| 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.
|
|
|