| Index: webrtc/modules/audio_processing/logging/apm_data_dumper.cc
|
| diff --git a/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..97233693a66c5f56e6b0cefbf4cee6e2536c3e81
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
|
| @@ -0,0 +1,85 @@
|
| +/*
|
| + * 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 "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
|
| +
|
| +#include "webrtc/base/stringutils.h"
|
| +
|
| +// Check to verify that the define is properly set.
|
| +#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
|
| + (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
|
| +#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
|
| +#endif
|
| +
|
| +namespace webrtc {
|
| +
|
| +namespace {
|
| +
|
| +#if WEBRTC_AEC_DEBUG_DUMP == 1
|
| +std::string FormFileName(const std::string& name,
|
| + int instance_index,
|
| + int reinit_index,
|
| + const std::string& suffix) {
|
| + char instance_index_string[10];
|
| + rtc::sprintfn(instance_index_string, sizeof(instance_index_string), "%d",
|
| + instance_index);
|
| + char reinit_index_string[10];
|
| + rtc::sprintfn(reinit_index_string, sizeof(reinit_index_string), "%d",
|
| + reinit_index);
|
| + return name + "_" + instance_index_string + "-" +
|
| + std::to_string(reinit_index) + suffix;
|
| +}
|
| +#endif
|
| +
|
| +} // namespace
|
| +
|
| +ApmDataDumper::~ApmDataDumper() {
|
| +#if WEBRTC_AEC_DEBUG_DUMP == 1
|
| + for (auto& raw_files_element : raw_files_) {
|
| + fclose(raw_files_element.second);
|
| + }
|
| +
|
| + // Deleting the wav files implicitly causes the files to be closed.
|
| + for (auto& wav_files_element : wav_files_) {
|
| + delete wav_files_element.second;
|
| + }
|
| +#endif
|
| +}
|
| +
|
| +#if WEBRTC_AEC_DEBUG_DUMP == 1
|
| +FILE* ApmDataDumper::GetRawFile(const std::string& name) {
|
| + std::string filename =
|
| + FormFileName(name, instance_index_, recording_set_index_, ".dat");
|
| + auto search = raw_files_.find(filename);
|
| + if (search != raw_files_.end()) {
|
| + return search->second;
|
| + }
|
| + FILE* file = fopen(filename.c_str(), "wb");
|
| + raw_files_[filename] = file;
|
| + return file;
|
| +}
|
| +
|
| +WavWriter* ApmDataDumper::GetWavFile(const std::string& name,
|
| + int sample_rate_hz,
|
| + int num_channels) {
|
| + std::string filename =
|
| + FormFileName(name, instance_index_, recording_set_index_, ".wav");
|
| + auto search = wav_files_.find(filename);
|
| + if (search != wav_files_.end()) {
|
| + return search->second;
|
| + }
|
| + WavWriter* file =
|
| + new WavWriter(filename.c_str(), sample_rate_hz, num_channels);
|
| + wav_files_[filename] = file;
|
| + return file;
|
| +}
|
| +#endif
|
| +
|
| +} // namespace webrtc
|
|
|