Convert channel counts to size_t.

talk/app/webrtc/test/fakeaudiocapturemodule_unittest.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 40 matching lines @@
   virtual void SetUp() {
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
     EXPECT_TRUE(fake_audio_capture_module_.get() != NULL);
   }
 
   // Callbacks inherited from webrtc::AudioTransport.
   // ADM is pushing data.
   int32_t RecordedDataIsAvailable(const void* audioSamples,
                                   const size_t nSamples,
                                   const size_t nBytesPerSample,
-                                  const uint8_t nChannels,
+                                  const size_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 {
     rtc::CritScope cs(&crit_);
     rec_buffer_bytes_ = nSamples * nBytesPerSample;
     if ((rec_buffer_bytes_ == 0) ||
         (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
          FakeAudioCaptureModule::kNumberBytesPerSample)) {
       ADD_FAILURE();
       return -1;
     }
     memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_);
     ++push_iterations_;
     newMicLevel = currentMicLevel;
     return 0;
   }
 
   // ADM is pulling data.
   int32_t NeedMorePlayData(const size_t nSamples,
                            const size_t nBytesPerSample,
-                           const uint8_t nChannels,
+                           const size_t nChannels,
                            const uint32_t samplesPerSec,
                            void* audioSamples,
                            size_t& nSamplesOut,
                            int64_t* elapsed_time_ms,
                            int64_t* ntp_time_ms) override {
     rtc::CritScope cs(&crit_);
     ++pull_iterations_;
     const size_t audio_buffer_size = nSamples * nBytesPerSample;
     const size_t bytes_out = RecordedDataReceived() ?
         CopyFromRecBuffer(audioSamples, audio_buffer_size):
@@ skipping 111 matching lines @@
 
   EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
   EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
 
   EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
   EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
 
   EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
   EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
 }