commit fe50b4d75081c3c0d52c74e97a1b8df57a571b7c
author asapersson <asapersson@webrtc.org> Thu Dec 22 07:53:51 2016 -0800
committer Commit bot <commit-bot@chromium.org> Thu Dec 22 15:53:51 2016 +0000
tree 8afc37f1bdf51d6ad20cce717fb97994d5e43ad7
parent bf5f5297c5b49251191bc2c9d76e4468d25beda4

Make class of static functions in rtp_to_ntp.h:
- UpdateRtcpList
- RtpToNtp

class RtpToNtpEstimator
- UpdateMeasurements
- Estimate

List with rtcp measurements is now private.

BUG=none

Review-Url: https://codereview.webrtc.org/2574133003
Cr-Commit-Position: refs/heads/master@{#15762}

webrtc/video/stream_synchronization_unittest.cc

diff --git a/webrtc/video/stream_synchronization_unittest.cc b/webrtc/video/stream_synchronization_unittest.cc
index 770bbf6..d5607a8 100644
--- a/webrtc/video/stream_synchronization_unittest.cc
+++ b/webrtc/video/stream_synchronization_unittest.cc
@@ -34,13 +34,6 @@
       : 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;
@@ -104,29 +97,29 @@
     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);