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author | Stefan Holmer <stefan@webrtc.org> | 2016-01-13 08:58:38 +0100 |
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committer | Stefan Holmer <stefan@webrtc.org> | 2016-01-13 07:58:52 +0000 |
commit | ea8c0f6fcbd79fd3933cf2f2511a8900c1a9344a (patch) | |
tree | 87736822d0fc24dede4fe67789520d0b79676613 | |
parent | 365543d0e782400a91ac714059273fd16101d256 (diff) | |
download | webrtc-ea8c0f6fcbd79fd3933cf2f2511a8900c1a9344a.tar.gz |
Fix capture ntp time issue introduced with r11187.
I think the problem was that I only introduced delay in one direction, and the estimation assumes that the RTT is evenly divided between the send direction and the receive direction, which was true for the old test.
BUG=chromium:576246
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1577853005 .
Cr-Commit-Position: refs/heads/master@{#11233}
-rw-r--r-- | webrtc/call/call_perf_tests.cc | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index 86cd9ae072..3adcb10b09 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -405,6 +405,11 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, sender_call, this, test::PacketTransport::kSender, net_config_); } + test::PacketTransport* CreateReceiveTransport() override { + return new test::PacketTransport( + nullptr, this, test::PacketTransport::kReceiver, net_config_); + } + void RenderFrame(const VideoFrame& video_frame, int time_to_render_ms) override { rtc::CritScope lock(&crit_); |