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 40e73ebd0e172c71e2df29f6af670a5906def5ca..0325c4fb9f549dfff876bfa0650f68fed2b27d9c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -413,7 +413,8 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation,
-    const RTPVideoHeader* rtp_video_hdr) {
+    const RTPVideoHeader* rtp_video_hdr,
+    uint32_t* frame_id_out) {
   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)) {
@@ -421,7 +422,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
   }
   return rtp_sender_.SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_size, fragmentation, rtp_video_hdr);
+      payload_size, fragmentation, rtp_video_hdr, frame_id_out);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,