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Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
69 lines
2.4 KiB
C++
69 lines
2.4 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 "modules/audio_processing/test/audio_processing_simulator.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/ignore_wundef.h"
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RTC_PUSH_IGNORING_WUNDEF()
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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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RTC_POP_IGNORING_WUNDEF()
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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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std::unique_ptr<AudioProcessingBuilder> ap_builder);
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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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private:
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void HandleMessage(const webrtc::audioproc::Init& msg);
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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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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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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator);
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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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