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. |