter.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
 
-class MockSmoothingFilter : public SmoothingFilter {
+class MockRtcpRttStats : public RtcpRttStats {
  public:
-  MOCK_METHOD1(AddSample, void(float));
-  MOCK_CONST_METHOD0(GetAverage, rtc::Optional<float>());
+  MOCK_METHOD1(OnRttUpdate, void(int64_t rtt));
+

[the sun, 2016/11/28 13:17:55]

nit: dd

[michaelt, 2016/11/28 13:47:35]

Done.

+  MOCK_CONST_METHOD0(LastProcessedRtt, int64_t());
 };
-
 }  // namespace webrtc
-
-#endif  // WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_