| Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
| index fd44a597261593be17cf99f32a8f28a41be569e4..e8fc545ec52e415784a7c3fedb16ecca5679de92 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
| @@ -112,15 +112,15 @@
|
|
|
| // Used by the codec module to deliver a video or audio frame for
|
| // packetization.
|
| - bool SendOutgoingData(FrameType frame_type,
|
| - int8_t payload_type,
|
| - uint32_t time_stamp,
|
| - int64_t capture_time_ms,
|
| - const uint8_t* payload_data,
|
| - size_t payload_size,
|
| - const RTPFragmentationHeader* fragmentation,
|
| - const RTPVideoHeader* rtp_video_header,
|
| - uint32_t* transport_frame_id_out) override;
|
| + int32_t SendOutgoingData(
|
| + FrameType frame_type,
|
| + int8_t payload_type,
|
| + uint32_t time_stamp,
|
| + int64_t capture_time_ms,
|
| + const uint8_t* payload_data,
|
| + size_t payload_size,
|
| + const RTPFragmentationHeader* fragmentation = NULL,
|
| + const RTPVideoHeader* rtp_video_header = NULL) override;
|
|
|
| bool TimeToSendPacket(uint32_t ssrc,
|
| uint16_t sequence_number,
|
|
|