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libtgvoip/webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.h

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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_LOGGING_APM_DATA_DUMPER_H_
#define MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
#include <stdint.h>
#include <stdio.h>
#if WEBRTC_APM_DEBUG_DUMP == 1
#include <unordered_map>
#endif
#include "api/array_view.h"
#if WEBRTC_APM_DEBUG_DUMP == 1
#include "common_audio/wav_file.h"
#endif
#include "rtc_base/constructormagic.h"
// Check to verify that the define is properly set.
#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
(WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
#endif
namespace webrtc {
#if WEBRTC_APM_DEBUG_DUMP == 1
// Functor used to use as a custom deleter in the map of file pointers to raw
// files.
struct RawFileCloseFunctor {
void operator()(FILE* f) const { fclose(f); }
};
#endif
// Class that handles dumping of variables into files.
class ApmDataDumper {
public:
// Constructor that takes an instance index that may
// be used to distinguish data dumped from different
// instances of the code.
explicit ApmDataDumper(int instance_index);
~ApmDataDumper();
// Activates or deactivate the dumping functionality.
static void SetActivated(bool activated) {
#if WEBRTC_APM_DEBUG_DUMP == 1
recording_activated_ = activated;
#endif
}
// Reinitializes the data dumping such that new versions
// of all files being dumped to are created.
void InitiateNewSetOfRecordings() {
#if WEBRTC_APM_DEBUG_DUMP == 1
++recording_set_index_;
#endif
}
// Methods for performing dumping of data of various types into
// various formats.
void DumpRaw(const char* name, double v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(&v, sizeof(v), 1, file);
}
#endif
}
void DumpRaw(const char* name, size_t v_length, const double* v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(v, sizeof(v[0]), v_length, file);
}
#endif
}
void DumpRaw(const char* name, rtc::ArrayView<const double> v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpRaw(name, v.size(), v.data());
}
#endif
}
void DumpRaw(const char* name, float v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(&v, sizeof(v), 1, file);
}
#endif
}
void DumpRaw(const char* name, size_t v_length, const float* v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(v, sizeof(v[0]), v_length, file);
}
#endif
}
void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpRaw(name, v.size(), v.data());
}
#endif
}
void DumpRaw(const char* name, bool v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpRaw(name, static_cast<int16_t>(v));
}
#endif
}
void DumpRaw(const char* name, size_t v_length, const bool* v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
for (size_t k = 0; k < v_length; ++k) {
int16_t value = static_cast<int16_t>(v[k]);
fwrite(&value, sizeof(value), 1, file);
}
}
#endif
}
void DumpRaw(const char* name, rtc::ArrayView<const bool> v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpRaw(name, v.size(), v.data());
}
#endif
}
void DumpRaw(const char* name, int16_t v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(&v, sizeof(v), 1, file);
}
#endif
}
void DumpRaw(const char* name, size_t v_length, const int16_t* v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(v, sizeof(v[0]), v_length, file);
}
#endif
}
void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpRaw(name, v.size(), v.data());
}
#endif
}
void DumpRaw(const char* name, int32_t v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(&v, sizeof(v), 1, file);
}
#endif
}
void DumpRaw(const char* name, size_t v_length, const int32_t* v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(v, sizeof(v[0]), v_length, file);
}
#endif
}
void DumpRaw(const char* name, size_t v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(&v, sizeof(v), 1, file);
}
#endif
}
void DumpRaw(const char* name, size_t v_length, const size_t* v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
FILE* file = GetRawFile(name);
fwrite(v, sizeof(v[0]), v_length, file);
}
#endif
}
void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpRaw(name, v.size(), v.data());
}
#endif
}
void DumpWav(const char* name,
size_t v_length,
const float* v,
int sample_rate_hz,
int num_channels) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
file->WriteSamples(v, v_length);
}
#endif
}
void DumpWav(const char* name,
rtc::ArrayView<const float> v,
int sample_rate_hz,
int num_channels) {
#if WEBRTC_APM_DEBUG_DUMP == 1
if (recording_activated_) {
DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
}
#endif
}
private:
#if WEBRTC_APM_DEBUG_DUMP == 1
static bool recording_activated_;
const int instance_index_;
int recording_set_index_ = 0;
std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>
raw_files_;
std::unordered_map<std::string, std::unique_ptr<WavWriter>> wav_files_;
FILE* GetRawFile(const char* name);
WavWriter* GetWavFile(const char* name, int sample_rate_hz, int num_channels);
#endif
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_