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 <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：