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

webrtc/modules/audio_coding/neteq/delay_manager.cc

 /*
  *  Copyright (c) 2012 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/delay_manager.h"
 
 #include <assert.h>
 #include <math.h>
 
 #include <algorithm>  // max, min
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
-DelayManager::DelayManager(int max_packets_in_buffer,
+DelayManager::DelayManager(size_t max_packets_in_buffer,
                            DelayPeakDetector* peak_detector)
     : first_packet_received_(false),
       max_packets_in_buffer_(max_packets_in_buffer),
       iat_vector_(kMaxIat + 1, 0),
       iat_factor_(0),
       packet_iat_count_ms_(0),
       base_target_level_(4),  // In Q0 domain.
       target_level_(base_target_level_ << 8),  // In Q8 domain.
       packet_len_ms_(0),
       streaming_mode_(false),
@@ skipping 196 matching lines @@
     int minimum_delay_packet_q8 =  (minimum_delay_ms_ << 8) / packet_len_ms_;
     target_level_ = std::max(target_level_, minimum_delay_packet_q8);
   }
 
   if (maximum_delay_ms_ > 0 && packet_len_ms_ > 0) {
     int maximum_delay_packet_q8 = (maximum_delay_ms_ << 8) / packet_len_ms_;
     target_level_ = std::min(target_level_, maximum_delay_packet_q8);
   }
 
   // Shift to Q8, then 75%.;
-  int max_buffer_packets_q8 = (3 * (max_packets_in_buffer_ << 8)) / 4;
+  int max_buffer_packets_q8 =
+      static_cast<int>((3 * (max_packets_in_buffer_ << 8)) / 4);
   target_level_ = std::min(target_level_, max_buffer_packets_q8);
 
   // Sanity check, at least 1 packet (in Q8).
   target_level_ = std::max(target_level_, 1 << 8);
 }
 
 int DelayManager::CalculateTargetLevel(int iat_packets) {
   int limit_probability = kLimitProbability;
   if (streaming_mode_) {
     limit_probability = kLimitProbabilityStreaming;
@@ skipping 129 matching lines @@
     last_pack_cng_or_dtmf_ = -1;
   }
 }
 
 bool DelayManager::SetMinimumDelay(int delay_ms) {
   // Minimum delay shouldn't be more than maximum delay, if any maximum is set.
   // Also, if possible check |delay| to less than 75% of
   // |max_packets_in_buffer_|.
   if ((maximum_delay_ms_ > 0 && delay_ms > maximum_delay_ms_) ||
       (packet_len_ms_ > 0 &&
-          delay_ms > 3 * max_packets_in_buffer_ * packet_len_ms_ / 4)) {
+       delay_ms >
+           static_cast<int>(3 * max_packets_in_buffer_ * packet_len_ms_ / 4))) {
     return false;
   }
   minimum_delay_ms_ = delay_ms;
   return true;
 }
 
 bool DelayManager::SetMaximumDelay(int delay_ms) {
   if (delay_ms == 0) {
     // Zero input unsets the maximum delay.
     maximum_delay_ms_ = 0;
@@ skipping 13 matching lines @@
 int DelayManager::base_target_level() const { return base_target_level_; }
 void DelayManager::set_streaming_mode(bool value) { streaming_mode_ = value; }
 int DelayManager::last_pack_cng_or_dtmf() const {
   return last_pack_cng_or_dtmf_;
 }
 
 void DelayManager::set_last_pack_cng_or_dtmf(int value) {
   last_pack_cng_or_dtmf_ = value;
 }
 }  // namespace webrtc