AAC音频格式分析

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

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：

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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;  
}