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efeeba0864
Bug: webrtc:15623 Change-Id: Ia184993769f74d51e68a5a536d5fdde26890bcfd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325481 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41058}
79 lines
2.9 KiB
C++
79 lines
2.9 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
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#define MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
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#include <fstream>
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#include <string>
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#include "modules/audio_processing/test/audio_processing_simulator.h"
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
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#else
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#include "modules/audio_processing/debug.pb.h"
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#endif
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namespace webrtc {
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namespace test {
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// Used to perform an audio processing simulation from an aec dump.
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class AecDumpBasedSimulator final : public AudioProcessingSimulator {
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public:
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AecDumpBasedSimulator(const SimulationSettings& settings,
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rtc::scoped_refptr<AudioProcessing> audio_processing,
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std::unique_ptr<AudioProcessingBuilder> ap_builder);
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AecDumpBasedSimulator() = delete;
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AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete;
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AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete;
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~AecDumpBasedSimulator() override;
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// Processes the messages in the aecdump file.
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void Process() override;
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// Analyzes the data in the aecdump file and reports the resulting statistics.
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void Analyze() override;
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private:
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void HandleEvent(const webrtc::audioproc::Event& event_msg,
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int& num_forward_chunks_processed,
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int& init_index);
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void HandleMessage(const webrtc::audioproc::Init& msg, int init_index);
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void HandleMessage(const webrtc::audioproc::Stream& msg);
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void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
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void HandleMessage(const webrtc::audioproc::Config& msg);
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void HandleMessage(const webrtc::audioproc::RuntimeSetting& msg);
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void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
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void PrepareReverseProcessStreamCall(
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const webrtc::audioproc::ReverseStream& msg);
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void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
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void MaybeOpenCallOrderFile();
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enum InterfaceType {
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kFixedInterface,
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kFloatInterface,
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kNotSpecified,
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};
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FILE* dump_input_file_;
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std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_;
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std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_;
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bool artificial_nearend_eof_reported_ = false;
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InterfaceType interface_used_ = InterfaceType::kNotSpecified;
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std::unique_ptr<std::ofstream> call_order_output_file_;
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bool finished_processing_specified_init_block_ = false;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
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