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/*
* 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
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