Reland "Update rtt on audio only calls".

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/common_audio/mocks/mock_smoothing_filter.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
similarity index 56%
copy from webrtc/common_audio/mocks/mock_smoothing_filter.h
copy to webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
index 6b2991b22f91d736fa1e5849fc94d5b8573716db..aeabafa730ff4bcb01f5def54134b94a9e4e592b 100644
--- a/webrtc/common_audio/mocks/mock_smoothing_filter.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
@@ -8,20 +8,19 @@
  *  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_