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

webrtc/video/stream_synchronization_unittest.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 80 matching lines @@
   bool DelayedStreams(int audio_delay_ms,
                       int video_delay_ms,
                       int current_audio_delay_ms,
                       int* extra_audio_delay_ms,
                       int* total_video_delay_ms) {
     int audio_frequency = static_cast<int>(kDefaultAudioFrequency *
                                            audio_clock_drift_ + 0.5);
     int audio_offset = 0;
     int video_frequency = static_cast<int>(kDefaultVideoFrequency *
                                            video_clock_drift_ + 0.5);
+    bool new_sr;
     int video_offset = 0;
     StreamSynchronization::Measurements audio;
     StreamSynchronization::Measurements video;
     // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP.
-    audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
-                                                   audio_offset));
+    RtcpMeasurement rtcp =
+        send_time_->GenerateRtcp(audio_frequency, audio_offset);
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+                               &audio.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(100);
     receive_time_->IncreaseTimeMs(100);
-    video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
-                                                   video_offset));
+    rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+                               &video.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(900);
     receive_time_->IncreaseTimeMs(900);
-    audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
-                                                   audio_offset));
+    rtcp = send_time_->GenerateRtcp(audio_frequency, audio_offset);
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+                               &audio.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(100);
     receive_time_->IncreaseTimeMs(100);
-    video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
-                                                   video_offset));
+    rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+                               &video.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(900);
     receive_time_->IncreaseTimeMs(900);
 
     // Capture an audio and a video frame at the same time.
     audio.latest_timestamp = send_time_->NowRtp(audio_frequency,
                                                 audio_offset);
     video.latest_timestamp = send_time_->NowRtp(video_frequency,
                                                 video_offset);
 
     if (audio_delay_ms > video_delay_ms) {
@@ skipping 425 matching lines @@
 
 TEST_F(StreamSynchronizationTest,
        BothDelayedVideoLaterVideoClockDriftWithBaseDelay) {
   int base_target_delay_ms = 2000;
   video_clock_drift_ = 1.05;
   sync_->SetTargetBufferingDelay(base_target_delay_ms);
   BothDelayedVideoLaterTest(base_target_delay_ms);
 }
 
 }  // namespace webrtc