Remove webrtc/base/scoped_ptr.h

webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc

 /*
  *  Copyright (c) 2013 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/rtp_rtcp/source/receive_statistics_impl.h"
 
 #include <math.h>
 
+#include <cstdlib>
+
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 
 namespace webrtc {
 
 const int64_t kStatisticsTimeoutMs = 8000;
 const int64_t kStatisticsProcessIntervalMs = 1000;
 
 StreamStatistician::~StreamStatistician() {}
 
@@ skipping 81 matching lines @@
 
 void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
                                           NtpTime receive_time) {
   uint32_t receive_time_rtp =
       NtpToRtp(receive_time, header.payload_type_frequency);
   uint32_t last_receive_time_rtp =
       NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
       (header.timestamp - last_received_timestamp_);
 
-  time_diff_samples = abs(time_diff_samples);
+  time_diff_samples = std::abs(time_diff_samples);
 
   // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
   // If this happens, don't update jitter value. Use 5 secs video frequency
   // as the threshold.
   if (time_diff_samples < 450000) {
     // Note we calculate in Q4 to avoid using float.
     int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
     jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
   }
 
   // Extended jitter report, RFC 5450.
   // Actual network jitter, excluding the source-introduced jitter.
   int32_t time_diff_samples_ext =
     (receive_time_rtp - last_receive_time_rtp) -
     ((header.timestamp +
       header.extension.transmissionTimeOffset) -
      (last_received_timestamp_ +
       last_received_transmission_time_offset_));
 
-  time_diff_samples_ext = abs(time_diff_samples_ext);
+  time_diff_samples_ext = std::abs(time_diff_samples_ext);
 
   if (time_diff_samples_ext < 450000) {
     int32_t jitter_diffQ4TransmissionTimeOffset =
       (time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_;
     jitter_q4_transmission_time_offset_ +=
       ((jitter_diffQ4TransmissionTimeOffset + 8) >> 4);
   }
 }
 
 void StreamStatisticianImpl::NotifyRtpCallback() {
@@ skipping 380 matching lines @@
 
 void NullReceiveStatistics::Process() {}
 
 void NullReceiveStatistics::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {}
 
 void NullReceiveStatistics::RegisterRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {}
 
 }  // namespace webrtc