Replace NULL with nullptr in webrtc/video.

webrtc/video/replay.cc

 /*
  *  Copyright (c) 2014 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 130 matching lines @@
   FileRenderPassthrough(const std::string& basename,
                         rtc::VideoSinkInterface<VideoFrame>* renderer)
       : basename_(basename),
         renderer_(renderer),
         file_(nullptr),
         count_(0),
         last_width_(0),
         last_height_(0) {}
 
   ~FileRenderPassthrough() {
-    if (file_ != nullptr)
+    if (file_)
       fclose(file_);
   }
 
  private:
   void OnFrame(const VideoFrame& video_frame) override {
-    if (renderer_ != nullptr)
+    if (renderer_)
       renderer_->OnFrame(video_frame);
     if (basename_.empty())
       return;
     if (last_width_ != video_frame.width() ||
         last_height_ != video_frame.height()) {
-      if (file_ != nullptr)
+      if (file_)
         fclose(file_);
       std::stringstream filename;
       filename << basename_;
       if (++count_ > 1)
         filename << '-' << count_;
       filename << '_' << video_frame.width() << 'x' << video_frame.height()
                << ".yuv";
       file_ = fopen(filename.str().c_str(), "wb");
-      if (file_ == nullptr) {
+      if (!file_) {
         fprintf(stderr,
                 "Couldn't open file for writing: %s\n",
                 filename.str().c_str());
       }
     }
     last_width_ = video_frame.width();
     last_height_ = video_frame.height();
-    if (file_ == nullptr)
+    if (!file_)
       return;
     PrintVideoFrame(video_frame, file_);
   }
 
   const std::string basename_;
   rtc::VideoSinkInterface<VideoFrame>* const renderer_;
   FILE* file_;
   size_t count_;
   int last_width_;
   int last_height_;
 };
 
 class DecoderBitstreamFileWriter : public EncodedFrameObserver {
  public:
   explicit DecoderBitstreamFileWriter(const char* filename)
       : file_(fopen(filename, "wb")) {
-    RTC_DCHECK(file_ != nullptr);
+    RTC_DCHECK(file_);
   }
   ~DecoderBitstreamFileWriter() { fclose(file_); }
 
   virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
     fwrite(encoded_frame.data_, 1, encoded_frame.length_, file_);
   }
 
  private:
   FILE* file_;
 };
@@ skipping 40 matching lines @@
     delete decoder.decoder;
     decoder.decoder = new test::FakeNullDecoder();
   }
   receive_config.decoders.push_back(decoder);
 
   VideoReceiveStream* receive_stream =
       call->CreateVideoReceiveStream(receive_config);
 
   std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
       test::RtpFileReader::kRtpDump, flags::InputFile()));
-  if (rtp_reader.get() == nullptr) {
+  if (!rtp_reader) {
     rtp_reader.reset(test::RtpFileReader::Create(test::RtpFileReader::kPcap,
                                                  flags::InputFile()));
-    if (rtp_reader.get() == nullptr) {
+    if (!rtp_reader) {
       fprintf(stderr,
               "Couldn't open input file as either a rtpdump or .pcap. Note "
               "that .pcapng is not supported.\nTrying to interpret the file as "
               "length/packet interleaved.\n");
       rtp_reader.reset(test::RtpFileReader::Create(
           test::RtpFileReader::kLengthPacketInterleaved, flags::InputFile()));
-      if (rtp_reader.get() == nullptr) {
+      if (!rtp_reader) {
         fprintf(stderr,
                 "Unable to open input file with any supported format\n");
         return;
       }
     }
   }
   receive_stream->Start();
 
   uint32_t last_time_ms = 0;
   int num_packets = 0;
@@ skipping 40 matching lines @@
 }
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   ::testing::InitGoogleTest(&argc, argv);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   webrtc::test::RunTest(webrtc::RtpReplay);
   return 0;
 }