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 276 matching lines @@
     */
     virtual void BitrateSent(uint32_t* totalRate,
                              uint32_t* videoRate,
                              uint32_t* fecRate,
                              uint32_t* nackRate) const = 0;
 
     /*
     *   Used by the codec module to deliver a video or audio frame for
     *   packetization.
     *
-    *   frameType       - type of frame to send
-    *   payloadType     - payload type of frame to send
+    *   frame_type      - type of frame to send
+    *   payload_type    - payload type of frame to send
     *   timestamp       - timestamp of frame to send
-    *   payloadData     - payload buffer of frame to send
-    *   payloadSize     - size of payload buffer 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
     *
     *   return -1 on failure else 0
     */
     virtual int32_t SendOutgoingData(
-        FrameType frameType,
-        int8_t payloadType,
-        uint32_t timeStamp,
+        FrameType frame_type,
+        int8_t payload_type,
+        uint32_t timestamp,
         int64_t capture_time_ms,
-        const uint8_t* payloadData,
-        size_t payloadSize,
-        const RTPFragmentationHeader* fragmentation = NULL,
-        const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
+        const uint8_t* payload_data,
+        size_t payload_size,
+        const RTPFragmentationHeader* fragmentation,
+        const RTPVideoHeader* rtp_video_header,
+        uint32_t* frame_id_out) = 0;
+
+    // Deprecated version of the methood above.

[stefan-webrtc, 2016/07/28 08:51:17]

method

[Sergey Ulanov, 2016/07/28 20:52:46]

Done.

+    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);
+    }
 
     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 318 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_