/* * Copyright (c) 2016 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. */ #ifndef MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ #define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ #include "modules/audio_device/include/audio_device_defines.h" #include "test/gmock.h" namespace webrtc { namespace test { class MockAudioTransport : public AudioTransport { public: MockAudioTransport() {} ~MockAudioTransport() {} MOCK_METHOD(int32_t, RecordedDataIsAvailable, (const void* audioSamples, size_t nSamples, size_t nBytesPerSample, size_t nChannels, uint32_t samplesPerSec, uint32_t totalDelayMS, int32_t clockDrift, uint32_t currentMicLevel, bool keyPressed, uint32_t& newMicLevel), (override)); MOCK_METHOD(int32_t, RecordedDataIsAvailable, (const void* audioSamples, size_t nSamples, size_t nBytesPerSample, size_t nChannels, uint32_t samplesPerSec, uint32_t totalDelayMS, int32_t clockDrift, uint32_t currentMicLevel, bool keyPressed, uint32_t& newMicLevel, absl::optional estimated_capture_time_ns), (override)); MOCK_METHOD(int32_t, NeedMorePlayData, (size_t nSamples, size_t nBytesPerSample, size_t nChannels, uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms), (override)); MOCK_METHOD(void, PullRenderData, (int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames, void* audio_data, int64_t* elapsed_time_ms, int64_t* ntp_time_ms), (override)); }; } // namespace test } // namespace webrtc #endif // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_