Delete VideoReceiveStream::Config::pre_render_callback.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 188 matching lines @@
   ss << ", c_name: " << c_name;
   ss << '}';
   return ss.str();
 }
 
 std::string VideoSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{encoder_settings: " << encoder_settings.ToString();
   ss << ", rtp: " << rtp.ToString();
   ss << ", pre_encode_callback: "
-     << (pre_encode_callback ? "(I420FrameCallback)" : "nullptr");
+     << (pre_encode_callback ? "(VideoSinkInterface)" : "nullptr");
   ss << ", post_encode_callback: "
      << (post_encode_callback ? "(EncodedFrameObserver)" : "nullptr");
   ss << ", render_delay_ms: " << render_delay_ms;
   ss << ", target_delay_ms: " << target_delay_ms;
   ss << ", suspend_below_min_bitrate: " << (suspend_below_min_bitrate ? "on"
                                                                       : "off");
   ss << '}';
   return ss.str();
 }
 
@@ skipping 1115 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc