| /* |
| * 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_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_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(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_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_CONST_METHOD1(GetStatistics, AudioProcessingStats(bool)); |
| |
| MOCK_CONST_METHOD0(GetConfig, AudioProcessing::Config()); |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_ |