AAC音頻格式分析

關於AAC音頻格式基本狀況，可參考維基百科http://en.wikipedia.org/wiki/Advanced_Audio_Coding

 

AAC音頻格式分析

AAC音頻格式有ADIF和ADTS： 

ADIF：Audio Data Interchange Format 音頻數據交換格式。這種格式的特徵是能夠肯定的找到這個音頻數據的開始，不需進行在音頻數據流中間開始的解碼，即它的解碼必須在明肯定義的開始處進行。故這種格式經常使用在磁盤文件中。 

ADTS：Audio Data Transport Stream 音頻數據傳輸流。這種格式的特徵是它是一個有同步字的比特流，解碼能夠在這個流中任何位置開始。它的特徵相似於mp3數據流格式。 

簡單說，ADTS能夠在任意幀解碼，也就是說它每一幀都有頭信息。ADIF只有一個統一的頭，因此必須獲得全部的數據後解碼。且這兩種的header的格式也是不一樣的，目前通常編碼後的和抽取出的都是ADTS格式的音頻流。 

語音系統對實時性要求較高，基本是這樣一個流程，採集音頻數據，本地編碼，數據上傳，服務器處理，數據下發，本地解碼 

ADTS是幀序列，自己具有流特徵，在音頻流的傳輸與處理方面更加合適。 

  

 

ADTS幀結構：

 

header

body

 

ADTS幀首部結構：

序號	域	長度（bits）	說明
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解碼

在解碼方面，使用了開源的FAAD，http://www.audiocoding.com/faad2.html

sdk解壓縮後，docs目錄有詳細的api說明文檔，主要用到的有如下幾個：

NeAACDecHandle NEAACAPI NeAACDecOpen(void);

建立解碼環境並返回一個句柄

void NEAACAPI NeAACDecClose(NeAACDecHandle hDecoder);

關閉解碼環境

NeAACDecConfigurationPtr NEAACAPI NeAACDecGetCurrentConfiguration(NeAACDecHandle hDecoder);

獲取當前解碼器庫的配置

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

爲解碼器庫設置一個配置結構

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

初始化解碼器庫

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

解碼AAC數據

 

 

對以上api作了簡單封裝，寫了一個解碼類，涵蓋了FAAD庫的基本用法，感興趣的朋友能夠看看

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;

    }

}

From http://www.cnblogs.com/caosiyang/