Revert of Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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 24 matching lines @@
   virtual ~RTPSenderVideo();
 
   virtual RtpVideoCodecTypes VideoCodecType() const;
 
   size_t FECPacketOverhead() const;
 
   static RtpUtility::Payload* CreateVideoPayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       int8_t payload_type);
 
-  bool SendVideo(RtpVideoCodecTypes video_type,
-                 FrameType frame_type,
-                 int8_t payload_type,
-                 uint32_t capture_timestamp,
-                 int64_t capture_time_ms,
-                 const uint8_t* payload_data,
-                 size_t payload_size,
-                 const RTPFragmentationHeader* fragmentation,
-                 const RTPVideoHeader* video_header);
+  int32_t SendVideo(RtpVideoCodecTypes video_type,
+                    FrameType frame_type,
+                    int8_t payload_type,
+                    uint32_t capture_timestamp,
+                    int64_t capture_time_ms,
+                    const uint8_t* payload_data,
+                    size_t payload_size,
+                    const RTPFragmentationHeader* fragmentation,
+                    const RTPVideoHeader* video_header);
 
   int32_t SendRTPIntraRequest();
 
   void SetVideoCodecType(RtpVideoCodecTypes type);
 
   // FEC
   void SetGenericFECStatus(bool enable,
                            uint8_t payload_type_red,
                            uint8_t payload_type_fec);
 
@@ skipping 53 matching lines @@
   // and any padding overhead.
   RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
   // Bitrate used for video payload and RTP headers.
   RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
   OneTimeEvent first_frame_sent_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_