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/*
* 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_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
#include "api/array_view.h"
#include "api/optional.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/render_delay_controller.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockRenderDelayController : public RenderDelayController {
public:
virtual ~MockRenderDelayController() = default;
MOCK_METHOD0(Reset, void());
MOCK_METHOD0(LogRenderCall, void());
MOCK_METHOD2(
GetDelay,
rtc::Optional<DelayEstimate>(const DownsampledRenderBuffer& render_buffer,
rtc::ArrayView<const float> capture));
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
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