Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/merge.h

Index: webrtc/modules/audio_coding/neteq/merge.h
diff --git a/webrtc/modules/audio_coding/neteq/merge.h b/webrtc/modules/audio_coding/neteq/merge.h
index 1b60aeca8f38ba3593ea6aa0411c09c7d83a5b23..4f74a94e7df8e8dd18da2347cdbe5be9eb8bed6b 100644
--- a/webrtc/modules/audio_coding/neteq/merge.h
+++ b/webrtc/modules/audio_coding/neteq/merge.h
@@ -46,11 +46,11 @@ class Merge {
   // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
   // will be used to scale the audio, and is updated in the process. The array
   // must have |num_channels_| elements.
-  virtual int Process(int16_t* input, size_t input_length,
-                      int16_t* external_mute_factor_array,
-                      AudioMultiVector* output);
+  virtual size_t Process(int16_t* input, size_t input_length,
+                         int16_t* external_mute_factor_array,
+                         AudioMultiVector* output);
 
-  virtual int RequiredFutureSamples();
+  virtual size_t RequiredFutureSamples();
 
  protected:
   const int fs_hz_;
@@ -58,38 +58,38 @@ class Merge {
 
  private:
   static const int kMaxSampleRate = 48000;
-  static const int kExpandDownsampLength = 100;
-  static const int kInputDownsampLength = 40;
-  static const int kMaxCorrelationLength = 60;
+  static const size_t kExpandDownsampLength = 100;
+  static const size_t kInputDownsampLength = 40;
+  static const size_t kMaxCorrelationLength = 60;
 
   // Calls |expand_| to get more expansion data to merge with. The data is
   // written to |expanded_signal_|. Returns the length of the expanded data,
   // while |expand_period| will be the number of samples in one expansion period
   // (typically one pitch period). The value of |old_length| will be the number
   // of samples that were taken from the |sync_buffer_|.
-  int GetExpandedSignal(int* old_length, int* expand_period);
+  size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
 
   // Analyzes |input| and |expanded_signal| to find maximum values. Returns
   // a muting factor (Q14) to be used on the new data.
-  int16_t SignalScaling(const int16_t* input, int input_length,
+  int16_t SignalScaling(const int16_t* input, size_t input_length,
                         const int16_t* expanded_signal,
                         int16_t* expanded_max, int16_t* input_max) const;
 
   // Downsamples |input| (|input_length| samples) and |expanded_signal| to
   // 4 kHz sample rate. The downsampled signals are written to
   // |input_downsampled_| and |expanded_downsampled_|, respectively.
-  void Downsample(const int16_t* input, int input_length,
-                  const int16_t* expanded_signal, int expanded_length);
+  void Downsample(const int16_t* input, size_t input_length,
+                  const int16_t* expanded_signal, size_t expanded_length);
 
   // Calculates cross-correlation between |input_downsampled_| and
   // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
   // lag is returned.
-  int16_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
-                                 int start_position, int input_length,
-                                 int expand_period) const;
+  size_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
+                                size_t start_position, size_t input_length,
+                                size_t expand_period) const;
 
-  const int fs_mult_;  // fs_hz_ / 8000.
-  const int timestamps_per_call_;
+  const size_t fs_mult_;  // fs_hz_ / 8000.

[hlundin-webrtc, 2015/08/10 11:30:01]

Not size_t

+  const size_t timestamps_per_call_;
   Expand* expand_;
   SyncBuffer* sync_buffer_;
   int16_t expanded_downsampled_[kExpandDownsampLength];