| Index: webrtc/call/rtp_transport_controller_send_interface.h
|
| diff --git a/webrtc/call/rtp_transport_controller_send_interface.h b/webrtc/call/rtp_transport_controller_send_interface.h
|
| index 31f07463a3a283ec364b0e96fa5c0edf0bcf758d..9d26e98dc2dfafef5ccf3549d03804f903ab46f8 100644
|
| --- a/webrtc/call/rtp_transport_controller_send_interface.h
|
| +++ b/webrtc/call/rtp_transport_controller_send_interface.h
|
| @@ -33,7 +33,7 @@ class TransportFeedbackObserver;
|
| //
|
| // This should also have a reference to the underlying
|
| // webrtc::Transport(s). Currently, webrtc::Transport is implemented by
|
| -// WebRtcVideoChannel2 and WebRtcVoiceMediaChannel, and owned by
|
| +// WebRtcVideoChannel and WebRtcVoiceMediaChannel, and owned by
|
| // WebrtcSession. Video and audio always uses different transport
|
| // objects, even in the common case where they are bundled over the
|
| // same underlying transport.
|
|
|