Revert of Add content type information to encoded images and corresponding rtp extension header

webrtc/video/video_stream_decoder.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 58 matching lines @@
   video_receiver_->RegisterReceiveStatisticsCallback(nullptr);
   video_receiver_->RegisterFrameTypeCallback(nullptr);
   video_receiver_->RegisterReceiveCallback(nullptr);
 }
 
 // Do not acquire the lock of |video_receiver_| in this function. Decode
 // callback won't necessarily be called from the decoding thread. The decoding
 // thread may have held the lock when calling VideoDecoder::Decode, Reset, or
 // Release. Acquiring the same lock in the path of decode callback can deadlock.
 int32_t VideoStreamDecoder::FrameToRender(VideoFrame& video_frame,
-                                          rtc::Optional<uint8_t> qp,
-                                          VideoContentType content_type) {
-  receive_stats_callback_->OnDecodedFrame(qp, content_type);
+                                          rtc::Optional<uint8_t> qp) {
+  receive_stats_callback_->OnDecodedFrame(qp);
   incoming_video_stream_->OnFrame(video_frame);
+
   return 0;
 }
 
 int32_t VideoStreamDecoder::ReceivedDecodedReferenceFrame(
   const uint64_t picture_id) {
   RTC_NOTREACHED();
   return 0;
 }
 
 void VideoStreamDecoder::OnIncomingPayloadType(int payload_type) {
@@ skipping 28 matching lines @@
 
 void VideoStreamDecoder::OnCompleteFrame(bool is_keyframe, size_t size_bytes) {}
 
 void VideoStreamDecoder::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
   video_receiver_->SetReceiveChannelParameters(max_rtt_ms);
 
   rtc::CritScope lock(&crit_);
   last_rtt_ms_ = avg_rtt_ms;
 }
 }  // namespace webrtc