[NOT FOR REVIEW] Convert channel counts to size_t.

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 31 matching lines @@
     TTDeviceRemoval = 13,
     TTMobileAPI = 14,
     TTTest = 66,
 };
 
 struct AudioPacket
 {
     uint8_t dataBuffer[4 * 960];
     size_t nSamples;
     size_t nBytesPerSample;
-    uint8_t nChannels;
+    size_t nChannels;
     uint32_t samplesPerSec;
 };
 
 class ProcessThread;
 
 namespace webrtc
 {
 
 class AudioDeviceModule;
 class AudioEventObserver;
@@ skipping 18 matching lines @@
 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------
 
 class AudioTransportImpl: public AudioTransport
 {
 public:
     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;
 
     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;
 
-    int OnDataAvailable(const int voe_channels[],
-                        int number_of_voe_channels,
-                        const int16_t* audio_data,
-                        int sample_rate,
-                        int number_of_channels,
-                        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,
-                         size_t number_of_frames) override;
-
-    void PullRenderData(int bits_per_sample, int sample_rate,
-                        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);
     void SetSpeakerVolume(bool enable)
     {
         _speakerVolume = enable;
     }
@@ skipping 90 matching lines @@
 
     rtc::scoped_ptr<ProcessThread> _processThread;
     AudioDeviceModule* _audioDevice;
     AudioEventObserver* _audioEventObserver;
     AudioTransportImpl* _audioTransport;
 };
 
 }  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H