Make class of static functions in rtp_to_ntp.h:

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 16 matching lines @@
 enum { kDefaultVideoFrequency = 90000 };
 const double kNtpFracPerMs = 4.294967296E6;
 static const int kSmoothingFilter = 4 * 2;
 
 class Time {
  public:
   explicit Time(int64_t offset)
       : kNtpJan1970(2208988800UL),
         time_now_ms_(offset) {}
 
-  RtcpMeasurement GenerateRtcp(int frequency, uint32_t offset) const {
-    RtcpMeasurement rtcp;
-    rtcp.ntp_time = GetNowNtp();
-    rtcp.rtp_timestamp = GetNowRtp(frequency, offset);
-    return rtcp;
-  }
-
   NtpTime GetNowNtp() const {
     uint32_t ntp_secs = time_now_ms_ / 1000 + kNtpJan1970;
     int64_t remainder_ms = time_now_ms_ % 1000;
     uint32_t ntp_frac = static_cast<uint32_t>(
         static_cast<double>(remainder_ms) * kNtpFracPerMs + 0.5);
     return NtpTime(ntp_secs, ntp_frac);
   }
 
   uint32_t GetNowRtp(int frequency, uint32_t offset) const {
     return frequency * time_now_ms_ / 1000 + offset;
@@ skipping 43 matching lines @@
     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.
-    RtcpMeasurement rtcp =
-        send_time_->GenerateRtcp(audio_frequency, audio_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
-                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
-                               &audio.rtcp, &new_sr));
+    NtpTime ntp_time = send_time_->GetNowNtp();
+    uint32_t rtp_timestamp =
+        send_time_->GetNowRtp(audio_frequency, audio_offset);
+    EXPECT_TRUE(audio.rtp_to_ntp.UpdateMeasurements(
+        ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr));
     send_time_->IncreaseTimeMs(100);
     receive_time_->IncreaseTimeMs(100);
-    rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
-                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
-                               &video.rtcp, &new_sr));
+    ntp_time = send_time_->GetNowNtp();
+    rtp_timestamp = send_time_->GetNowRtp(video_frequency, video_offset);
+    EXPECT_TRUE(video.rtp_to_ntp.UpdateMeasurements(
+        ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr));
     send_time_->IncreaseTimeMs(900);
     receive_time_->IncreaseTimeMs(900);
-    rtcp = send_time_->GenerateRtcp(audio_frequency, audio_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
-                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
-                               &audio.rtcp, &new_sr));
+    ntp_time = send_time_->GetNowNtp();
+    rtp_timestamp = send_time_->GetNowRtp(audio_frequency, audio_offset);
+    EXPECT_TRUE(audio.rtp_to_ntp.UpdateMeasurements(
+        ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr));
     send_time_->IncreaseTimeMs(100);
     receive_time_->IncreaseTimeMs(100);
-    rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
-                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
-                               &video.rtcp, &new_sr));
+    ntp_time = send_time_->GetNowNtp();
+    rtp_timestamp = send_time_->GetNowRtp(video_frequency, video_offset);
+    EXPECT_TRUE(video.rtp_to_ntp.UpdateMeasurements(
+        ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &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_->GetNowRtp(audio_frequency, audio_offset);
     video.latest_timestamp =
         send_time_->GetNowRtp(video_frequency, video_offset);
 
@@ skipping 426 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