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

webrtc/modules/audio_processing/include/mock_audio_processing.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 196 matching lines @@
   MOCK_CONST_METHOD0(num_reverse_channels,
       int());
   MOCK_METHOD1(set_output_will_be_muted,
       void(bool muted));
   MOCK_CONST_METHOD0(output_will_be_muted,
       bool());
   MOCK_METHOD1(ProcessStream,
       int(AudioFrame* frame));
   MOCK_METHOD7(ProcessStream,
       int(const float* const* src,
-          int samples_per_channel,
+          size_t samples_per_channel,
           int input_sample_rate_hz,
           ChannelLayout input_layout,
           int output_sample_rate_hz,
           ChannelLayout output_layout,
           float* const* dest));
   MOCK_METHOD4(ProcessStream,
                int(const float* const* src,
                    const StreamConfig& input_config,
                    const StreamConfig& output_config,
                    float* const* dest));
   MOCK_METHOD1(AnalyzeReverseStream,
       int(AudioFrame* frame));
   MOCK_METHOD4(AnalyzeReverseStream,
-      int(const float* const* data, int frames, int sample_rate_hz,
+      int(const float* const* data, size_t frames, int sample_rate_hz,
           ChannelLayout input_layout));
   MOCK_METHOD2(AnalyzeReverseStream,
       int(const float* const* data, const StreamConfig& reverse_config));
   MOCK_METHOD1(set_stream_delay_ms,
       int(int delay));
   MOCK_CONST_METHOD0(stream_delay_ms,
       int());
   MOCK_CONST_METHOD0(was_stream_delay_set,
       bool());
   MOCK_METHOD1(set_stream_key_pressed,
@@ skipping 39 matching lines @@
   rtc::scoped_ptr<MockGainControl> gain_control_;
   rtc::scoped_ptr<MockHighPassFilter> high_pass_filter_;
   rtc::scoped_ptr<MockLevelEstimator> level_estimator_;
   rtc::scoped_ptr<MockNoiseSuppression> noise_suppression_;
   rtc::scoped_ptr<MockVoiceDetection> voice_detection_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_