Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/decision_logic_normal.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/decision_logic_normal.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/interface/module_common_types.h"
 
 namespace webrtc {
 
 Operations DecisionLogicNormal::GetDecisionSpecialized(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
     bool* reset_decoder) {
   assert(playout_mode_ == kPlayoutOn || playout_mode_ == kPlayoutStreaming);
   // Guard for errors, to avoid getting stuck in error mode.
   if (prev_mode == kModeError) {
     if (!packet_header) {
       return kExpand;
     } else {
@@ skipping 101 matching lines @@
       if (buffer_level_filter_->filtered_current_level() < low_limit)
         return kPreemptiveExpand;
     }
   }
   return kNormal;
 }
 
 Operations DecisionLogicNormal::FuturePacketAvailable(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     Modes prev_mode,
     uint32_t target_timestamp,
     uint32_t available_timestamp,
     bool play_dtmf) {
   // Required packet is not available, but a future packet is.
   // Check if we should continue with an ongoing expand because the new packet
   // is too far into the future.
   uint32_t timestamp_leap = available_timestamp - target_timestamp;
   if ((prev_mode == kModeExpand) &&
       !ReinitAfterExpands(timestamp_leap) &&
       !MaxWaitForPacket() &&
       PacketTooEarly(timestamp_leap) &&
       UnderTargetLevel()) {
     if (play_dtmf) {
       // Still have DTMF to play, so do not do expand.
       return kDtmf;
     } else {
       // Nothing to play.
       return kExpand;
     }
   }
 
-  const int samples_left = static_cast<int>(sync_buffer.FutureLength() -
-      expand.overlap_length());
-  const int cur_size_samples = samples_left +
+  const size_t samples_left =
+      sync_buffer.FutureLength() - expand.overlap_length();
+  const size_t cur_size_samples = samples_left +
       packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
 
   // If previous was comfort noise, then no merge is needed.
   if (prev_mode == kModeRfc3389Cng ||
       prev_mode == kModeCodecInternalCng) {
     // Keep the same delay as before the CNG (or maximum 70 ms in buffer as
     // safety precaution), but make sure that the number of samples in buffer
     // is no higher than 4 times the optimal level. (Note that TargetLevel()
     // is in Q8.)
     if (static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) >=
@@ skipping 40 matching lines @@
 bool DecisionLogicNormal::PacketTooEarly(uint32_t timestamp_leap) const {
   return timestamp_leap >
       static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
 }
 
 bool DecisionLogicNormal::MaxWaitForPacket() const {
   return num_consecutive_expands_ >= kMaxWaitForPacket;
 }
 
 }  // namespace webrtc