Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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74
modules/audio_processing/logging/apm_data_dumper.cc
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74
modules/audio_processing/logging/apm_data_dumper.cc
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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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#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
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#include <sstream>
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#include "webrtc/rtc_base/stringutils.h"
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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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namespace {
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#if WEBRTC_APM_DEBUG_DUMP == 1
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std::string FormFileName(const char* name,
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int instance_index,
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int reinit_index,
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const std::string& suffix) {
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std::stringstream ss;
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ss << name << "_" << instance_index << "-" << reinit_index << suffix;
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return ss.str();
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}
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#endif
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} // namespace
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#if WEBRTC_APM_DEBUG_DUMP == 1
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ApmDataDumper::ApmDataDumper(int instance_index)
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: instance_index_(instance_index) {}
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#else
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ApmDataDumper::ApmDataDumper(int instance_index) {}
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#endif
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ApmDataDumper::~ApmDataDumper() {}
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#if WEBRTC_APM_DEBUG_DUMP == 1
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FILE* ApmDataDumper::GetRawFile(const char* name) {
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std::string filename =
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FormFileName(name, instance_index_, recording_set_index_, ".dat");
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auto& f = raw_files_[filename];
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if (!f) {
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f.reset(fopen(filename.c_str(), "wb"));
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}
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return f.get();
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}
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WavWriter* ApmDataDumper::GetWavFile(const char* name,
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int sample_rate_hz,
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int num_channels) {
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std::string filename =
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FormFileName(name, instance_index_, recording_set_index_, ".wav");
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auto& f = wav_files_[filename];
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if (!f) {
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f.reset(new WavWriter(filename.c_str(), sample_rate_hz, num_channels));
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}
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return f.get();
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}
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#endif
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} // namespace webrtc
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212
modules/audio_processing/logging/apm_data_dumper.h
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212
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 WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
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#include <stdio.h>
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#include <memory>
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#include <string>
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#include <unordered_map>
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#include "webrtc/api/array_view.h"
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#include "webrtc/common_audio/wav_file.h"
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#include "webrtc/rtc_base/constructormagic.h"
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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();
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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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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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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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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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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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DumpRaw(name, v.size(), v.data());
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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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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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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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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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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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DumpRaw(name, v.size(), v.data());
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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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DumpRaw(name, static_cast<int16_t>(v));
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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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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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#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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DumpRaw(name, v.size(), v.data());
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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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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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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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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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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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DumpRaw(name, v.size(), v.data());
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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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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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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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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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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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FILE* file = GetRawFile(name);
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fwrite(&v, sizeof(v), 1, file);
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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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FILE* file = GetRawFile(name);
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fwrite(v, sizeof(v[0]), v_length, file);
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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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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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WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
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file->WriteSamples(v, v_length);
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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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DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
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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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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, int sample_rate_hz, int num_channels);
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#endif
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
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