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

webrtc/modules/audio_device/test/func_test_manager.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 29 matching lines @@
     TTMicrophoneAGC = 10,
     TTLoopback = 11,
     TTDeviceRemoval = 13,
     TTMobileAPI = 14,
     TTTest = 66,
 };
 
 struct AudioPacket
 {
     uint8_t dataBuffer[4 * 960];
-    uint16_t nSamples;
-    uint16_t nBytesPerSample;
+    size_t nSamples;
+    size_t nBytesPerSample;
     uint8_t nChannels;
     uint32_t samplesPerSec;
 };
 
 class ProcessThread;
 
 namespace webrtc
 {
 
 class AudioDeviceModule;
@@ skipping 17 matching lines @@
 };
 
 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------
 
 class AudioTransportImpl: public AudioTransport
 {
 public:
     int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                    const uint32_t nSamples,
-                                    const uint8_t nBytesPerSample,
+                                    const size_t nSamples,
+                                    const size_t nBytesPerSample,
                                     const uint8_t nChannels,
                                     const uint32_t samplesPerSec,
                                     const uint32_t totalDelayMS,
                                     const int32_t clockDrift,
                                     const uint32_t currentMicLevel,
                                     const bool keyPressed,
                                     uint32_t& newMicLevel) override;
 
-    int32_t NeedMorePlayData(const uint32_t nSamples,
-                             const uint8_t nBytesPerSample,
+    int32_t NeedMorePlayData(const size_t nSamples,
+                             const size_t nBytesPerSample,
                              const uint8_t nChannels,
                              const uint32_t samplesPerSec,
                              void* audioSamples,
-                             uint32_t& nSamplesOut,
+                             size_t& nSamplesOut,
                              int64_t* elapsed_time_ms,
                              int64_t* ntp_time_ms) override;
 
     int OnDataAvailable(const int voe_channels[],
                         int number_of_voe_channels,
                         const int16_t* audio_data,
                         int sample_rate,
                         int number_of_channels,
-                        int number_of_frames,
+                        size_t number_of_frames,
                         int audio_delay_milliseconds,
                         int current_volume,
                         bool key_pressed,
                         bool need_audio_processing) override;
 
     void PushCaptureData(int voe_channel, const void* audio_data,
                          int bits_per_sample, int sample_rate,
                          int number_of_channels,
-                         int number_of_frames) override;
+                         size_t number_of_frames) override;
 
     void PullRenderData(int bits_per_sample, int sample_rate,
-                        int number_of_channels, int number_of_frames,
+                        int number_of_channels, size_t number_of_frames,
                         void* audio_data,
                         int64_t* elapsed_time_ms,
                         int64_t* ntp_time_ms) override;
 
     AudioTransportImpl(AudioDeviceModule* audioDevice);
     ~AudioTransportImpl();
 
 public:
     int32_t SetFilePlayout(bool enable, const char* fileName = NULL);
     void SetFullDuplex(bool enable);
@@ skipping 94 matching lines @@
 
     rtc::scoped_ptr<ProcessThread> _processThread;
     AudioDeviceModule* _audioDevice;
     AudioEventObserver* _audioEventObserver;
     AudioTransportImpl* _audioTransport;
 };
 
 }  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H