webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc - Issue 1412683004: Fix bug in how send timestamps are converted to 24 bits.

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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Issue 1412683004: Fix bug in how send timestamps are converted to 24 bits. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: some stuff Created 5 years, 2 months ago

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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

index e4ace67a4851346b7b99cd8b97ecf668eb8d67f8..9be888c92a24755678e85d41590ac42fe1feb7c3 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -64,7 +64,7 @@ size_t GetPayloadDataLength(const RTPHeader& rtp_header,

}

uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {

- return 0x00fffffful & ((time_ms << 18) / 1000);

+ return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;

}

class LoopbackTransportTest : public webrtc::Transport {

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