753eada3aa
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
58 lines
1.9 KiB
C++
58 lines
1.9 KiB
C++
/*
|
|
* 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
|