Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 207 matching lines @@
 
   // Used by the codec module to deliver a video or audio frame for
   // packetization.
   // |frame_type|    - type of frame to send
   // |payload_type|  - payload type of frame to send
   // |timestamp|     - timestamp of frame to send
   // |payload_data|  - payload buffer of frame to send
   // |payload_size|  - size of payload buffer to send
   // |fragmentation| - fragmentation offset data for fragmented frames such
   //                   as layers or RED
-  // Returns -1 on failure else 0.
-  virtual int32_t SendOutgoingData(
+  // |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,
+                                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) = 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) = 0;
+      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,
                                 bool retransmission,
                                 int probe_cluster_id) = 0;
 
   virtual size_t TimeToSendPadding(size_t bytes, int probe_cluster_id) = 0;
 
   // Called on generation of new statistics after an RTP send.
@@ skipping 211 matching lines @@
   virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
 
   // Sends a request for a keyframe.
   // Returns -1 on failure else 0.
   virtual int32_t RequestKeyFrame() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_