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/*
* 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 <fstream>
#include <string>
#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<AudioProcessing> audio_processing,
std::unique_ptr<AudioProcessingBuilder> 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<ChannelBuffer<float>> artificial_nearend_buf_;
std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_;
bool artificial_nearend_eof_reported_ = false;
InterfaceType interface_used_ = InterfaceType::kNotSpecified;
std::unique_ptr<std::ofstream> 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_
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