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Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format

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making it symmetric to AbsoluteSendTime::Parse function.

Bug: None
Change-Id: I9c71d840768064022ebebbbeb2962aeeecc68392
Reviewed-on: https://chromium-review.googlesource.com/531044
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18555}
This commit is contained in:
Danil Chapovalov
2017-06-12 15:43:55 +02:00
committed by Commit Bot
parent 29f0d453aa
commit f3ba6484e3
4 changed files with 11 additions and 8 deletions
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2
webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
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@ -126,7 +126,7 @@ RtpPacketToSend GenerateRtpPacket(const RtpHeaderExtensionMap* extensions,
rtp_packet.SetExtension<TransmissionOffset>(prng->Rand(0x00ffffff));
rtp_packet.SetExtension<AudioLevel>(prng->Rand<bool>(), prng->Rand(127));
rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand<int32_t>());
rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand(0x00ffffff));
rtp_packet.SetExtension<VideoOrientation>(prng->Rand(2));
rtp_packet.SetExtension<TransportSequenceNumber>(prng->Rand<uint16_t>());