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Unified Diff: webrtc/voice_engine/channel.cc

Issue 2007743003: Add sender controlled playout delay limits (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@cleanup_rtp_hdr_extensions
Patch Set: Addressed comments Created 4 years, 7 months ago
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Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index c2b95420537f4763893d1d5cea4d188240e9d527..ed41f472ffd22b5c77bdd3e8ae4c496dbdc4377b 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2536,6 +2536,18 @@ int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
return 0;
}
+void Channel::EnableSendPlayoutDelayLimit(int id) {
+ int ret = _rtpRtcpModule->RegisterSendRtpHeaderExtension(
+ kRtpExtensionPlayoutDelay, id);
+ RTC_DCHECK_EQ(0, ret);
+}
+
+void Channel::EnableReceivePlayoutDelayLimit(int id) {
+ bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionPlayoutDelay, id);
+ RTC_DCHECK(ret);
+}
+
void Channel::EnableSendTransportSequenceNumber(int id) {
int ret =
SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);

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