diff options
Diffstat (limited to 'third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc')
-rw-r--r-- | third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc | 57 |
1 files changed, 57 insertions, 0 deletions
diff --git a/third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc b/third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc new file mode 100644 index 00000000..3a434714 --- /dev/null +++ b/third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc @@ -0,0 +1,57 @@ +/* + * Copyright (c) 2015 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/aec_logging_file_handling.h" + +#include <stdint.h> +#include <stdio.h> + +#include "webrtc/base/checks.h" +#include "webrtc/base/stringutils.h" +#include "webrtc/common_audio/wav_file.h" +#include "webrtc/typedefs.h" + +#ifdef WEBRTC_AEC_DEBUG_DUMP +void WebRtcAec_ReopenWav(const char* name, + int instance_index, + int process_rate, + int sample_rate, + rtc_WavWriter** wav_file) { + if (*wav_file) { + if (rtc_WavSampleRate(*wav_file) == sample_rate) + return; + rtc_WavClose(*wav_file); + } + char filename[64]; + int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.wav", name, + instance_index, process_rate); + + // Ensure there was no buffer output error. + RTC_DCHECK_GE(written, 0); + // Ensure that the buffer size was sufficient. + RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename)); + + *wav_file = rtc_WavOpen(filename, sample_rate, 1); +} + +void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file) { + char filename[64]; + int written = rtc::sprintfn(filename, sizeof(filename), "%s_%d.dat", name, + instance_index); + + // Ensure there was no buffer output error. + RTC_DCHECK_GE(written, 0); + // Ensure that the buffer size was sufficient. + RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename)); + + *file = fopen(filename, "wb"); +} + +#endif // WEBRTC_AEC_DEBUG_DUMP |