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Unified Diff: webrtc/video/video_send_stream.cc

Issue 1984983002: Remove use of RtpHeaderExtension and clean up (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 4 years, 7 months ago
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Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index dcacf56388b37f3ae6a5ee387c9a010d3dcc7c0f..0d574cbe08da8769701687f23d8fe51bcefd1c49 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -420,7 +420,7 @@ VideoSendStream::VideoSendStream(
video_sender_->RegisterProtectionCallback(this);
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
- const std::string& extension = config_.rtp.extensions[i].name;
+ const std::string& extension = config_.rtp.extensions[i].uri;
int id = config_.rtp.extensions[i].id;
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(id, 1);

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