| /* | |
| * Copyright (c) 2013 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_INCLUDE_MOCK_AUDIO_PROCESSING_H_ | |
| #define MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_ | |
| #include <memory> | |
| #include "modules/audio_processing/audio_buffer.h" | |
| #include "modules/audio_processing/include/aec_dump.h" | |
| #include "modules/audio_processing/include/audio_processing.h" | |
| #include "modules/audio_processing/include/audio_processing_statistics.h" | |
| #include "test/gmock.h" | |
| namespace webrtc { | |
| namespace test { | |
| class MockCustomProcessing : public CustomProcessing { | |
| public: | |
| virtual ~MockCustomProcessing() {} | |
| MOCK_METHOD2(Initialize, void(int sample_rate_hz, int num_channels)); | |
| MOCK_METHOD1(Process, void(AudioBuffer* audio)); | |
| MOCK_METHOD1(SetRuntimeSetting, | |
| void(AudioProcessing::RuntimeSetting setting)); | |
| MOCK_CONST_METHOD0(ToString, std::string()); | |
| }; | |
| class MockCustomAudioAnalyzer : public CustomAudioAnalyzer { | |
| public: | |
| virtual ~MockCustomAudioAnalyzer() {} | |
| MOCK_METHOD2(Initialize, void(int sample_rate_hz, int num_channels)); | |
| MOCK_METHOD1(Analyze, void(const AudioBuffer* audio)); | |
| MOCK_CONST_METHOD0(ToString, std::string()); | |
| }; | |
| class MockEchoControl : public EchoControl { | |
| public: | |
| virtual ~MockEchoControl() {} | |
| MOCK_METHOD1(AnalyzeRender, void(AudioBuffer* render)); | |
| MOCK_METHOD1(AnalyzeCapture, void(AudioBuffer* capture)); | |
| MOCK_METHOD2(ProcessCapture, | |
| void(AudioBuffer* capture, bool echo_path_change)); | |
| MOCK_METHOD3(ProcessCapture, | |
| void(AudioBuffer* capture, | |
| AudioBuffer* linear_output, | |
| bool echo_path_change)); | |
| MOCK_CONST_METHOD0(GetMetrics, Metrics()); | |
| MOCK_METHOD1(SetAudioBufferDelay, void(int delay_ms)); | |
| MOCK_CONST_METHOD0(ActiveProcessing, bool()); | |
| }; | |
| class MockAudioProcessing : public ::testing::NiceMock<AudioProcessing> { | |
| public: | |
| MockAudioProcessing() {} | |
| virtual ~MockAudioProcessing() {} | |
| MOCK_METHOD0(Initialize, int()); | |
| MOCK_METHOD6(Initialize, | |
| int(int capture_input_sample_rate_hz, | |
| int capture_output_sample_rate_hz, | |
| int render_sample_rate_hz, | |
| ChannelLayout capture_input_layout, | |
| ChannelLayout capture_output_layout, | |
| ChannelLayout render_input_layout)); | |
| MOCK_METHOD1(Initialize, int(const ProcessingConfig& processing_config)); | |
| MOCK_METHOD1(ApplyConfig, void(const Config& config)); | |
| MOCK_METHOD1(SetExtraOptions, void(const webrtc::Config& config)); | |
| MOCK_CONST_METHOD0(proc_sample_rate_hz, int()); | |
| MOCK_CONST_METHOD0(proc_split_sample_rate_hz, int()); | |
| MOCK_CONST_METHOD0(num_input_channels, size_t()); | |
| MOCK_CONST_METHOD0(num_proc_channels, size_t()); | |
| MOCK_CONST_METHOD0(num_output_channels, size_t()); | |
| MOCK_CONST_METHOD0(num_reverse_channels, size_t()); | |
| MOCK_METHOD1(set_output_will_be_muted, void(bool muted)); | |
| MOCK_METHOD1(SetRuntimeSetting, void(RuntimeSetting setting)); | |
| MOCK_METHOD1(ProcessStream, int(AudioFrame* frame)); | |
| MOCK_METHOD4(ProcessStream, | |
| int(const int16_t* const src, | |
| const StreamConfig& input_config, | |
| const StreamConfig& output_config, | |
| int16_t* const dest)); | |
| MOCK_METHOD7(ProcessStream, | |
| int(const float* const* src, | |
| size_t samples_per_channel, | |
| int input_sample_rate_hz, | |
| ChannelLayout input_layout, | |
| int output_sample_rate_hz, | |
| ChannelLayout output_layout, | |
| float* const* dest)); | |
| MOCK_METHOD4(ProcessStream, | |
| int(const float* const* src, | |
| const StreamConfig& input_config, | |
| const StreamConfig& output_config, | |
| float* const* dest)); | |
| MOCK_METHOD1(ProcessReverseStream, int(AudioFrame* frame)); | |
| MOCK_METHOD4(ProcessReverseStream, | |
| int(const int16_t* const src, | |
| const StreamConfig& input_config, | |
| const StreamConfig& output_config, | |
| int16_t* const dest)); | |
| MOCK_METHOD4(AnalyzeReverseStream, | |
| int(const float* const* data, | |
| size_t samples_per_channel, | |
| int sample_rate_hz, | |
| ChannelLayout layout)); | |
| MOCK_METHOD2(AnalyzeReverseStream, | |
| int(const float* const* data, | |
| const StreamConfig& reverse_config)); | |
| MOCK_METHOD4(ProcessReverseStream, | |
| int(const float* const* src, | |
| const StreamConfig& input_config, | |
| const StreamConfig& output_config, | |
| float* const* dest)); | |
| MOCK_CONST_METHOD1( | |
| GetLinearAecOutput, | |
| bool(rtc::ArrayView<std::array<float, 160>> linear_output)); | |
| MOCK_METHOD1(set_stream_delay_ms, int(int delay)); | |
| MOCK_CONST_METHOD0(stream_delay_ms, int()); | |
| MOCK_CONST_METHOD0(was_stream_delay_set, bool()); | |
| MOCK_METHOD1(set_stream_key_pressed, void(bool key_pressed)); | |
| MOCK_METHOD1(set_delay_offset_ms, void(int offset)); | |
| MOCK_CONST_METHOD0(delay_offset_ms, int()); | |
| MOCK_METHOD1(set_stream_analog_level, void(int)); | |
| MOCK_CONST_METHOD0(recommended_stream_analog_level, int()); | |
| virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {} | |
| MOCK_METHOD0(DetachAecDump, void()); | |
| virtual void AttachPlayoutAudioGenerator( | |
| std::unique_ptr<AudioGenerator> audio_generator) {} | |
| MOCK_METHOD0(DetachPlayoutAudioGenerator, void()); | |
| MOCK_METHOD0(UpdateHistogramsOnCallEnd, void()); | |
| MOCK_METHOD0(GetStatistics, AudioProcessingStats()); | |
| MOCK_METHOD1(GetStatistics, AudioProcessingStats(bool)); | |
| MOCK_CONST_METHOD0(GetConfig, AudioProcessing::Config()); | |
| }; | |
| } // namespace test | |
| } // namespace webrtc | |
| #endif // MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_ |