Cleanup of webrtc::VideoRenderer

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 <stdio.h>
@@ skipping 181 matching lines @@
   bool SendRtcp(const uint8_t* packet, size_t length) override {
     return transport_->SendRtcp(packet, length);
   }
 
   void EncodedFrameCallback(const EncodedFrame& frame) override {
     rtc::CritScope lock(&comparison_lock_);
     if (frames_recorded_ < frames_to_process_)
       encoded_frame_size_.AddSample(frame.length_);
   }
 
-  void RenderFrame(const VideoFrame& video_frame,
-                   int time_to_render_ms) override {
+  void OnFrame(const VideoFrame& video_frame) override {
     int64_t render_time_ms =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
 
     rtc::CritScope lock(&crit_);
     uint32_t send_timestamp =
         wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
 
     while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
       if (last_rendered_frame_.IsZeroSize()) {
         // No previous frame rendered, this one was dropped after sending but
@@ skipping 19 matching lines @@
       // Look at RTPSender::BuildRTPHeader.
       ++send_timestamp;
     }
     ASSERT_EQ(reference_timestamp, send_timestamp);
 
     AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
 
     last_rendered_frame_ = video_frame;
   }
 
-  bool IsTextureSupported() const override { return false; }
-
   void Wait() {
     // Frame comparisons can be very expensive. Wait for test to be done, but
     // at time-out check if frames_processed is going up. If so, give it more
     // time, otherwise fail. Hopefully this will reduce test flakiness.
 
     stats_polling_thread_.Start();
 
     int last_frames_processed = -1;
     int iteration = 0;
     while (!done_.Wait(VideoQualityTest::kDefaultTimeoutMs)) {
@@ skipping 869 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc