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 0640d5cc19a1dc9eb425abc0a886eb1ffe801d23..0a3e201c749b75cc74bc502afd241b1c9e671e59 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h |
@@ -16,6 +16,7 @@ |
namespace webrtc { |
+class RtpAudioFeedback; |
class RTPPayloadRegistry; |
class TelephoneEventHandler { |
@@ -45,6 +46,14 @@ class RtpReceiver { |
// Creates an audio-enabled RTP receiver. |
static RtpReceiver* CreateAudioReceiver( |
Clock* clock, |
+ RtpData* incoming_payload_callback, |
+ 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, |