/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
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#include "testing/gmock/include/gmock/gmock.h"
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#include "webrtc/modules/audio_device/audio_device_buffer.h"
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namespace webrtc {
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class MockAudioDeviceBuffer : public AudioDeviceBuffer {
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public:
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MockAudioDeviceBuffer() {}
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virtual ~MockAudioDeviceBuffer() {}
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MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples));
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MOCK_METHOD1(GetPlayoutData, int32_t(void* audioBuffer));
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MOCK_METHOD2(SetRecordedBuffer,
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int32_t(const void* audioBuffer, size_t nSamples));
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MOCK_METHOD3(SetVQEData,
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void(int playDelayMS, int recDelayMS, int clockDrift));
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MOCK_METHOD0(DeliverRecordedData, int32_t());
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
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