Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
287
webrtc/modules/audio_processing/logging/apm_data_dumper.h
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287
webrtc/modules/audio_processing/logging/apm_data_dumper.h
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/*
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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_LOGGING_APM_DATA_DUMPER_H_
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#define MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
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#include <stdint.h>
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#include <stdio.h>
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#include <string.h>
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#include <string>
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#if WEBRTC_APM_DEBUG_DUMP == 1
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#include <memory>
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#include <unordered_map>
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#endif
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#include "api/array_view.h"
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#if WEBRTC_APM_DEBUG_DUMP == 1
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#include "common_audio/wav_file.h"
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#include "rtc_base/checks.h"
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#endif
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// Check to verify that the define is properly set.
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#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
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(WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
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#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
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#endif
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namespace webrtc {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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// Functor used to use as a custom deleter in the map of file pointers to raw
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// files.
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struct RawFileCloseFunctor {
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void operator()(FILE* f) const { fclose(f); }
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};
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#endif
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// Class that handles dumping of variables into files.
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class ApmDataDumper {
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public:
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// Constructor that takes an instance index that may
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// be used to distinguish data dumped from different
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// instances of the code.
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explicit ApmDataDumper(int instance_index);
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ApmDataDumper() = delete;
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ApmDataDumper(const ApmDataDumper&) = delete;
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ApmDataDumper& operator=(const ApmDataDumper&) = delete;
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~ApmDataDumper();
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// Activates or deactivate the dumping functionality.
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static void SetActivated(bool activated) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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recording_activated_ = activated;
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#endif
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}
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// Set an optional output directory.
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static void SetOutputDirectory(const std::string& output_dir) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength);
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strncpy(output_dir_, output_dir.c_str(), output_dir.size());
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#endif
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}
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// Reinitializes the data dumping such that new versions
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// of all files being dumped to are created.
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void InitiateNewSetOfRecordings() {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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++recording_set_index_;
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#endif
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}
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// Methods for performing dumping of data of various types into
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// various formats.
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void DumpRaw(const char* name, double v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v_length, const double* v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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}
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#endif
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}
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void DumpRaw(const char* name, rtc::ArrayView<const double> v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpRaw(name, v.size(), v.data());
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}
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#endif
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}
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void DumpRaw(const char* name, float v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v_length, const float* v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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}
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#endif
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}
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void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpRaw(name, v.size(), v.data());
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}
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#endif
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}
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void DumpRaw(const char* name, bool v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpRaw(name, static_cast<int16_t>(v));
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v_length, const bool* v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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for (size_t k = 0; k < v_length; ++k) {
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int16_t value = static_cast<int16_t>(v[k]);
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fwrite(&value, sizeof(value), 1, file);
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}
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}
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#endif
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}
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void DumpRaw(const char* name, rtc::ArrayView<const bool> v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpRaw(name, v.size(), v.data());
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}
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#endif
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}
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void DumpRaw(const char* name, int16_t v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v_length, const int16_t* v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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}
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#endif
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}
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void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpRaw(name, v.size(), v.data());
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}
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#endif
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}
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void DumpRaw(const char* name, int32_t v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v_length, const int32_t* v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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}
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#endif
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}
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void DumpRaw(const char* name, size_t v_length, const size_t* v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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}
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#endif
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}
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void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpRaw(name, v.size(), v.data());
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}
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#endif
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}
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void DumpRaw(const char* name, rtc::ArrayView<const size_t> v) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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DumpRaw(name, v.size(), v.data());
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#endif
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}
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void DumpWav(const char* name,
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size_t v_length,
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const float* v,
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int sample_rate_hz,
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int num_channels) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels,
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WavFile::SampleFormat::kFloat);
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file->WriteSamples(v, v_length);
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}
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#endif
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}
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void DumpWav(const char* name,
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rtc::ArrayView<const float> v,
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int sample_rate_hz,
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int num_channels) {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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if (recording_activated_) {
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DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
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}
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#endif
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}
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private:
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#if WEBRTC_APM_DEBUG_DUMP == 1
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static bool recording_activated_;
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static constexpr size_t kOutputDirMaxLength = 1024;
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static char output_dir_[kOutputDirMaxLength];
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const int instance_index_;
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int recording_set_index_ = 0;
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std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>
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raw_files_;
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std::unordered_map<std::string, std::unique_ptr<WavWriter>> wav_files_;
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FILE* GetRawFile(const char* name);
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WavWriter* GetWavFile(const char* name,
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int sample_rate_hz,
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int num_channels,
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WavFile::SampleFormat format);
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#endif
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
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