Add EncodedImageCallback::OnEncodedImage().

webrtc/voice_engine/channel.cc

 /*
  *  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 342 matching lines @@
   if (_includeAudioLevelIndication) {
     // Store current audio level in the RTP/RTCP module.
     // The level will be used in combination with voice-activity state
     // (frameType) to add an RTP header extension
     _rtpRtcpModule->SetAudioLevel(rms_level_.RMS());
   }
 
   // Push data from ACM to RTP/RTCP-module to deliver audio frame for
   // packetization.
   // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
-  if (_rtpRtcpModule->SendOutgoingData(
+  if (!_rtpRtcpModule->SendOutgoingData(
           (FrameType&)frameType, payloadType, timeStamp,
           // Leaving the time when this frame was
           // received from the capture device as
           // undefined for voice for now.
-          -1, payloadData, payloadSize, fragmentation) == -1) {
+          -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "Channel::SendData() failed to send data to RTP/RTCP module");
     return -1;
   }
 
   _lastLocalTimeStamp = timeStamp;
   _lastPayloadType = payloadType;
 
   return 0;
@@ skipping 3212 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc