Chromium Code Reviews| 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]; |
|
hlundin-webrtc
2016/04/25 11:29:59
I think the C++ way of doing this (in the absence
peah-webrtc
2016/04/25 11:46:40
Great! Thanks!! With that the code became much nic
|
| + 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 |