| Index: webrtc/modules/rtp_rtcp/include/rtp_receiver.h
|
| diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
|
| index 0a3e201c749b75cc74bc502afd241b1c9e671e59..f393e41eabd6110afc1f685c333f6bb21556d05b 100644
|
| --- a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
|
| +++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
|
| @@ -16,7 +16,6 @@
|
|
|
| namespace webrtc {
|
|
|
| -class RtpAudioFeedback;
|
| class RTPPayloadRegistry;
|
|
|
| class TelephoneEventHandler {
|
| @@ -50,15 +49,6 @@ class RtpReceiver {
|
| RtpFeedback* incoming_messages_callback,
|
| RTPPayloadRegistry* rtp_payload_registry);
|
|
|
| - // DEPRECATED: Creates an audio-enabled RTP receiver.
|
| - // TODO(solenberg): Remove, after updating downstream code.
|
| - static RtpReceiver* CreateAudioReceiver(
|
| - Clock* clock,
|
| - RtpAudioFeedback* incoming_audio_feedback,
|
| - RtpData* incoming_payload_callback,
|
| - RtpFeedback* incoming_messages_callback,
|
| - RTPPayloadRegistry* rtp_payload_registry);
|
| -
|
| virtual ~RtpReceiver() {}
|
|
|
| // Returns a TelephoneEventHandler if available.
|
|
|