1 /* 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ 12 #define MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ 13 14 #include <fstream> 15 #include <string> 16 17 #include "modules/audio_processing/test/audio_processing_simulator.h" 18 #include "rtc_base/ignore_wundef.h" 19 20 RTC_PUSH_IGNORING_WUNDEF() 21 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD 22 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" 23 #else 24 #include "modules/audio_processing/debug.pb.h" 25 #endif RTC_POP_IGNORING_WUNDEF()26RTC_POP_IGNORING_WUNDEF() 27 28 namespace webrtc { 29 namespace test { 30 31 // Used to perform an audio processing simulation from an aec dump. 32 class AecDumpBasedSimulator final : public AudioProcessingSimulator { 33 public: 34 AecDumpBasedSimulator(const SimulationSettings& settings, 35 rtc::scoped_refptr<AudioProcessing> audio_processing, 36 std::unique_ptr<AudioProcessingBuilder> ap_builder); 37 38 AecDumpBasedSimulator() = delete; 39 AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete; 40 AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete; 41 42 ~AecDumpBasedSimulator() override; 43 44 // Processes the messages in the aecdump file. 45 void Process() override; 46 47 // Analyzes the data in the aecdump file and reports the resulting statistics. 48 void Analyze() override; 49 50 private: 51 void HandleEvent(const webrtc::audioproc::Event& event_msg, 52 int& num_forward_chunks_processed, 53 int& init_index); 54 void HandleMessage(const webrtc::audioproc::Init& msg, int init_index); 55 void HandleMessage(const webrtc::audioproc::Stream& msg); 56 void HandleMessage(const webrtc::audioproc::ReverseStream& msg); 57 void HandleMessage(const webrtc::audioproc::Config& msg); 58 void HandleMessage(const webrtc::audioproc::RuntimeSetting& msg); 59 void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg); 60 void PrepareReverseProcessStreamCall( 61 const webrtc::audioproc::ReverseStream& msg); 62 void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg); 63 void MaybeOpenCallOrderFile(); 64 enum InterfaceType { 65 kFixedInterface, 66 kFloatInterface, 67 kNotSpecified, 68 }; 69 70 FILE* dump_input_file_; 71 std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_; 72 std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_; 73 bool artificial_nearend_eof_reported_ = false; 74 InterfaceType interface_used_ = InterfaceType::kNotSpecified; 75 std::unique_ptr<std::ofstream> call_order_output_file_; 76 bool finished_processing_specified_init_block_ = false; 77 }; 78 79 } // namespace test 80 } // namespace webrtc 81 82 #endif // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ 83