Allow AbsSendTime extension to be used for audio streams.
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}
diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc
index cb5032d..c3f14d8 100644
--- a/api/rtp_parameters.cc
+++ b/api/rtp_parameters.cc
@@ -155,6 +155,7 @@
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 ||