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

talk/app/webrtc/test/fakeaudiocapturemodule.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 597 matching lines @@
   // remote side unless a packet containing a sample of that magnitude has been
   // sent to it. Note that the audio processing pipeline will likely distort the
   // original signal.
   SetSendBuffer(kHighSampleValue);
   last_process_time_ms_ = rtc::Time();
   return true;
 }
 
 void FakeAudioCaptureModule::SetSendBuffer(int value) {
   Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
-  const int buffer_size_in_samples =
+  const size_t buffer_size_in_samples =
       sizeof(send_buffer_) / kNumberBytesPerSample;
-  for (int i = 0; i < buffer_size_in_samples; ++i) {
+  for (size_t i = 0; i < buffer_size_in_samples; ++i) {
     buffer_ptr[i] = value;
   }
 }
 
 void FakeAudioCaptureModule::ResetRecBuffer() {
   memset(rec_buffer_, 0, sizeof(rec_buffer_));
 }
 
 bool FakeAudioCaptureModule::CheckRecBuffer(int value) {
   const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_);
-  const int buffer_size_in_samples =
+  const size_t buffer_size_in_samples =
       sizeof(rec_buffer_) / kNumberBytesPerSample;
-  for (int i = 0; i < buffer_size_in_samples; ++i) {
+  for (size_t i = 0; i < buffer_size_in_samples; ++i) {
     if (buffer_ptr[i] >= value) return true;
   }
   return false;
 }
 
 bool FakeAudioCaptureModule::ShouldStartProcessing() {
   return recording_ || playing_;
 }
 
 void FakeAudioCaptureModule::UpdateProcessing(bool start) {
@@ skipping 47 matching lines @@
 }
 
 void FakeAudioCaptureModule::ReceiveFrameP() {
   ASSERT(process_thread_->IsCurrent());
   {
     rtc::CritScope cs(&crit_callback_);
     if (!audio_callback_) {
       return;
     }
     ResetRecBuffer();
-    uint32_t nSamplesOut = 0;
+    size_t nSamplesOut = 0;
     int64_t elapsed_time_ms = 0;
     int64_t ntp_time_ms = 0;
     if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
                                          kNumberOfChannels, kSamplesPerSecond,
                                          rec_buffer_, nSamplesOut,
                                          &elapsed_time_ms, &ntp_time_ms) != 0) {
       ASSERT(false);
     }
     ASSERT(nSamplesOut == kNumberSamples);
   }
@@ skipping 23 matching lines @@
                                               kNumberOfChannels,
                                               kSamplesPerSecond, kTotalDelayMs,
                                               kClockDriftMs, current_mic_level,
                                               key_pressed,
                                               current_mic_level) != 0) {
     ASSERT(false);
   }
   SetMicrophoneVolume(current_mic_level);
 }