Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/aec/aec_core.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.
  */
 
 /*
  * Specifies the interface for the AEC core.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_H_
 
+#include <stddef.h>
+
 #include "webrtc/typedefs.h"
 
 #define FRAME_LEN 80
 #define PART_LEN 64               // Length of partition
 #define PART_LEN1 (PART_LEN + 1)  // Unique fft coefficients
 #define PART_LEN2 (PART_LEN * 2)  // Length of partition * 2
 #define NUM_HIGH_BANDS_MAX  2     // Max number of high bands
 
 typedef float complex_t[2];
 // For performance reasons, some arrays of complex numbers are replaced by twice
@@ skipping 30 matching lines @@
 #if defined(MIPS_FPU_LE)
 void WebRtcAec_InitAec_mips(void);
 #endif
 #if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 void WebRtcAec_InitAec_neon(void);
 #endif
 
 void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend);
 void WebRtcAec_ProcessFrames(AecCore* aec,
                              const float* const* nearend,
-                             int num_bands,
-                             int num_samples,
+                             size_t num_bands,
+                             size_t num_samples,
                              int knownDelay,
                              float* const* out);
 
 // A helper function to call WebRtc_MoveReadPtr() for all far-end buffers.
 // Returns the number of elements moved, and adjusts |system_delay| by the
 // corresponding amount in ms.
 int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements);
 
 // Calculates the median, standard deviation and amount of poor values among the
 // delay estimates aggregated up to the first call to the function. After that
@@ skipping 38 matching lines @@
 // Returns the current |system_delay|, i.e., the buffered difference between
 // far-end and near-end.
 int WebRtcAec_system_delay(AecCore* self);
 
 // Sets the |system_delay| to |value|.  Note that if the value is changed
 // improperly, there can be a performance regression.  So it should be used with
 // care.
 void WebRtcAec_SetSystemDelay(AecCore* self, int delay);
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_H_