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Allow AbsSendTime extension to be used for audio streams.

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Bug: webrtc:10742
Change-Id: I565b58e9f8d70e09976775e0c87fe44c8f026e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146701
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28655}
This commit is contained in:
Sebastian Jansson
2019-07-23 20:55:49 +02:00
committed by Commit Bot
parent e1795f4158
commit 46bbdec1ab
1 changed files with 1 additions and 0 deletions
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api/rtp_parameters.cc
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@ -155,6 +155,7 @@ constexpr int RtpExtension::kOneByteHeaderExtensionMaxValueSize;
bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
return uri == webrtc::RtpExtension::kAudioLevelUri ||
uri == webrtc::RtpExtension::kAbsSendTimeUri ||
// TODO(bugs.webrtc.org/10739): Uncomment once the audio impl is ready.
// uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||