Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 25016e053f2b43a948f803846fda364423d31dbe..e1f6ceb172b1cf76051214b0902acc08c15c9740 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -455,7 +455,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
                                     const uint8_t* payload_data,
                                     size_t payload_size,
                                     const RTPFragmentationHeader* fragmentation,
-                                    const RTPVideoHeader* rtp_hdr) {
+                                    const RTPVideoHeader* rtp_hdr,
+                                    uint32_t* frame_id_out) {
   uint32_t ssrc;
   uint16_t sequence_number;
   {
@@ -512,6 +513,13 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
         capture_time_ms, payload_data, payload_size, fragmentation, rtp_hdr);
   }
 
+  if (frame_id_out) {
+    rtc::CritScope lock(&send_critsect_);
+    // TODO(sergeyu): Move RTP timestamp calculation from BuildRTPheader() to
+    // SendOutgoingData() and pass it to SendVideo()/SendAudio() calls.
+    *frame_id_out = timestamp_;
+  }
+
   rtc::CritScope cs(&statistics_crit_);
   // Note: This is currently only counting for video.
   if (frame_type == kVideoFrameKey) {