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

webrtc/modules/audio_coding/neteq/interface/neteq.h

 /*
  *  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.
  */
 
@@ skipping 27 matching lines @@
   uint16_t speech_expand_rate;  // Fraction (of original stream) of synthesized
                                 // speech inserted through expansion (in Q14).
   uint16_t preemptive_rate;  // Fraction of data inserted through pre-emptive
                              // expansion (in Q14).
   uint16_t accelerate_rate;  // Fraction of data removed through acceleration
                              // (in Q14).
   uint16_t secondary_decoded_rate;  // Fraction of data coming from secondary
                                     // decoding (in Q14).
   int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
                            // (positive or negative).
-  int added_zero_samples;  // Number of zero samples added in "off" mode.
+  size_t added_zero_samples;  // Number of zero samples added in "off" mode.
 };
 
 enum NetEqOutputType {
   kOutputNormal,
   kOutputPLC,
   kOutputCNG,
   kOutputPLCtoCNG,
   kOutputVADPassive
 };
 
@@ skipping 21 matching lines @@
           // |max_delay_ms| has the same effect as calling SetMaximumDelay().
           max_delay_ms(2000),
           background_noise_mode(kBgnOff),
           playout_mode(kPlayoutOn),
           enable_fast_accelerate(false) {}
 
     std::string ToString() const;
 
     int sample_rate_hz;  // Initial value. Will change with input data.
     bool enable_audio_classifier;
-    int max_packets_in_buffer;
+    size_t max_packets_in_buffer;
     int max_delay_ms;
     BackgroundNoiseMode background_noise_mode;
     NetEqPlayoutMode playout_mode;
     bool enable_fast_accelerate;
   };
 
   enum ReturnCodes {
     kOK = 0,
     kFail = -1,
     kNotImplemented = -2
@@ skipping 57 matching lines @@
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
   // |output_audio|, which can hold (at least) |max_length| elements.
   // The number of channels that were written to the output is provided in
   // the output variable |num_channels|, and each channel contains
   // |samples_per_channel| elements. If more than one channel is written,
   // the samples are interleaved.
   // The speech type is written to |type|, if |type| is not NULL.
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(size_t max_length, int16_t* output_audio,
-                       int* samples_per_channel, int* num_channels,
+                       size_t* samples_per_channel, int* num_channels,
                        NetEqOutputType* type) = 0;
 
   // Associates |rtp_payload_type| with |codec| and stores the information in
   // the codec database. Returns 0 on success, -1 on failure.
   virtual int RegisterPayloadType(enum NetEqDecoder codec,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
   // decoder database. The decoder implements a decoder of type |codec| and
   // associates it with |rtp_payload_type|. The decoder will produce samples
@@ skipping 100 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_