Delete method webrtc::VideoFrame::Reset.

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>
 
 #include <algorithm>
 #include <deque>
 #include <map>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ skipping 105 matching lines @@
     rtc::CritScope crit(&comparison_lock_);
     samples_encode_time_ms_[ntp_time_ms] = encode_time_ms;
   }
 
   void IncomingCapturedFrame(const VideoFrame& video_frame) override {
     VideoFrame copy = video_frame;
     copy.set_timestamp(copy.ntp_time_ms() * 90);
 
     {
       rtc::CritScope lock(&crit_);
-      if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
-        first_send_frame_ = copy;
+      if (!first_send_timestamp_ && rtp_timestamp_delta_ == 0)
+        first_send_timestamp_ = rtc::Optional<uint32_t>(copy.timestamp());
 
       frames_.push_back(copy);
     }
 
     input_->IncomingCapturedFrame(video_frame);
   }
 
   bool SendRtp(const uint8_t* packet,
                size_t length,
                const PacketOptions& options) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
 
     int64_t current_time =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     bool result = transport_->SendRtp(packet, length, options);
     {
       rtc::CritScope lock(&crit_);
 
       if (rtp_timestamp_delta_ == 0) {
-        rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
-        first_send_frame_.Reset();
+        rtp_timestamp_delta_ = header.timestamp - *first_send_timestamp_;
+        first_send_timestamp_ = rtc::Optional<uint32_t>();
       }
       int64_t timestamp =
           wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
       send_times_[timestamp] = current_time;
       if (!transport_->DiscardedLastPacket() &&
           header.ssrc == ssrc_to_analyze_) {
         encoded_frame_sizes_[timestamp] +=
             length - (header.headerLength + header.paddingLength);
       }
     }
@@ skipping 431 matching lines @@
   int64_t last_render_time_;
   uint32_t rtp_timestamp_delta_;
 
   rtc::CriticalSection crit_;
   std::deque<VideoFrame> frames_ GUARDED_BY(crit_);
   VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
   rtc::TimestampWrapAroundHandler wrap_handler_ GUARDED_BY(crit_);
   std::map<int64_t, int64_t> send_times_ GUARDED_BY(crit_);
   std::map<int64_t, int64_t> recv_times_ GUARDED_BY(crit_);
   std::map<int64_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
-  VideoFrame first_send_frame_ GUARDED_BY(crit_);
+  rtc::Optional<uint32_t> first_send_timestamp_ GUARDED_BY(crit_);
   const double avg_psnr_threshold_;
   const double avg_ssim_threshold_;
 
   rtc::CriticalSection comparison_lock_;
   std::vector<rtc::PlatformThread*> comparison_thread_pool_;
   rtc::PlatformThread stats_polling_thread_;
   rtc::Event comparison_available_event_;
   std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
   rtc::Event done_;
 };
@@ skipping 462 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc