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. |