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Unified Diff: webrtc/audio/audio_send_stream.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: Created 4 years, 7 months ago
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Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index e9659c67e8c48ecd3bf09369d27562322b1e5be3..92be961a0ae2e2070e34c5ca480daa249a57fc08 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -85,6 +85,8 @@ AudioSendStream::AudioSendStream(
channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
+ } else if (extension.uri == RtpExtension::kPlayoutDelayUri) {
+ channel_proxy_->EnableSendPlayoutDelayLimit(extension.id);
} else {
RTC_NOTREACHED() << "Registering unsupported RTP extension.";
}

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