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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/logging | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/logging')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc | 93 | ||||
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h | 413 |
2 files changed, 506 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc new file mode 100644 index 0000000000..65d2167d37 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc @@ -0,0 +1,93 @@ +/* + * 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. + */ + +#include "modules/audio_processing/logging/apm_data_dumper.h" + +#include "absl/strings/string_view.h" +#include "rtc_base/strings/string_builder.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 { +namespace { + +#if WEBRTC_APM_DEBUG_DUMP == 1 + +#if defined(WEBRTC_WIN) +constexpr char kPathDelimiter = '\\'; +#else +constexpr char kPathDelimiter = '/'; +#endif + +std::string FormFileName(absl::string_view output_dir, + absl::string_view name, + int instance_index, + int reinit_index, + absl::string_view suffix) { + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); + if (!output_dir.empty()) { + ss << output_dir; + if (output_dir.back() != kPathDelimiter) { + ss << kPathDelimiter; + } + } + ss << name << "_" << instance_index << "-" << reinit_index << suffix; + return ss.str(); +} +#endif + +} // namespace + +#if WEBRTC_APM_DEBUG_DUMP == 1 +ApmDataDumper::ApmDataDumper(int instance_index) + : instance_index_(instance_index) {} +#else +ApmDataDumper::ApmDataDumper(int instance_index) {} +#endif + +ApmDataDumper::~ApmDataDumper() = default; + +#if WEBRTC_APM_DEBUG_DUMP == 1 +bool ApmDataDumper::recording_activated_ = false; +absl::optional<int> ApmDataDumper::dump_set_to_use_; +char ApmDataDumper::output_dir_[] = ""; + +FILE* ApmDataDumper::GetRawFile(absl::string_view name) { + std::string filename = FormFileName(output_dir_, name, instance_index_, + recording_set_index_, ".dat"); + auto& f = raw_files_[filename]; + if (!f) { + f.reset(fopen(filename.c_str(), "wb")); + RTC_CHECK(f.get()) << "Cannot write to " << filename << "."; + } + return f.get(); +} + +WavWriter* ApmDataDumper::GetWavFile(absl::string_view name, + int sample_rate_hz, + int num_channels, + WavFile::SampleFormat format) { + std::string filename = FormFileName(output_dir_, name, instance_index_, + recording_set_index_, ".wav"); + auto& f = wav_files_[filename]; + if (!f) { + f.reset( + new WavWriter(filename.c_str(), sample_rate_hz, num_channels, format)); + } + return f.get(); +} +#endif + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h new file mode 100644 index 0000000000..76f8b34c0b --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h @@ -0,0 +1,413 @@ +/* + * 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 <memory> +#include <string> +#include <unordered_map> +#endif + +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/array_view.h" +#if WEBRTC_APM_DEBUG_DUMP == 1 +#include "common_audio/wav_file.h" +#include "rtc_base/checks.h" +#include "rtc_base/string_utils.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 + } + + // Returns whether dumping functionality is enabled/available. + static bool IsAvailable() { +#if WEBRTC_APM_DEBUG_DUMP == 1 + return true; +#else + return false; +#endif + } + + // Default dump set. + static constexpr size_t kDefaultDumpSet = 0; + + // Specifies what dump set to use. All dump commands with a different dump set + // than the one specified will be discarded. If not specificed, all dump sets + // will be used. + static void SetDumpSetToUse(int dump_set_to_use) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + dump_set_to_use_ = dump_set_to_use; +#endif + } + + // Set an optional output directory. + static void SetOutputDirectory(absl::string_view output_dir) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength); + rtc::strcpyn(output_dir_, kOutputDirMaxLength, output_dir); +#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(absl::string_view name, + double v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(&v, sizeof(v), 1, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v_length, + const double* v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(v, sizeof(v[0]), v_length, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + rtc::ArrayView<const double> v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + DumpRaw(name, v.size(), v.data()); + } +#endif + } + + void DumpRaw(absl::string_view name, + float v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(&v, sizeof(v), 1, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v_length, + const float* v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(v, sizeof(v[0]), v_length, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + rtc::ArrayView<const float> v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + DumpRaw(name, v.size(), v.data()); + } +#endif + } + + void DumpRaw(absl::string_view name, bool v, int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + DumpRaw(name, static_cast<int16_t>(v)); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v_length, + const bool* v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + 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(absl::string_view name, + rtc::ArrayView<const bool> v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + DumpRaw(name, v.size(), v.data()); + } +#endif + } + + void DumpRaw(absl::string_view name, + int16_t v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(&v, sizeof(v), 1, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v_length, + const int16_t* v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(v, sizeof(v[0]), v_length, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + rtc::ArrayView<const int16_t> v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + DumpRaw(name, v.size(), v.data()); + } +#endif + } + + void DumpRaw(absl::string_view name, + int32_t v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(&v, sizeof(v), 1, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v_length, + const int32_t* v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(v, sizeof(v[0]), v_length, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(&v, sizeof(v), 1, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + size_t v_length, + const size_t* v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + FILE* file = GetRawFile(name); + fwrite(v, sizeof(v[0]), v_length, file); + } +#endif + } + + void DumpRaw(absl::string_view name, + rtc::ArrayView<const int32_t> v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + DumpRaw(name, v.size(), v.data()); + } +#endif + } + + void DumpRaw(absl::string_view name, + rtc::ArrayView<const size_t> v, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + DumpRaw(name, v.size(), v.data()); +#endif + } + + void DumpWav(absl::string_view name, + size_t v_length, + const float* v, + int sample_rate_hz, + int num_channels, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + if (recording_activated_) { + WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels, + WavFile::SampleFormat::kFloat); + file->WriteSamples(v, v_length); + } +#endif + } + + void DumpWav(absl::string_view name, + rtc::ArrayView<const float> v, + int sample_rate_hz, + int num_channels, + int dump_set = kDefaultDumpSet) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + if (dump_set_to_use_ && *dump_set_to_use_ != dump_set) + return; + + 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 absl::optional<int> dump_set_to_use_; + 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(absl::string_view name); + WavWriter* GetWavFile(absl::string_view name, + int sample_rate_hz, + int num_channels, + WavFile::SampleFormat format); +#endif +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_ |