Revert of Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index e58ac3c61bb6f41c998c743ef3c9a0d5d8d59cec..190136d014b717b7564028aa8182fe625625b32a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -384,7 +384,7 @@
   return rtp_sender_.SendingMedia();
 }
 
-bool ModuleRtpRtcpImpl::SendOutgoingData(
+int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     FrameType frame_type,
     int8_t payload_type,
     uint32_t time_stamp,
@@ -392,8 +392,7 @@
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation,
-    const RTPVideoHeader* rtp_video_header,
-    uint32_t* transport_frame_id_out) {
+    const RTPVideoHeader* rtp_video_header) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
   // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
@@ -401,7 +400,7 @@
   }
   return rtp_sender_.SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
+      payload_size, fragmentation, rtp_video_header);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,