Add stats for frequency offset when converting RTP timestamp to NTP time.

webrtc/video/BUILD.gn

 # Copyright (c) 2014 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.
 
 import("../build/webrtc.gni")
 
@@ skipping 66 matching lines @@
 if (rtc_include_tests) {
   # TODO(pbos): Rename test suite.
   rtc_source_set("video_tests") {
     testonly = true
     sources = [
       "call_stats_unittest.cc",
       "encoder_state_feedback_unittest.cc",
       "end_to_end_tests.cc",
       "overuse_frame_detector_unittest.cc",
       "payload_router_unittest.cc",
+      "receive_statistics_proxy_unittest.cc",
       "report_block_stats_unittest.cc",
       "send_delay_stats_unittest.cc",
       "send_statistics_proxy_unittest.cc",
       "stats_counter_unittest.cc",
       "stream_synchronization_unittest.cc",
       "video_decoder_unittest.cc",
       "video_encoder_unittest.cc",
       "video_send_stream_tests.cc",
       "vie_encoder_unittest.cc",
       "vie_remb_unittest.cc",
     ]
     deps = [
       ":video",
       "//testing/gmock",
       "//testing/gtest",
     ]
     if (is_clang) {
       # Suppress warnings from the Chromium Clang plugin.
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }