Index: webrtc/audio_send_stream.h |
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h |
index dd8d9e96ea69a4ad392336e7d2b15cd1da762dac..d1af9e01034099efb8f95be568f83f8d2762eff5 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. |