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authorStefan Holmer <stefan@webrtc.org>2016-01-13 08:58:38 +0100
committerStefan Holmer <stefan@webrtc.org>2016-01-13 07:58:52 +0000
commitea8c0f6fcbd79fd3933cf2f2511a8900c1a9344a (patch)
tree87736822d0fc24dede4fe67789520d0b79676613
parent365543d0e782400a91ac714059273fd16101d256 (diff)
downloadwebrtc-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.cc5
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_);