| /* |
| * 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> |
| #include <string.h> |
| |
| #include <string> |
| #if WEBRTC_APM_DEBUG_DUMP == 1 |
| #include <memory> |
| #include <unordered_map> |
| #endif |
| |
| #include "api/array_view.h" |
| #if WEBRTC_APM_DEBUG_DUMP == 1 |
| #include "common_audio/wav_file.h" |
| #include "rtc_base/checks.h" |
| #endif |
| |
| // 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() = delete; |
| ApmDataDumper(const ApmDataDumper&) = delete; |
| ApmDataDumper& operator=(const ApmDataDumper&) = delete; |
| |
| ~ApmDataDumper(); |
| |
| // Activates or deactivate the dumping functionality. |
| static void SetActivated(bool activated) { |
| #if WEBRTC_APM_DEBUG_DUMP == 1 |
| recording_activated_ = activated; |
| #endif |
| } |
| |
| // Set an optional output directory. |
| static void SetOutputDirectory(const std::string& output_dir) { |
| #if WEBRTC_APM_DEBUG_DUMP == 1 |
| RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength); |
| strncpy(output_dir_, output_dir.c_str(), output_dir.size()); |
| #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 DumpRaw(const char* name, rtc::ArrayView<const size_t> v) { |
| #if WEBRTC_APM_DEBUG_DUMP == 1 |
| 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, |
| WavFile::SampleFormat::kFloat); |
| 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_; |
| static constexpr size_t kOutputDirMaxLength = 1024; |
| static char output_dir_[kOutputDirMaxLength]; |
| 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, |
| WavFile::SampleFormat format); |
| #endif |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_ |