Replace NULL with nullptr in all C++ files.

webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc

 /*
  *  Copyright (c) 2016 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 "webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h"
@@ skipping 33 matching lines @@
 }
 
 // Generates a new frame for this stream. If called too soon after the
 // previous frame, no frame will be generated. The frame is split into
 // packets.
 int64_t RtpStream::GenerateFrame(int64_t time_now_us,
                                  std::vector<PacketInfo>* packets) {
   if (time_now_us < next_rtp_time_) {
     return next_rtp_time_;
   }
-  RTC_CHECK(packets != NULL);
+  RTC_CHECK(packets != nullptr);
   size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
   size_t n_packets =
       std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
   size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
   for (size_t i = 0; i < n_packets; ++i) {
     PacketInfo packet(-1, sequence_number_++);
     packet.send_time_ms = (time_now_us + kSendSideOffsetUs) / 1000;
     packet.payload_size = payload_size;
     packet.probe_cluster_id = PacketInfo::kNotAProbe;
     packets->push_back(packet);
@@ skipping 54 matching lines @@
     total_bitrate_after += stream->bitrate_bps();
   }
   ASSERT_EQ(bitrate_before, total_bitrate_before);
   EXPECT_EQ(total_bitrate_after, bitrate_bps);
 }
 
 // TODO(holmer): Break out the channel simulation part from this class to make
 // it possible to simulate different types of channels.
 int64_t StreamGenerator::GenerateFrame(std::vector<PacketInfo>* packets,
                                        int64_t time_now_us) {
-  RTC_CHECK(packets != NULL);
+  RTC_CHECK(packets != nullptr);
   RTC_CHECK(packets->empty());
   RTC_CHECK_GT(capacity_, 0);
   auto it =
       std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
   (*it)->GenerateFrame(time_now_us, packets);
   int i = 0;
   for (PacketInfo& packet : *packets) {
     int capacity_bpus = capacity_ / 1000;
     int64_t required_network_time_us =
         (8 * 1000 * packet.payload_size + capacity_bpus / 2) / capacity_bpus;
@@ skipping 353 matching lines @@
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, 1000);
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   uint32_t bitrate_after = 0;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc