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

webrtc/modules/audio_coding/neteq/decision_logic_normal.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Implementation of the DecisionLogic class for playout modes kPlayoutOn and
 // kPlayoutStreaming.
 class DecisionLogicNormal : public DecisionLogic {
  public:
   // Constructor.
   DecisionLogicNormal(int fs_hz,
-                      int output_size_samples,
+                      size_t output_size_samples,
                       NetEqPlayoutMode playout_mode,
                       DecoderDatabase* decoder_database,
                       const PacketBuffer& packet_buffer,
                       DelayManager* delay_manager,
                       BufferLevelFilter* buffer_level_filter)
       : DecisionLogic(fs_hz, output_size_samples, playout_mode,
                       decoder_database, packet_buffer, delay_manager,
                       buffer_level_filter) {
   }
 
  protected:
   static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
   static const int kReinitAfterExpands = 100;
   static const int kMaxWaitForPacket = 10;
 
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
   // obtained from the last decoded frame. If there is a packet available, the
   // packet header should be supplied in |packet_header|; otherwise it should
   // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
   // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
   // should be set to true. The output variable |reset_decoder| will be set to
   // true if a reset is required; otherwise it is left unchanged (i.e., it can
   // remain true if it was true before the call).
   Operations 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) override;
 
   // Returns the operation to do given that the expected packet is not
   // available, but a packet further into the future is at hand.
   virtual Operations 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);
 
   // Returns the operation to do given that the expected packet is available.
   virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
 
   // Returns the operation given that no packets are available (except maybe
   // a DTMF event, flagged by setting |play_dtmf| true).
@@ skipping 22 matching lines @@
   bool PacketTooEarly(uint32_t timestamp_leap) const;
 
   // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
   bool MaxWaitForPacket() const;
 
   DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_