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/*
* 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<int64_t> 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_
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