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. |