| /* |
| * 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 |