| Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h | 
| diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h | 
| index 9fa3959c906128a81596a7fc8f97e9a269167bbc..0c36117cea64d482a4044c89ea2237e521f4b702 100644 | 
| --- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h | 
| +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h | 
| @@ -227,7 +227,6 @@ class RtpRtcp : public Module { | 
| //                   as layers or RED | 
| // |transport_frame_id_out| - set to RTP timestamp. | 
| // Returns true on success. | 
| - | 
| virtual bool SendOutgoingData(FrameType frame_type, | 
| int8_t payload_type, | 
| uint32_t timestamp, | 
| @@ -238,24 +237,6 @@ class RtpRtcp : public Module { | 
| const RTPVideoHeader* rtp_video_header, | 
| uint32_t* transport_frame_id_out) = 0; | 
|  | 
| -  // Deprecated version of the method above. | 
| -  int32_t SendOutgoingData( | 
| -      FrameType frame_type, | 
| -      int8_t payload_type, | 
| -      uint32_t timestamp, | 
| -      int64_t capture_time_ms, | 
| -      const uint8_t* payload_data, | 
| -      size_t payload_size, | 
| -      const RTPFragmentationHeader* fragmentation = nullptr, | 
| -      const RTPVideoHeader* rtp_video_header = nullptr) { | 
| -    return SendOutgoingData(frame_type, payload_type, timestamp, | 
| -                            capture_time_ms, payload_data, payload_size, | 
| -                            fragmentation, rtp_video_header, | 
| -                            /*frame_id_out=*/nullptr) | 
| -               ? 0 | 
| -               : -1; | 
| -  } | 
| - | 
| virtual bool TimeToSendPacket(uint32_t ssrc, | 
| uint16_t sequence_number, | 
| int64_t capture_time_ms, | 
|  |