diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc | 65 |
1 files changed, 65 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc new file mode 100644 index 00000000000..491196e0972 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc @@ -0,0 +1,65 @@ +/* + * 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 <sstream> + +#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 char* name, + int instance_index, + int reinit_index, + const std::string& suffix) { + std::stringstream ss; + ss << name << "_" << instance_index << "-" << reinit_index << suffix; + return ss.str(); +} +#endif + +} // namespace + +#if WEBRTC_AEC_DEBUG_DUMP == 1 +FILE* ApmDataDumper::GetRawFile(const char* name) { + std::string filename = + FormFileName(name, instance_index_, recording_set_index_, ".dat"); + auto& f = raw_files_[filename]; + if (!f) { + f.reset(fopen(filename.c_str(), "wb")); + } + return f.get(); +} + +WavWriter* ApmDataDumper::GetWavFile(const char* name, + int sample_rate_hz, + int num_channels) { + std::string filename = + FormFileName(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)); + } + return f.get(); +} + +#endif + +} // namespace webrtc |