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

webrtc/modules/audio_device/test/audio_device_test_api.cc

 /*
  *  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 65 matching lines @@
 class AudioTransportAPI: public AudioTransport {
  public:
   AudioTransportAPI(AudioDeviceModule* audioDevice)
       : rec_count_(0),
         play_count_(0) {
   }
 
   ~AudioTransportAPI() {}
 
   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 sampleRate,
                                   const uint32_t totalDelay,
                                   const int32_t clockSkew,
                                   const uint32_t currentMicLevel,
                                   const bool keyPressed,
                                   uint32_t& newMicLevel) override {
     rec_count_++;
     if (rec_count_ % 100 == 0) {
       if (nChannels == 1) {
         // mono
         TEST_LOG("-");
       } else if ((nChannels == 2) && (nBytesPerSample == 2)) {
         // stereo but only using one channel
         TEST_LOG("-|");
       } else {
         // stereo
         TEST_LOG("--");
       }
     }
     return 0;
   }
 
-  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 sampleRate,
                            void* audioSamples,
-                           uint32_t& nSamplesOut,
+                           size_t& nSamplesOut,
                            int64_t* elapsed_time_ms,
                            int64_t* ntp_time_ms) override {
     play_count_++;
     if (play_count_ % 100 == 0) {
       if (nChannels == 1) {
         TEST_LOG("+");
       } else {
         TEST_LOG("++");
       }
     }
     nSamplesOut = 480;
     return 0;
   }
 
   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 {
     return 0;
   }
 
   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 {}
  private:
   uint32_t rec_count_;
   uint32_t play_count_;
 };
 
 class AudioDeviceAPITest: public testing::Test {
  protected:
@@ skipping 1661 matching lines @@
   // TODO(kjellander): Fix so these tests pass on Mac.
 #if !defined(WEBRTC_MAC)
   EXPECT_EQ(0, audio_device_->InitPlayout());
   EXPECT_EQ(0, audio_device_->StartPlayout());
 #endif
 
   EXPECT_EQ(-1, audio_device_->GetLoudspeakerStatus(&loudspeakerOn));
 #endif
   EXPECT_EQ(0, audio_device_->StopPlayout());
 }