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

webrtc/test/fake_audio_device.cc

 /*
  *  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 81 matching lines @@
 }
 
 void FakeAudioDevice::CaptureAudio() {
   {
     rtc::CritScope cs(&lock_);
     if (capturing_) {
       int bytes_read = file_utility_->ReadPCMData(
           *input_stream_.get(), captured_audio_, kBufferSizeBytes);
       if (bytes_read <= 0)
         return;
-      int num_samples = bytes_read / 2;  // 2 bytes per sample.
+      // 2 bytes per sample.
+      size_t num_samples = static_cast<size_t>(bytes_read / 2);
       uint32_t new_mic_level;
       EXPECT_EQ(0,
                 audio_callback_->RecordedDataIsAvailable(captured_audio_,
                                                          num_samples,
                                                          2,
                                                          1,
                                                          kFrequencyHz,
                                                          0,
                                                          0,
                                                          0,
                                                          false,
                                                          new_mic_level));
-      uint32_t samples_needed = kFrequencyHz / 100;
+      size_t samples_needed = kFrequencyHz / 100;
       int64_t now_ms = clock_->TimeInMilliseconds();
       uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
       if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
-        samples_needed = std::min(kFrequencyHz / time_since_last_playout_ms,
-                                  kBufferSizeBytes / 2);
+        samples_needed = std::min(
+            static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
+            kBufferSizeBytes / 2);
       }
-      uint32_t samples_out = 0;
+      size_t samples_out = 0;
       int64_t elapsed_time_ms = -1;
       int64_t ntp_time_ms = -1;
       EXPECT_EQ(0,
                 audio_callback_->NeedMorePlayData(samples_needed,
                                                   2,
                                                   1,
                                                   kFrequencyHz,
                                                   playout_buffer_,
                                                   samples_out,
                                                   &elapsed_time_ms,
                                                   &ntp_time_ms));
     }
   }
   tick_->Wait(WEBRTC_EVENT_INFINITE);
 }
 
 void FakeAudioDevice::Start() {
   rtc::CritScope cs(&lock_);
   capturing_ = true;
 }
 
 void FakeAudioDevice::Stop() {
   rtc::CritScope cs(&lock_);
   capturing_ = false;
 }
 }  // namespace test
 }  // namespace webrtc