From 46bbdec1ab1fdc740c918e6389c82aa8fee6905a Mon Sep 17 00:00:00 2001 From: Sebastian Jansson Date: Tue, 23 Jul 2019 20:55:49 +0200 Subject: [PATCH] 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 Commit-Queue: Sebastian Jansson Cr-Commit-Position: refs/heads/master@{#28655} --- api/rtp_parameters.cc | 1 + 1 file changed, 1 insertion(+) diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc index cb5032d9c1..c3f14d8f32 100644 --- a/api/rtp_parameters.cc +++ b/api/rtp_parameters.cc @@ -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 ||