| /* |
| * Copyright (c) 2016 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_TEST_AEC_DUMP_BASED_SIMULATOR_H_ |
| #define MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ |
| |
| #include <fstream> |
| #include <string> |
| |
| #include "modules/audio_processing/test/audio_processing_simulator.h" |
| |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/ignore_wundef.h" |
| |
| RTC_PUSH_IGNORING_WUNDEF() |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" |
| #else |
| #include "modules/audio_processing/debug.pb.h" |
| #endif |
| RTC_POP_IGNORING_WUNDEF() |
| |
| namespace webrtc { |
| namespace test { |
| |
| // Used to perform an audio processing simulation from an aec dump. |
| class AecDumpBasedSimulator final : public AudioProcessingSimulator { |
| public: |
| AecDumpBasedSimulator(const SimulationSettings& settings, |
| std::unique_ptr<AudioProcessingBuilder> ap_builder); |
| ~AecDumpBasedSimulator() override; |
| |
| // Processes the messages in the aecdump file. |
| void Process() override; |
| |
| private: |
| void HandleMessage(const webrtc::audioproc::Init& msg); |
| void HandleMessage(const webrtc::audioproc::Stream& msg); |
| void HandleMessage(const webrtc::audioproc::ReverseStream& msg); |
| void HandleMessage(const webrtc::audioproc::Config& msg); |
| void HandleMessage(const webrtc::audioproc::RuntimeSetting& msg); |
| void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg); |
| void PrepareReverseProcessStreamCall( |
| const webrtc::audioproc::ReverseStream& msg); |
| void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg); |
| void MaybeOpenCallOrderFile(); |
| enum InterfaceType { |
| kFixedInterface, |
| kFloatInterface, |
| kNotSpecified, |
| }; |
| |
| FILE* dump_input_file_; |
| std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_; |
| std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_; |
| bool artificial_nearend_eof_reported_ = false; |
| InterfaceType interface_used_ = InterfaceType::kNotSpecified; |
| std::unique_ptr<std::ofstream> call_order_output_file_; |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator); |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ |