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656d609a95
Bug: webrtc:9616 Change-Id: I1350212f0b8835fb64427483269da96d51670c01 Reviewed-on: https://webrtc-review.googlesource.com/92620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24267}
52 lines
1.7 KiB
C++
52 lines
1.7 KiB
C++
/*
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* Copyright (c) 2017 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 MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
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#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
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#include <memory>
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#include "modules/audio_processing/include/aec_dump.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockAecDump : public AecDump {
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public:
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MockAecDump();
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virtual ~MockAecDump();
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MOCK_METHOD2(WriteInitMessage,
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void(const ProcessingConfig& api_format, int64_t time_now_ms));
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MOCK_METHOD1(AddCaptureStreamInput,
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void(const AudioFrameView<const float>& src));
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MOCK_METHOD1(AddCaptureStreamOutput,
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void(const AudioFrameView<const float>& src));
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MOCK_METHOD1(AddCaptureStreamInput, void(const AudioFrame& frame));
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MOCK_METHOD1(AddCaptureStreamOutput, void(const AudioFrame& frame));
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MOCK_METHOD1(AddAudioProcessingState,
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void(const AudioProcessingState& state));
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MOCK_METHOD0(WriteCaptureStreamMessage, void());
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MOCK_METHOD1(WriteRenderStreamMessage, void(const AudioFrame& frame));
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MOCK_METHOD1(WriteRenderStreamMessage,
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void(const AudioFrameView<const float>& src));
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MOCK_METHOD1(WriteConfig, void(const InternalAPMConfig& config));
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
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