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The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
67 lines
2.3 KiB
C++
67 lines
2.3 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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explicit AecDumpBasedSimulator(const SimulationSettings& settings);
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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 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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