代码主要实现的功能是将一段aac的音频转换成wav
因为是新版的ffmpeg，所以使用了avcodec_decode_audio4解码音频，这样的话只能获取到一帧音频包，所以代码使用了 音频重采样swr_convert，将frame转换成buff
main.cpp
#include <stdio.h>
#include "audio.h"//这两个头文件在下面
#include "neaacdec.h"
extern "C"
{
#include <libavcodec\avcodec.h>
#include <libavformat\avformat.h>
#include <libswresample\swresample.h>
}
#pragma comment (lib, "avcodec.lib")
#pragma comment (lib, "avformat.lib")
#pragma comment (lib, "avutil.lib")
#pragma comment (lib, "swresample.lib")
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 19200
int main(int argc, char* argv[])
{
AVFormatContext *in_fctx = nullptr;
AVCodecContext *in_ast_cctx;
AVCodec *in_ast_codec;
AVPacket packet;
audio_file *aufile;
AVFrame *frame;
int out_size = 0;
int samples = 0;
int ret = 0;
int ast_idx = -1;
int i, first_time = 1;
char *filename = "F://Test.aac";//这个是文件输入的acc路径
char *wavfile = "F://Ouput.wav";//这个是输出wav的文件路径
ret = avformat_open_input(&in_fctx, filename, NULL, NULL);
if (ret != 0) {
printf("open input audio file[%s] fail", filename);
return -1;
}
ret = avformat_find_stream_info(in_fctx, NULL);
if (ret < 0) {
printf("find stream in audio file[%s] fail", filename);
return -1;
}
//这里我们假设，如果一个文件包含多个音频流，
//只对第一个音频流做转码，而对于视频流则忽略
for (i = 0; i<(int)in_fctx->nb_streams; ++i) {
if (in_fctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
ast_idx = i;
break;
}
}
if (ast_idx == -1) {
printf("there is not any audio stream in [%s]", filename);
return 0;
}
else {
printf("find audio stream in [%s]\n", filename);
}
in_ast_cctx = in_fctx->streams[ast_idx]->codec;
in_ast_codec = avcodec_find_decoder(in_ast_cctx->codec_id);
if (!in_ast_codec) {
printf("find decoder for codec_id[%d] fail, file[%s]", in_ast_cctx->codec_id, filename);
return -1;
}
ret = avcodec_open2(in_ast_cctx, in_ast_codec, NULL);
if (ret >= 0) {
printf("open codec[name:%s] for stream[idx:%d] of file[%s]\n", in_ast_codec->name, ast_idx, filename);
}
//如果需要谨慎的话，这些创建的都需要判断一下是否创建成功
SwrContext *swrCtx = swr_alloc();
//获取解码器
AVCodecContext *codecCtx = in_fctx->streams[ast_idx]->codec;
//输入的采样格式
enum AVSampleFormat in_sample_fmt = codecCtx->sample_fmt;
//输入采样率
int in_sample_rate = codecCtx->sample_rate;
//获取输入的声道布局
uint64_t in_ch_layout = codecCtx->channel_layout;
//输出采样格式16bit PCM
enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16;
//输出采样率
int out_sample_rate = 16000;
//输出的声道布局（立体声）
uint64_t out_ch_layout = AV_CH_LAYOUT_MONO;
swr_alloc_set_opts(swrCtx,
out_ch_layout, out_sample_fmt, out_sample_rate,
in_ch_layout, in_sample_fmt, in_sample_rate,
0, NULL);
//音频重采样初始化
swr_init(swrCtx);
frame = av_frame_alloc();
if (!frame)
return AVERROR(ENOMEM);
//这buff用于存储解码后，解析frame得到的buff
uint8_t *out_buffer = (uint8_t *)av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
if (!out_buffer)
return AVERROR(ENOMEM);
while (av_read_frame(in_fctx, &packet) >= 0) {
ret = avcodec_decode_audio4(in_ast_cctx, frame, &out_size, &packet);
if (first_time) {
aufile = open_audio_file(wavfile, in_ast_cctx->sample_rate, in_ast_cctx->channels, FAAD_FMT_16BIT, OUTPUT_WAV, 0);
first_time = 0;
}
swr_convert(swrCtx, &out_buffer, AVCODEC_MAX_AUDIO_FRAME_SIZE, (const uint8_t **)frame->data, frame->nb_samples);
samples = in_ast_cctx->frame_size * in_ast_cctx->channels;
write_audio_file(aufile, out_buffer, samples, 0);
av_free_packet(&packet);
}
if (!first_time)
close_audio_file(aufile);
swr_free(&swrCtx);
av_frame_free(&frame);
avcodec_close(in_ast_cctx);
avformat_close_input(&in_fctx);
av_free(out_buffer);
return 0;
}
audio.c
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: audio.c,v 1.29 2008/09/19 22:50:17 menno Exp $
**/
#ifdef _WIN32
#include <io.h>
#endif
#include <stdlib.h>
#include <stdio.h>
#include <fcntl.h>
#include <math.h>
#include "neaacdec.h"
#include "audio.h"
audio_file *open_audio_file(char *infile, int samplerate, int channels,
int outputFormat, int fileType, long channelMask)
{
audio_file *aufile = (audio_file*)malloc(sizeof(audio_file));
aufile->outputFormat = outputFormat;
aufile->samplerate = samplerate;
aufile->channels = channels;
aufile->total_samples = 0;
aufile->fileType = fileType;
aufile->channelMask = channelMask;
switch (outputFormat)
{
case FAAD_FMT_16BIT:
aufile->bits_per_sample = 16;
break;
case FAAD_FMT_24BIT:
aufile->bits_per_sample = 24;
break;
case FAAD_FMT_32BIT:
case FAAD_FMT_FLOAT:
aufile->bits_per_sample = 32;
break;
default:
if (aufile) free(aufile);
return NULL;
}
if(infile[0] == '-')
{
#ifdef _WIN32
setmode(fileno(stdout), O_BINARY);
#endif
aufile->sndfile = stdout;
aufile->toStdio = 1;
} else {
aufile->toStdio = 0;
aufile->sndfile = fopen(infile, "wb");
}
if (aufile->sndfile == NULL)
{
if (aufile) free(aufile);
return NULL;
}
if (aufile->fileType == OUTPUT_WAV)
{
if (aufile->channelMask)
write_wav_extensible_header(aufile, aufile->channelMask);
else
write_wav_header(aufile);
}
return aufile;
}
int write_audio_file(audio_file *aufile, void *sample_buffer, int samples, int offset)
{
char *buf = (char *)sample_buffer;
switch (aufile->outputFormat)
{
case FAAD_FMT_16BIT:
return write_audio_16bit(aufile, buf + offset*2, samples);
case FAAD_FMT_24BIT:
return write_audio_24bit(aufile, buf + offset*4, samples);
case FAAD_FMT_32BIT:
return write_audio_32bit(aufile, buf + offset*4, samples);
case FAAD_FMT_FLOAT:
return write_audio_float(aufile, buf + offset*4, samples);
default:
return 0;
}
return 0;
}
void close_audio_file(audio_file *aufile)
{
if ((aufile->fileType == OUTPUT_WAV) && (aufile->toStdio == 0))
{
fseek(aufile->sndfile, 0, SEEK_SET);
if (aufile->channelMask)
write_wav_extensible_header(aufile, aufile->channelMask);
else
write_wav_header(aufile);
}
if (aufile->toStdio == 0)
fclose(aufile->sndfile);
if (aufile) free(aufile);
}
static int write_wav_header(audio_file *aufile)
{
unsigned char header[44];
unsigned char* p = header;
unsigned int bytes = (aufile->bits_per_sample + 7) / 8;
float data_size = (float)bytes * aufile->total_samples;
unsigned long word32;
*p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F';
word32 = (data_size + (44 - 8) < (float)MAXWAVESIZE) ?
(unsigned long)data_size + (44 - 8) : (unsigned long)MAXWAVESIZE;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
*p++ = 'W'; *p++ = 'A'; *p++ = 'V'; *p++ = 'E';
*p++ = 'f'; *p++ = 'm'; *p++ = 't'; *p++ = ' ';
*p++ = 0x10; *p++ = 0x00; *p++ = 0x00; *p++ = 0x00;
if (aufile->outputFormat == FAAD_FMT_FLOAT)
{
*p++ = 0x03; *p++ = 0x00;
} else {
*p++ = 0x01; *p++ = 0x00;
}
*p++ = (unsigned char)(aufile->channels >> 0);
*p++ = (unsigned char)(aufile->channels >> 8);
word32 = (unsigned long)(aufile->samplerate + 0.5);
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
word32 = aufile->samplerate * bytes * aufile->channels;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
word32 = bytes * aufile->channels;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(aufile->bits_per_sample >> 0);
*p++ = (unsigned char)(aufile->bits_per_sample >> 8);
*p++ = 'd'; *p++ = 'a'; *p++ = 't'; *p++ = 'a';
word32 = data_size < MAXWAVESIZE ?
(unsigned long)data_size : (unsigned long)MAXWAVESIZE;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
return fwrite(header, sizeof(header), 1, aufile->sndfile);
}
static int write_wav_extensible_header(audio_file *aufile, long channelMask)
{
unsigned char header[68];
unsigned char* p = header;
unsigned int bytes = (aufile->bits_per_sample + 7) / 8;
float data_size = (float)bytes * aufile->total_samples;
unsigned long word32;
*p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F';
word32 = (data_size + (68 - 8) < (float)MAXWAVESIZE) ?
(unsigned long)data_size + (68 - 8) : (unsigned long)MAXWAVESIZE;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
*p++ = 'W'; *p++ = 'A'; *p++ = 'V'; *p++ = 'E';
*p++ = 'f'; *p++ = 'm'; *p++ = 't'; *p++ = ' ';
*p++ = /*0x10*/0x28; *p++ = 0x00; *p++ = 0x00; *p++ = 0x00;
/* WAVE_FORMAT_EXTENSIBLE */
*p++ = 0xFE; *p++ = 0xFF;
*p++ = (unsigned char)(aufile->channels >> 0);
*p++ = (unsigned char)(aufile->channels >> 8);
word32 = (unsigned long)(aufile->samplerate + 0.5);
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
word32 = aufile->samplerate * bytes * aufile->channels;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
word32 = bytes * aufile->channels;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(aufile->bits_per_sample >> 0);
*p++ = (unsigned char)(aufile->bits_per_sample >> 8);
/* cbSize */
*p++ = (unsigned char)(22);
*p++ = (unsigned char)(0);
/* WAVEFORMATEXTENSIBLE */
/* wValidBitsPerSample */
*p++ = (unsigned char)(aufile->bits_per_sample >> 0);
*p++ = (unsigned char)(aufile->bits_per_sample >> 8);
/* dwChannelMask */
word32 = channelMask;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
/* SubFormat */
if (aufile->outputFormat == FAAD_FMT_FLOAT)
{
/* KSDATAFORMAT_SUBTYPE_IEEE_FLOAT: 00000003-0000-0010-8000-00aa00389b71 */
*p++ = 0x03;
*p++ = 0x00;
*p++ = 0x00;
*p++ = 0x00;
*p++ = 0x00; *p++ = 0x00; *p++ = 0x10; *p++ = 0x00; *p++ = 0x80; *p++ = 0x00;
*p++ = 0x00; *p++ = 0xaa; *p++ = 0x00; *p++ = 0x38; *p++ = 0x9b; *p++ = 0x71;
} else {
/* KSDATAFORMAT_SUBTYPE_PCM: 00000001-0000-0010-8000-00aa00389b71 */
*p++ = 0x01;
*p++ = 0x00;
*p++ = 0x00;
*p++ = 0x00;
*p++ = 0x00; *p++ = 0x00; *p++ = 0x10; *p++ = 0x00; *p++ = 0x80; *p++ = 0x00;
*p++ = 0x00; *p++ = 0xaa; *p++ = 0x00; *p++ = 0x38; *p++ = 0x9b; *p++ = 0x71;
}
/* end WAVEFORMATEXTENSIBLE */
*p++ = 'd'; *p++ = 'a'; *p++ = 't'; *p++ = 'a';
word32 = data_size < MAXWAVESIZE ?
(unsigned long)data_size : (unsigned long)MAXWAVESIZE;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
return fwrite(header, sizeof(header), 1, aufile->sndfile);
}
static int write_audio_16bit(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
short *sample_buffer16 = (short*)sample_buffer;
char *data = (char*)malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
if (aufile->channels == 6 && aufile->channelMask)
{
for (i = 0; i < samples; i += aufile->channels)
{
short r1, r2, r3, r4, r5, r6;
r1 = sample_buffer16[i];
r2 = sample_buffer16[i+1];
r3 = sample_buffer16[i+2];
r4 = sample_buffer16[i+3];
r5 = sample_buffer16[i+4];
r6 = sample_buffer16[i+5];
sample_buffer16[i] = r2;
sample_buffer16[i+1] = r3;
sample_buffer16[i+2] = r1;
sample_buffer16[i+3] = r6;
sample_buffer16[i+4] = r4;
sample_buffer16[i+5] = r5;
}
}
for (i = 0; i < samples; i++)
{
data[i*2] = (char)(sample_buffer16[i] & 0xFF);
data[i*2+1] = (char)((sample_buffer16[i] >> 8) & 0xFF);
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
static int write_audio_24bit(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
long *sample_buffer24 = (long*)sample_buffer;
char *data = (char*)malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
if (aufile->channels == 6 && aufile->channelMask)
{
for (i = 0; i < samples; i += aufile->channels)
{
long r1, r2, r3, r4, r5, r6;
r1 = sample_buffer24[i];
r2 = sample_buffer24[i+1];
r3 = sample_buffer24[i+2];
r4 = sample_buffer24[i+3];
r5 = sample_buffer24[i+4];
r6 = sample_buffer24[i+5];
sample_buffer24[i] = r2;
sample_buffer24[i+1] = r3;
sample_buffer24[i+2] = r1;
sample_buffer24[i+3] = r6;
sample_buffer24[i+4] = r4;
sample_buffer24[i+5] = r5;
}
}
for (i = 0; i < samples; i++)
{
data[i*3] = (char)(sample_buffer24[i] & 0xFF);
data[i*3+1] = (char)((sample_buffer24[i] >> 8) & 0xFF);
data[i*3+2] = (char)((sample_buffer24[i] >> 16) & 0xFF);
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
static int write_audio_32bit(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
long *sample_buffer32 = (long*)sample_buffer;
char *data = (char*)malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
if (aufile->channels == 6 && aufile->channelMask)
{
for (i = 0; i < samples; i += aufile->channels)
{
long r1, r2, r3, r4, r5, r6;
r1 = sample_buffer32[i];
r2 = sample_buffer32[i+1];
r3 = sample_buffer32[i+2];
r4 = sample_buffer32[i+3];
r5 = sample_buffer32[i+4];
r6 = sample_buffer32[i+5];
sample_buffer32[i] = r2;
sample_buffer32[i+1] = r3;
sample_buffer32[i+2] = r1;
sample_buffer32[i+3] = r6;
sample_buffer32[i+4] = r4;
sample_buffer32[i+5] = r5;
}
}
for (i = 0; i < samples; i++)
{
data[i*4] = (char)(sample_buffer32[i] & 0xFF);
data[i*4+1] = (char)((sample_buffer32[i] >> 8) & 0xFF);
data[i*4+2] = (char)((sample_buffer32[i] >> 16) & 0xFF);
data[i*4+3] = (char)((sample_buffer32[i] >> 24) & 0xFF);
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
static int write_audio_float(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
float *sample_buffer_f = (float*)sample_buffer;
unsigned char *data = (unsigned char *)malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
if (aufile->channels == 6 && aufile->channelMask)
{
for (i = 0; i < samples; i += aufile->channels)
{
float r1, r2, r3, r4, r5, r6;
r1 = sample_buffer_f[i];
r2 = sample_buffer_f[i+1];
r3 = sample_buffer_f[i+2];
r4 = sample_buffer_f[i+3];
r5 = sample_buffer_f[i+4];
r6 = sample_buffer_f[i+5];
sample_buffer_f[i] = r2;
sample_buffer_f[i+1] = r3;
sample_buffer_f[i+2] = r1;
sample_buffer_f[i+3] = r6;
sample_buffer_f[i+4] = r4;
sample_buffer_f[i+5] = r5;
}
}
for (i = 0; i < samples; i++)
{
int exponent, mantissa, negative = 0 ;
float in = sample_buffer_f[i];
data[i*4] = 0; data[i*4+1] = 0; data[i*4+2] = 0; data[i*4+3] = 0;
if (in == 0.0)
continue;
if (in < 0.0)
{
in *= -1.0;
negative = 1;
}
in = (float)frexp(in, &exponent);
exponent += 126;
in *= (float)0x1000000;
mantissa = (((int)in) & 0x7FFFFF);
if (negative)
data[i*4+3] |= 0x80;
if (exponent & 0x01)
data[i*4+2] |= 0x80;
data[i*4] = mantissa & 0xFF;
data[i*4+1] = (mantissa >> 8) & 0xFF;
data[i*4+2] |= (mantissa >> 16) & 0x7F;
data[i*4+3] |= (exponent >> 1) & 0x7F;
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
audio.h
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: audio.h,v 1.19 2007/11/01 12:33:29 menno Exp $
**/
#ifndef AUDIO_H_INCLUDED
#define AUDIO_H_INCLUDED
#ifdef __cplusplus
extern "C" {
#endif
#define MAXWAVESIZE 4294967040LU
#define OUTPUT_WAV 1
#define OUTPUT_RAW 2
typedef struct
{
int toStdio;
int outputFormat;
FILE *sndfile;
unsigned int fileType;
unsigned long samplerate;
unsigned int bits_per_sample;
unsigned int channels;
unsigned long total_samples;
long channelMask;
} audio_file;
audio_file *open_audio_file(char *infile, int samplerate, int channels,
int outputFormat, int fileType, long channelMask);
int write_audio_file(audio_file *aufile, void *sample_buffer, int samples, int offset);
void close_audio_file(audio_file *aufile);
static int write_wav_header(audio_file *aufile);
static int write_wav_extensible_header(audio_file *aufile, long channelMask);
static int write_audio_16bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
static int write_audio_24bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
static int write_audio_32bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
static int write_audio_float(audio_file *aufile, void *sample_buffer,
unsigned int samples);
#ifdef __cplusplus
}
#endif
#endif
neaacdec.h
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: neaacdec.h,v 1.13 2009/01/26 23:51:15 menno Exp $
**/
#ifndef __NEAACDEC_H__
#define __NEAACDEC_H__
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#if 1
/* MACROS FOR BACKWARDS COMPATIBILITY */
/* structs */
#define faacDecHandle NeAACDecHandle
#define faacDecConfiguration NeAACDecConfiguration
#define faacDecConfigurationPtr NeAACDecConfigurationPtr
#define faacDecFrameInfo NeAACDecFrameInfo
/* functions */
#define faacDecGetErrorMessage NeAACDecGetErrorMessage
#define faacDecSetConfiguration NeAACDecSetConfiguration
#define faacDecGetCurrentConfiguration NeAACDecGetCurrentConfiguration
#define faacDecInit NeAACDecInit
#define faacDecInit2 NeAACDecInit2
#define faacDecInitDRM NeAACDecInitDRM
#define faacDecPostSeekReset NeAACDecPostSeekReset
#define faacDecOpen NeAACDecOpen
#define faacDecClose NeAACDecClose
#define faacDecDecode NeAACDecDecode
#define AudioSpecificConfig NeAACDecAudioSpecificConfig
#endif
#ifdef _WIN32
#pragma pack(push, 8)
#ifndef NEAACDECAPI
#define NEAACDECAPI __cdecl
#endif
#else
#ifndef NEAACDECAPI
#define NEAACDECAPI
#endif
#endif
#define FAAD2_VERSION "2.7"
/* object types for AAC */
#define MAIN 1
#define LC 2
#define SSR 3
#define LTP 4
#define HE_AAC 5
#define ER_LC 17
#define ER_LTP 19
#define LD 23
#define DRM_ER_LC 27 /* special object type for DRM */
/* header types */
#define RAW 0
#define ADIF 1
#define ADTS 2
#define LATM 3
/* SBR signalling */
#define NO_SBR 0
#define SBR_UPSAMPLED 1
#define SBR_DOWNSAMPLED 2
#define NO_SBR_UPSAMPLED 3
/* library output formats */
#define FAAD_FMT_16BIT 1
#define FAAD_FMT_24BIT 2
#define FAAD_FMT_32BIT 3
#define FAAD_FMT_FLOAT 4
#define FAAD_FMT_FIXED FAAD_FMT_FLOAT
#define FAAD_FMT_DOUBLE 5
/* Capabilities */
#define LC_DEC_CAP (1<<0) /* Can decode LC */
#define MAIN_DEC_CAP (1<<1) /* Can decode MAIN */
#define LTP_DEC_CAP (1<<2) /* Can decode LTP */
#define LD_DEC_CAP (1<<3) /* Can decode LD */
#define ERROR_RESILIENCE_CAP (1<<4) /* Can decode ER */
#define FIXED_POINT_CAP (1<<5) /* Fixed point */
/* Channel definitions */
#define FRONT_CHANNEL_CENTER (1)
#define FRONT_CHANNEL_LEFT (2)
#define FRONT_CHANNEL_RIGHT (3)
#define SIDE_CHANNEL_LEFT (4)
#define SIDE_CHANNEL_RIGHT (5)
#define BACK_CHANNEL_LEFT (6)
#define BACK_CHANNEL_RIGHT (7)
#define BACK_CHANNEL_CENTER (8)
#define LFE_CHANNEL (9)
#define UNKNOWN_CHANNEL (0)
/* DRM channel definitions */
#define DRMCH_MONO 1
#define DRMCH_STEREO 2
#define DRMCH_SBR_MONO 3
#define DRMCH_SBR_STEREO 4
#define DRMCH_SBR_PS_STEREO 5
/* A decode call can eat up to FAAD_MIN_STREAMSIZE bytes per decoded channel,
so at least so much bytes per channel should be available in this stream */
#define FAAD_MIN_STREAMSIZE 768 /* 6144 bits/channel */
typedef void *NeAACDecHandle;
typedef struct mp4AudioSpecificConfig
{
/* Audio Specific Info */
unsigned char objectTypeIndex;
unsigned char samplingFrequencyIndex;
unsigned long samplingFrequency;
unsigned char channelsConfiguration;
/* GA Specific Info */
unsigned char frameLengthFlag;
unsigned char dependsOnCoreCoder;
unsigned short coreCoderDelay;
unsigned char extensionFlag;
unsigned char aacSectionDataResilienceFlag;
unsigned char aacScalefactorDataResilienceFlag;
unsigned char aacSpectralDataResilienceFlag;
unsigned char epConfig;
char sbr_present_flag;
char forceUpSampling;
char downSampledSBR;
} mp4AudioSpecificConfig;
typedef struct NeAACDecConfiguration
{
unsigned char defObjectType;
unsigned long defSampleRate;
unsigned char outputFormat;
unsigned char downMatrix;
unsigned char useOldADTSFormat;
unsigned char dontUpSampleImplicitSBR;
} NeAACDecConfiguration, *NeAACDecConfigurationPtr;
typedef struct NeAACDecFrameInfo
{
unsigned long bytesconsumed;
unsigned long samples;
unsigned char channels;
unsigned char error;
unsigned long samplerate;
/* SBR: 0: off, 1: on; upsample, 2: on; downsampled, 3: off; upsampled */
unsigned char sbr;
/* MPEG-4 ObjectType */
unsigned char object_type;
/* AAC header type; MP4 will be signalled as RAW also */
unsigned char header_type;
/* multichannel configuration */
unsigned char num_front_channels;
unsigned char num_side_channels;
unsigned char num_back_channels;
unsigned char num_lfe_channels;
unsigned char channel_position[64];
/* PS: 0: off, 1: on */
unsigned char ps;
} NeAACDecFrameInfo;
char* NEAACDECAPI NeAACDecGetErrorMessage(unsigned char errcode);
unsigned long NEAACDECAPI NeAACDecGetCapabilities(void);
NeAACDecHandle NEAACDECAPI NeAACDecOpen(void);
NeAACDecConfigurationPtr NEAACDECAPI NeAACDecGetCurrentConfiguration(NeAACDecHandle hDecoder);
unsigned char NEAACDECAPI NeAACDecSetConfiguration(NeAACDecHandle hDecoder,
NeAACDecConfigurationPtr config);
/* Init the library based on info from the AAC file (ADTS/ADIF) */
long NEAACDECAPI NeAACDecInit(NeAACDecHandle hDecoder,
unsigned char *buffer,
unsigned long buffer_size,
unsigned long *samplerate,
unsigned char *channels);
/* Init the library using a DecoderSpecificInfo */
char NEAACDECAPI NeAACDecInit2(NeAACDecHandle hDecoder,
unsigned char *pBuffer,
unsigned long SizeOfDecoderSpecificInfo,
unsigned long *samplerate,
unsigned char *channels);
/* Init the library for DRM */
char NEAACDECAPI NeAACDecInitDRM(NeAACDecHandle *hDecoder, unsigned long samplerate,
unsigned char channels);
void NEAACDECAPI NeAACDecPostSeekReset(NeAACDecHandle hDecoder, long frame);
void NEAACDECAPI NeAACDecClose(NeAACDecHandle hDecoder);
void* NEAACDECAPI NeAACDecDecode(NeAACDecHandle hDecoder,
NeAACDecFrameInfo *hInfo,
unsigned char *buffer,
unsigned long buffer_size);
void* NEAACDECAPI NeAACDecDecode2(NeAACDecHandle hDecoder,
NeAACDecFrameInfo *hInfo,
unsigned char *buffer,
unsigned long buffer_size,
void **sample_buffer,
unsigned long sample_buffer_size);
char NEAACDECAPI NeAACDecAudioSpecificConfig(unsigned char *pBuffer,
unsigned long buffer_size,
mp4AudioSpecificConfig *mp4ASC);
#ifdef _WIN32
#pragma pack(pop)
#endif
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif
如果懒得找ffmpeg库资源的话，这里有一个编好的资源,代码都一样的就是多了ffmpeg资源，能编译
使用VS2015（x64）可以直接编译通过，ffmpeg 4.3 版本
资源连接：https://download.csdn.net/download/qq_36351159/86890587
因下载的代码使用的ffmpeg版本是旧的，导致新版的不能使用，调整一下代码，确认能正常使用，便于需要使用的人参考(主要是没有CSND的下载B)，使用的音频解码接口是avcodec_decode_audio2.这样的话不需要音频重采样，可以获取一个buff
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