zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock.

Browse Source 
render_time time field (means capture time for sender side) is used by rtcp SenderReport to calculate offset since last frame and to estimate rtp timestamp for the time SenderReport should be send at.
mapping between rtp timestamp and ntp time in SenderReport is used for stream synchronization.

calculation of rtp_timestamp (using ntp_time of incoming video frame) for rtp packets is unchanged.

BUG=webrtc:5433, webrtc:5504, webrtc:5505

Review URL: https://codereview.webrtc.org/1693443002

Cr-Commit-Position: refs/heads/master@{#11820}
This commit is contained in:
danilchap
2016-02-29 12:17:04 -08:00
committed by Commit bot
parent b9338ac62b
commit ac287ee8b5
5 changed files with 22 additions and 23 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/call/call_perf_tests.cc
View File

@ -383,9 +383,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
VoiceEngine::Delete(voice_engine);
}
// TODO(danilchap): Reenable after adding support for frame capture clock
// that is not in sync with local TickTime clock.
TEST_F(CallPerfTest, DISABLED_PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::PercentsFaster(10.0f),
DriftingClock::kNoDrift, DriftingClock::kNoDrift);