Remove unnecessary RTPFragmentationHeader from VideoProcessor callbacks.

webrtc/modules/video_coding/codecs/test/videoprocessor.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 268 matching lines @@
 int VideoProcessor::NumberDroppedFrames() {
   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
   return num_dropped_frames_;
 }
 
 int VideoProcessor::NumberSpatialResizes() {
   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
   return num_spatial_resizes_;
 }
 
-void VideoProcessor::FrameEncoded(
-    webrtc::VideoCodecType codec,
-    const EncodedImage& encoded_image,
-    const webrtc::RTPFragmentationHeader* fragmentation) {
+void VideoProcessor::FrameEncoded(webrtc::VideoCodecType codec,
+                                  const EncodedImage& encoded_image) {
   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
 
   // For the highest measurement accuracy of the encode time, the start/stop
   // time recordings should wrap the Encode call as tightly as possible.
   int64_t encode_stop_ns = rtc::TimeNanos();
 
   if (encoded_frame_writer_) {
     RTC_CHECK(encoded_frame_writer_->WriteFrame(encoded_image, codec));
   }
 
@@ skipping 85 matching lines @@
   memcpy(&copied_image, &encoded_image, sizeof(copied_image));
   copied_image._size = copied_buffer_size;
   copied_image._buffer = copied_buffer.get();
 
   if (!exclude_this_frame) {
     frame_stat->packets_dropped =
         packet_manipulator_->ManipulatePackets(&copied_image);
   }
   frame_info->manipulated_length = copied_image._length;
 
-  // Keep track of if frames are lost due to packet loss so we can tell
-  // this to the encoder (this is handled by the RTP logic in the full stack).
-  // TODO(kjellander): Pass fragmentation header to the decoder when
-  // CL 172001 has been submitted and PacketManipulator supports this.
-
   // For the highest measurement accuracy of the decode time, the start/stop
   // time recordings should wrap the Decode call as tightly as possible.
   frame_info->decode_start_ns = rtc::TimeNanos();
   frame_stat->decode_return_code =
       decoder_->Decode(copied_image, last_frame_missing, nullptr);
 
   if (frame_stat->decode_return_code != WEBRTC_VIDEO_CODEC_OK) {
     // Write the last successful frame the output file to avoid getting it out
     // of sync with the source file for SSIM and PSNR comparisons.
     RTC_DCHECK_EQ(last_decoded_frame_buffer_.size(),
@@ skipping 93 matching lines @@
 
   RTC_DCHECK_GT(timestamp, 0);
   const int ticks_per_frame =
       kRtpClockRateHz / config_.codec_settings.maxFramerate;
   RTC_DCHECK_EQ(timestamp % ticks_per_frame, 0);
   return (timestamp / ticks_per_frame) - 1;
 }
 
 }  // namespace test
 }  // namespace webrtc