Arun Raghavan 753eada3aa Update audio_processing module
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
2015-10-15 16:18:45 +05:30

42 lines
1.3 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
#include <stdio.h>
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/typedefs.h"
#ifdef __cplusplus
extern "C" {
#endif
#ifdef WEBRTC_AEC_DEBUG_DUMP
// Opens a new Wav file for writing. If it was already open with a different
// sample frequency, it closes it first.
void WebRtcAec_ReopenWav(const char* name,
int instance_index,
int process_rate,
int sample_rate,
rtc_WavWriter** wav_file);
// Opens dumpfile with instance-specific filename.
void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file);
#endif // WEBRTC_AEC_DEBUG_DUMP
#ifdef __cplusplus
}
#endif
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_