Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/decision_logic.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.
  */
 
@@ skipping 16 matching lines @@
 class SyncBuffer;
 struct RTPHeader;
 
 // This is the base class for the decision tree implementations. Derived classes
 // must implement the method GetDecisionSpecialized().
 class DecisionLogic {
  public:
   // Static factory function which creates different types of objects depending
   // on the |playout_mode|.
   static DecisionLogic* Create(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);
 
   // Constructor.
   DecisionLogic(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);
 
   // Destructor.
   virtual ~DecisionLogic() {}
 
   // Resets object to a clean state.
   void Reset();
 
   // Resets parts of the state. Typically done when switching codecs.
   void SoftReset();
 
   // Sets the sample rate and the output block size.
-  void SetSampleRate(int fs_hz, int output_size_samples);
+  void SetSampleRate(int fs_hz, size_t output_size_samples);
 
   // 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).
   // This method end with calling GetDecisionSpecialized to get the actual
   // return value.
   Operations GetDecision(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);
 
   // These methods test the |cng_state_| for different conditions.
   bool CngRfc3389On() const { return cng_state_ == kCngRfc3389On; }
   bool CngOff() const { return cng_state_ == kCngOff; }
 
   // Resets the |cng_state_| to kCngOff.
   void SetCngOff() { cng_state_ = kCngOff; }
 
   // Reports back to DecisionLogic whether the decision to do expand remains or
   // not. Note that this is necessary, since an expand decision can be changed
   // to kNormal in NetEqImpl::GetDecision if there is still enough data in the
   // sync buffer.
   virtual void ExpandDecision(Operations operation);
 
   // Adds |value| to |sample_memory_|.
   void AddSampleMemory(int32_t value) {
     sample_memory_ += value;
   }
 
   // Accessors and mutators.
   void set_sample_memory(int32_t value) { sample_memory_ = value; }
-  int generated_noise_samples() const { return generated_noise_samples_; }
-  void set_generated_noise_samples(int value) {
+  size_t generated_noise_samples() const { return generated_noise_samples_; }
+  void set_generated_noise_samples(size_t value) {
     generated_noise_samples_ = value;
   }
-  int packet_length_samples() const { return packet_length_samples_; }
-  void set_packet_length_samples(int value) {
+  size_t packet_length_samples() const { return packet_length_samples_; }
+  void set_packet_length_samples(size_t value) {
     packet_length_samples_ = value;
   }
   void set_prev_time_scale(bool value) { prev_time_scale_ = value; }
   NetEqPlayoutMode playout_mode() const { return playout_mode_; }
 
  protected:
   // The value 6 sets maximum time-stretch rate to about 100 ms/s.
   static const int kMinTimescaleInterval = 6;
 
   enum CngState {
     kCngOff,
     kCngRfc3389On,
     kCngInternalOn
   };
 
   // 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).
   // Should be implemented by derived classes.
   virtual 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) = 0;
 
   // Updates the |buffer_level_filter_| with the current buffer level
   // |buffer_size_packets|.
-  void FilterBufferLevel(int buffer_size_packets, Modes prev_mode);
+  void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
 
   DecoderDatabase* decoder_database_;
   const PacketBuffer& packet_buffer_;
   DelayManager* delay_manager_;
   BufferLevelFilter* buffer_level_filter_;
   int fs_mult_;
-  int output_size_samples_;
+  size_t output_size_samples_;
   CngState cng_state_;  // Remember if comfort noise is interrupted by other
                         // event (e.g., DTMF).
-  int generated_noise_samples_;
-  int packet_length_samples_;
+  size_t generated_noise_samples_;
+  size_t packet_length_samples_;
   int sample_memory_;
   bool prev_time_scale_;
   int timescale_hold_off_;
   int num_consecutive_expands_;
   const NetEqPlayoutMode playout_mode_;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_