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/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
new file mode 100644
index 0000000000000000000000000000000000000000..5f2fae9e736961dbce0d64d35761d902b9719c07
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
@@ -0,0 +1,25 @@
+/*
+ *  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_