From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../test/aec_dump_based_simulator.h | 82 ++++++++++++++++++++++ 1 file changed, 82 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h (limited to 'third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h') diff --git a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h new file mode 100644 index 0000000000..e2c1f3e4ba --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h @@ -0,0 +1,82 @@ +/* + * 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 +#include + +#include "modules/audio_processing/test/audio_processing_simulator.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, + rtc::scoped_refptr audio_processing, + std::unique_ptr ap_builder); + + AecDumpBasedSimulator() = delete; + AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete; + AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete; + + ~AecDumpBasedSimulator() override; + + // Processes the messages in the aecdump file. + void Process() override; + + // Analyzes the data in the aecdump file and reports the resulting statistics. + void Analyze() override; + + private: + void HandleEvent(const webrtc::audioproc::Event& event_msg, + int& num_forward_chunks_processed, + int& init_index); + void HandleMessage(const webrtc::audioproc::Init& msg, int init_index); + 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> artificial_nearend_buf_; + std::unique_ptr artificial_nearend_buffer_reader_; + bool artificial_nearend_eof_reported_ = false; + InterfaceType interface_used_ = InterfaceType::kNotSpecified; + std::unique_ptr call_order_output_file_; + bool finished_processing_specified_init_block_ = false; +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ -- cgit v1.2.3