AAC音频格式详解

关于AAC音频格式基本情况，可参考维基百科http://en.wikipedia.org/wiki/Advanced_Audio_Coding

AAC音频格式分析

AAC音频格式有ADIF和ADTS：

ADIF：Audio Data Interchange Format 音频数据交换格式。这种格式的特征是可以确定的找到这个音频数据的开始，不需进行在音频数据流中间开始的解码，即它的解码必须在明确定义的开始处进行。故这种格式常用在磁盘文件中。

ADTS：Audio Data Transport Stream 音频数据传输流。这种格式的特征是它是一个有同步字的比特流，解码可以在这个流中任何位置开始。它的特征类似于mp3数据流格式。

简单说，ADTS可以在任意帧解码，也就是说它每一帧都有头信息。ADIF只有一个统一的头，所以必须得到所有的数据后解码。且这两种的header的格式也是不同的，目前一般编码后的和抽取出的都是ADTS格式的音频流。

语音系统对实时性要求较高，基本是这样一个流程，采集音频数据，本地编码，数据上传，服务器处理，数据下发，本地解码

ADTS是帧序列，本身具备流特征，在音频流的传输与处理方面更加合适。

ADTS帧结构：

header

body

ADTS帧首部结构：

序号	域	长度（bits）	说明
1	Syncword	12	all bits must be 1
2	MPEG version	1	0 for MPEG-4, 1 for MPEG-2
3	Layer	2	always 0
4	Protection Absent	1	et to 1 if there is no CRC and 0 if there is CRC
5	Profile	2	the MPEG-4 Audio Object Type minus 1
6	MPEG-4 Sampling Frequency Index	4	MPEG-4 Sampling Frequency Index (15 is forbidden)
7	Private Stream	1	set to 0 when encoding, ignore when decoding
8	MPEG-4 Channel Configuration	3	MPEG-4 Channel Configuration (in the case of 0, the channel configuration is sent via an inband PCE)
9	Originality	1	set to 0 when encoding, ignore when decoding
10	Home	1	set to 0 when encoding, ignore when decoding
11	Copyrighted Stream	1	set to 0 when encoding, ignore when decoding
12	Copyrighted Start	1	set to 0 when encoding, ignore when decoding
13	Frame Length	13	this value must include 7 or 9 bytes of header length: FrameLength = (ProtectionAbsent == 1 ? 7 : 9) + size(AACFrame)
14	Buffer Fullness	11	buffer fullness
15	Number of AAC Frames	2	number of AAC frames (RDBs) in ADTS frame minus 1, for maximum compatibility always use 1 AAC frame per ADTS frame
16	CRC	16	CRC if protection absent is 0

AAC解码

在解码方面，使用了开源的FAAD，http://www.audiocoding.com/faad2.html

sdk解压缩后，docs目录有详细的api说明文档，主要用到的有以下几个：

NeAACDecHandle NEAACAPI NeAACDecOpen(void);

创建解码环境并返回一个句柄

void NEAACAPI NeAACDecClose(NeAACDecHandle hDecoder);

关闭解码环境

NeAACDecConfigurationPtr NEAACAPI NeAACDecGetCurrentConfiguration(NeAACDecHandle hDecoder);

获取当前解码器库的配置

unsigned char NEAACAPI NeAACDecSetConfiguration(NeAACDecHandle hDecoder, NeAACDecConfigurationPtr config);

为解码器库设置一个配置结构

long NEAACAPI NeAACDecInit(NeAACDecHandle hDecoder, unsigned char *buffer, unsigned long buffer_size, unsigned long *samplerate, unsigned char *channels);

初始化解码器库

void* NEAACAPI NeAACDecDecode(NeAACDecHandle hDecoder, NeAACDecFrameInfo *hInfo, unsigned char *buffer, unsigned long buffer_size);

解码AAC数据

对以上api做了简单封装，写了一个解码类，涵盖了FAAD库的基本用法，感兴趣的朋友可以看看

MyAACDecoder.h：

/**

 * filename: MyAACDecoder.h

 * summary: convert aac to wave

 * author: caosiyang

 * email: csy3228@gmail.com

*/

#ifndef __MYAACDECODER_H__

#define __MYAACDECODER_H__

#include "Buffer.h"

#include "mytools.h"

#include "WaveFormat.h"

#include "faad.h"

#include <iostream>

using namespace std;

class MyAACDecoder {

public:

    MyAACDecoder();

    ~MyAACDecoder();

    int32_t Decode(char *aacbuf, uint32_t aacbuflen);

    const char* WavBodyData() const {

        return _mybuffer.Data();

    uint32_t WavBodyLength() const {

        return _mybuffer.Length();

    const char* WavHeaderData() const {

        return _wave_format.getHeaderData();

    uint32_t WavHeaderLength() const {

        return _wave_format.getHeaderLength();

private:

    MyAACDecoder(const MyAACDecoder &dec);

    MyAACDecoder& operator=(const MyAACDecoder &rhs);

    //init AAC decoder

    int32_t _init_aac_decoder(char *aacbuf, int32_t aacbuflen);

    //destroy aac decoder

    void _destroy_aac_decoder();

    //parse AAC ADTS header, get frame length

    uint32_t _get_frame_length(const char *aac_header) const;

    //AAC decoder properties

    NeAACDecHandle _handle;

    unsigned long _samplerate;

    unsigned char _channel;

    Buffer _mybuffer;

    WaveFormat _wave_format;

};

#endif /*__MYAACDECODER_H__*/

MyAACDecoder.cpp：

#include "MyAACDecoder.h"

MyAACDecoder::MyAACDecoder(): _handle(NULL), _samplerate(44100), _channel(2), _mybuffer(4096, 4096) {

MyAACDecoder::~MyAACDecoder() {

    _destroy_aac_decoder();

int32_t MyAACDecoder::Decode(char *aacbuf, uint32_t aacbuflen) {

    int32_t res = 0;

    if (!_handle) {

        if (_init_aac_decoder(aacbuf, aacbuflen) != 0) {

            ERR1(":::: init aac decoder failed ::::");

            return -1;

    //clean _mybuffer

    _mybuffer.Clean();

    uint32_t donelen = 0;

    uint32_t wav_data_len = 0;

    while (donelen < aacbuflen) {

        uint32_t framelen = _get_frame_length(aacbuf + donelen);

        if (donelen + framelen > aacbuflen) {

            break;

        //decode

        NeAACDecFrameInfo info;

        void *buf = NeAACDecDecode(_handle, &info, (unsigned char*)aacbuf + donelen, framelen);

        if (buf && info.error == 0) {

            if (info.samplerate == 44100) {

                //44100Hz

                //src: 2048 samples, 4096 bytes

                //dst: 2048 samples, 4096 bytes

                uint32_t tmplen = info.samples * 16 / 8;

                _mybuffer.Fill((const char*)buf, tmplen);

                wav_data_len += tmplen;

            } else if (info.samplerate == 22050) {

                //22050Hz

                //src: 1024 samples, 2048 bytes

                //dst: 2048 samples, 4096 bytes

                short *ori = (short*)buf;

                short tmpbuf[info.samples * 2];

                uint32_t tmplen = info.samples * 16 / 8 * 2;

                for (int32_t i = 0, j = 0; i < info.samples; i += 2) {

                    tmpbuf[j++] = ori[i];

                    tmpbuf[j++] = ori[i + 1];

                    tmpbuf[j++] = ori[i];

                    tmpbuf[j++] = ori[i + 1];

                _mybuffer.Fill((const char*)tmpbuf, tmplen);

                wav_data_len += tmplen;

        } else {

            ERR1("NeAACDecDecode() failed");

        donelen += framelen;

    //generate Wave header

    _wave_format.setSampleRate(_samplerate);

    _wave_format.setChannel(_channel);

    _wave_format.setSampleBit(16);

    _wave_format.setBandWidth(_samplerate * 16 * _channel / 8);

    _wave_format.setDataLength(wav_data_len);

    _wave_format.setTotalLength(wav_data_len + 44);

    _wave_format.GenerateHeader();

    return 0;

uint32_t MyAACDecoder::_get_frame_length(const char *aac_header) const {

    uint32_t len = *(uint32_t *)(aac_header + 3);

    len = ntohl(len); //Little Endian

    len = len << 6;

    len = len >> 19;

    return len;

int32_t MyAACDecoder::_init_aac_decoder(char* aacbuf, int32_t aacbuflen) {

    unsigned long cap = NeAACDecGetCapabilities();

    _handle = NeAACDecOpen();

    if (!_handle) {

        ERR1("NeAACDecOpen() failed");

        _destroy_aac_decoder();

        return -1;

    NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(_handle);

    if (!conf) {

        ERR1("NeAACDecGetCurrentConfiguration() failed");

        _destroy_aac_decoder();

        return -1;

    NeAACDecSetConfiguration(_handle, conf);

    long res = NeAACDecInit(_handle, (unsigned char *)aacbuf, aacbuflen, &_samplerate, &_channel);

    if (res < 0) {

        ERR1("NeAACDecInit() failed");

        _destroy_aac_decoder();

        return -1;

    //fprintf(stdout, "SampleRate = %d\n", _samplerate);

    //fprintf(stdout, "Channel    = %d\n", _channel);

    //fprintf(stdout, ":::: init aac decoder done ::::\n");

    return 0;

void MyAACDecoder::_destroy_aac_decoder() {

    if (_handle) {

        NeAACDecClose(_handle);

        _handle = NULL;

1.ADTS是个啥

ADTS全称是(Audio Data Transport Stream)，是AAC的一种十分常见的传输格式。

记得第一次做demux的时候，把AAC音频的ES流从FLV封装格式中抽出来送给硬件解码器时，不能播;保存到本地用pc的播放器播时，我靠也不能播。当时崩溃了，后来通过查找资料才知道。一般的AAC解码器都需要把AAC的ES流打包成ADTS的格式，一般是在AAC ES流前添加7个字节的ADTS header。也就是说你可以吧ADTS这个头看作是AAC的frameheader。

ADTS AAC
ADTS_header	AAC ES	ADTS_header	AAC ES	...	ADTS_header	AAC ES

2.ADTS内容及结构

ADTS 头中相对有用的信息 采样率、声道数、帧长度。想想也是，我要是解码器的话，你给我一堆得AAC音频ES流我也解不出来。每一个带ADTS头信息的AAC流会清晰的告送解码器他需要的这些信息。

一般情况下ADTS的头信息都是7个字节，分为2部分：

adts_fixed_header();

adts_variable_header();

syncword ：同步头总是0xFFF, all bits must be 1，代表着一个ADTS帧的开始

ID：MPEG Version: 0 for MPEG-4, 1 for MPEG-2

Layer：always: '00'

profile：表示使用哪个级别的AAC，有些芯片只支持AAC LC 。在MPEG-2 AAC中定义了3种：

sampling_frequency_index：表示使用的采样率下标，通过这个下标在 Sampling Frequencies[ ]数组中查找得知采样率的值。

There are 13 supported frequencies:

0: 96000 Hz
1: 88200 Hz
2: 64000 Hz
3: 48000 Hz
4: 44100 Hz
5: 32000 Hz
6: 24000 Hz
7: 22050 Hz
8: 16000 Hz
9: 12000 Hz
10: 11025 Hz
11: 8000 Hz
12: 7350 Hz
13: Reserved
14: Reserved
15: frequency is written explictly

channel_configuration: 表示声道数

0: Defined in AOT Specifc Config
1: 1 channel: front-center
2: 2 channels: front-left, front-right
3: 3 channels: front-center, front-left, front-right
4: 4 channels: front-center, front-left, front-right, back-center
5: 5 channels: front-center, front-left, front-right, back-left, back-right
6: 6 channels: front-center, front-left, front-right, back-left, back-right, LFE-channel
7: 8 channels: front-center, front-left, front-right, side-left, side-right, back-left, back-right, LFE-channel
8-15: Reserved

frame_length : 一个ADTS帧的长度包括ADTS头和AAC原始流.

adts_buffer_fullness：0x7FF 说明是码率可变的码流

3.将AAC打包成ADTS格式

如果是通过嵌入式高清解码芯片做产品的话，一般情况的解码工作都是由硬件来完成的。所以大部分的工作是把AAC原始流打包成ADTS的格式，然后丢给硬件就行了。

通过对ADTS格式的了解，很容易就能把AAC打包成ADTS。我们只需得到封装格式里面关于音频采样率、声道数、元数据长度、aac格式类型等信息。然后在每个AAC原始流前面加上个ADTS头就OK了。

贴上ffmpeg中添加ADTS头的代码，就可以很清晰的了解ADTS头的结构：

int ff_adts_write_frame_header(ADTSContext *ctx,
uint8_t *buf, int size, int pce_size)
{
PutBitContext pb;
init_put_bits(&pb, buf, ADTS_HEADER_SIZE);
/* adts_fixed_header */
put_bits(&pb, 12, 0xfff); /* syncword */
put_bits(&pb, 1, 0); /* ID */
put_bits(&pb, 2, 0); /* layer */
put_bits(&pb, 1, 1); /* protection_absent */
put_bits(&pb, 2, ctx->objecttype); /* profile_objecttype */
put_bits(&pb, 4, ctx->sample_rate_index);
put_bits(&pb, 1, 0); /* private_bit */
put_bits(&pb, 3, ctx->channel_conf); /* channel_configuration */
put_bits(&pb, 1, 0); /* original_copy */
put_bits(&pb, 1, 0); /* home */
/* adts_variable_header */
put_bits(&pb, 1, 0); /* copyright_identification_bit */
put_bits(&pb, 1, 0); /* copyright_identification_start */
put_bits(&pb, 13, ADTS_HEADER_SIZE + size + pce_size); /* aac_frame_length */
put_bits(&pb, 11, 0x7ff); /* adts_buffer_fullness */
put_bits(&pb, 2, 0); /* number_of_raw_data_blocks_in_frame */
flush_put_bits(&pb);
return 0;
}

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