blob: 65306a7b285b6d304350f21c7c0ab0b792caea1f [file] [log] [blame]
/*
* Copyright (c) 2017 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_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
#include <memory>
#include "modules/audio_processing/include/aec_dump.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockAecDump : public AecDump {
public:
MockAecDump();
virtual ~MockAecDump();
MOCK_METHOD2(WriteInitMessage,
void(const ProcessingConfig& api_format, int64_t time_now_ms));
MOCK_METHOD1(AddCaptureStreamInput,
void(const AudioFrameView<const float>& src));
MOCK_METHOD1(AddCaptureStreamOutput,
void(const AudioFrameView<const float>& src));
MOCK_METHOD3(AddCaptureStreamInput,
void(const int16_t* const data,
int num_channels,
int samples_per_channel));
MOCK_METHOD3(AddCaptureStreamOutput,
void(const int16_t* const data,
int num_channels,
int samples_per_channel));
MOCK_METHOD1(AddAudioProcessingState,
void(const AudioProcessingState& state));
MOCK_METHOD0(WriteCaptureStreamMessage, void());
MOCK_METHOD3(WriteRenderStreamMessage,
void(const int16_t* const data,
int num_channels,
int samples_per_channel));
MOCK_METHOD1(WriteRenderStreamMessage,
void(const AudioFrameView<const float>& src));
MOCK_METHOD1(WriteConfig, void(const InternalAPMConfig& config));
MOCK_METHOD1(WriteRuntimeSetting,
void(const AudioProcessing::RuntimeSetting& config));
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_