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Unified Diff: webrtc/audio_send_stream.h

Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comment Created 5 years ago
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Index: webrtc/audio_send_stream.h
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index 7069c377d36e7257b236c7e3e58fea138da85970..465afd2b49622f32ddd1954cf9ff4a1c90b62aea 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -64,7 +64,7 @@ class AudioSendStream : public SendStream {
// Sender SSRC.
uint32_t ssrc = 0;
- // RTP header extensions used for the received stream.
+ // RTP header extensions used for the sent stream.
std::vector<RtpExtension> extensions;
// RTCP CNAME, see RFC 3550.

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