Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/neteq/statistics_calculator.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/audio_coding/neteq/statistics_calculator.h"
 
 #include <assert.h>
 #include <string.h>  // memset
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
 
 namespace webrtc {
 
 StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
     const std::string& uma_name,
     int report_interval_ms,
     int max_value)
@@ skipping 106 matching lines @@
   lost_timestamps_ = 0;
   timestamps_since_last_report_ = 0;
 }
 
 void StatisticsCalculator::ResetWaitingTimeStatistics() {
   memset(waiting_times_, 0, kLenWaitingTimes * sizeof(waiting_times_[0]));
   len_waiting_times_ = 0;
   next_waiting_time_index_ = 0;
 }
 
-void StatisticsCalculator::ExpandedVoiceSamples(int num_samples) {
+void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples) {
   expanded_speech_samples_ += num_samples;
 }
 
-void StatisticsCalculator::ExpandedNoiseSamples(int num_samples) {
+void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples) {
   expanded_noise_samples_ += num_samples;
 }
 
-void StatisticsCalculator::PreemptiveExpandedSamples(int num_samples) {
+void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
   preemptive_samples_ += num_samples;
 }
 
-void StatisticsCalculator::AcceleratedSamples(int num_samples) {
+void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
   accelerate_samples_ += num_samples;
 }
 
-void StatisticsCalculator::AddZeros(int num_samples) {
+void StatisticsCalculator::AddZeros(size_t num_samples) {
   added_zero_samples_ += num_samples;
 }
 
-void StatisticsCalculator::PacketsDiscarded(int num_packets) {
+void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
   discarded_packets_ += num_packets;
 }
 
-void StatisticsCalculator::LostSamples(int num_samples) {
+void StatisticsCalculator::LostSamples(size_t num_samples) {
   lost_timestamps_ += num_samples;
 }
 
-void StatisticsCalculator::IncreaseCounter(int num_samples, int fs_hz) {
-  const int time_step_ms = rtc::CheckedDivExact(1000 * num_samples, fs_hz);
+void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
+  const int time_step_ms =
+      rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
   delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
   excess_buffer_delay_.AdvanceClock(time_step_ms);
   timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
   if (timestamps_since_last_report_ >
       static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
     lost_timestamps_ = 0;
     timestamps_since_last_report_ = 0;
     discarded_packets_ = 0;
   }
 }
@@ skipping 17 matching lines @@
   if (next_waiting_time_index_ >= kLenWaitingTimes) {
     next_waiting_time_index_ = 0;
   }
   if (len_waiting_times_ < kLenWaitingTimes) {
     len_waiting_times_++;
   }
 }
 
 void StatisticsCalculator::GetNetworkStatistics(
     int fs_hz,
-    int num_samples_in_buffers,
-    int samples_per_packet,
+    size_t num_samples_in_buffers,
+    size_t samples_per_packet,
     const DelayManager& delay_manager,
     const DecisionLogic& decision_logic,
     NetEqNetworkStatistics *stats) {
   if (fs_hz <= 0 || !stats) {
     assert(false);
     return;
   }
 
   stats->added_zero_samples = added_zero_samples_;
   stats->current_buffer_size_ms =
       static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
-  const int ms_per_packet = decision_logic.packet_length_samples() /
-      (fs_hz / 1000);
+  const int ms_per_packet = rtc::checked_cast<int>(
+      decision_logic.packet_length_samples() / (fs_hz / 1000));
   stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) *
       ms_per_packet;
   stats->jitter_peaks_found = delay_manager.PeakFound();
   stats->clockdrift_ppm = delay_manager.AverageIAT();
 
   stats->packet_loss_rate =
       CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
 
-  const unsigned discarded_samples = discarded_packets_ * samples_per_packet;
+  const size_t discarded_samples = discarded_packets_ * samples_per_packet;
   stats->packet_discard_rate =
       CalculateQ14Ratio(discarded_samples, timestamps_since_last_report_);
 
   stats->accelerate_rate =
       CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);
 
   stats->preemptive_rate =
       CalculateQ14Ratio(preemptive_samples_, timestamps_since_last_report_);
 
   stats->expand_rate =
@@ skipping 14 matching lines @@
 }
 
 void StatisticsCalculator::WaitingTimes(std::vector<int>* waiting_times) {
   if (!waiting_times) {
     return;
   }
   waiting_times->assign(waiting_times_, waiting_times_ + len_waiting_times_);
   ResetWaitingTimeStatistics();
 }
 
-uint16_t StatisticsCalculator::CalculateQ14Ratio(uint32_t numerator,
+uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
                                                  uint32_t denominator) {
   if (numerator == 0) {
     return 0;
   } else if (numerator < denominator) {
     // Ratio must be smaller than 1 in Q14.
     assert((numerator << 14) / denominator < (1 << 14));
     return static_cast<uint16_t>((numerator << 14) / denominator);
   } else {
     // Will not produce a ratio larger than 1, since this is probably an error.
     return 1 << 14;
   }
 }
 
 }  // namespace webrtc