| 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,
|
|
|