一SADTS格式：

???????ADTS的全稱是Audio Data Transport Stream。是AAC音頻的傳輸流格式。

???????AAC音頻格式在MPEG-2（ISO-13318-7 2003）中有定義。AAC后來又被采用到MPEG-4標準中。

???????1. adts_sequence()

???????{

??????????????while (nextbits() == syncword)

??????????????{

?????????????????????adts_frame();

??????????????}

???????}

?

???????2. adts_frame()

???????{

??????????????adts_fixed_header();

??????????????adts_variable_header();

??????????????if (number_of_raw_data_blocks_in_frame == 0)

??????????????{

?????????????????????adts_error_check();

?????????????????????raw_data_block();

??????????????} else

??????????????{

?????????????????????adts_header_error_check();

?????????????????????for (i = 0; i <= number_of_raw_data_blocks_in_frame; i++)

?????????????????????{

?????????????????????????????raw_data_block();

?????????????????????????????adts_raw_data_block_error_check();

?????????????????????}

??????????????}

???????}

?

???????3. adts_fixed_header()

???????{

??????????????syncword: 12 bslbf

??????????????ID: 1 bslbf

??????????????layer: 2 uimsbf

??????????????protection_absent: 1 bslbf

??????????????profile: 2 uimsbf

??????????????sampling_frequency_index: 4 uimsbf

??????????????private_bit: 1 bslbf

??????????????channel_configuration: 3 uimsbf

??????????????original/copy: 1 bslbf

??????????????home: 1 bslbf

???????}

??????

???????adts_variable_header()

???????{

??????????????copyright_identification_bit: 1 bslbf

??????????????copyright_identification_start: 1 bslbf

??????????????frame_length: 13 bslbf

??????????????adts_buffer_fullness: 11 bslbf

??????????????number_of_raw_data_blocks_in_frame: 2 uimsfb

???????}

?

???????詳細說明下ADTS頭的重要數據部分：

???????syncword?同步字The bit string?‘1111 1111 1111’，說明一個ADTS幀的開始。

???????ID????MPEG?標示符,?設置為1.

???????layer Indicates which layer is used. Set to ‘00’

???????protection_absent?表示是否誤碼校驗

???????profile?表示使用哪個級別的AAC，如01 Low Complexity(LC)--- AACLC

???????sampling_frequency_index?表示使用的采樣率下標

???????sampling_frequency_index sampling frequeny [Hz]

???????0x0???????96000

???????0x1???????????88200

???????0x2??????????64000

???????0x3??????????48000

???????0x4??????????44100

???????0x5??????????32000

???????0x6????????????24000

???????0x7??????????22050

???????0x8??????????16000

???????0x9??????????2000

???????0xa??????????11025

???????0xb??????????8000

???????0xc??????????reserved

???????0xd??????????reserved

???????0xe??????????reserved

???????0xf???????????reserved

???????channel_configuration?表示聲道數

???????frame_length?一個ADTS幀的長度包括ADTS頭和raw data block.

???????adts_buffer_fullness??0x7FF?說明是碼率可變的碼流

???????number_of_raw_data_blocks_in_frame

??????表示ADTS幀中有number_of_raw_data_blocks_in_frame + 1個AAC原始幀.

???????所以說number_of_raw_data_blocks_in_frame == 0?表示說ADTS幀中有一個AAC數據塊并不是說沒有。(一個AAC原始幀包含一段時間內1024個采樣及相關數據)

二 封裝AAC為ADTS幀?

???????一個AAC原始數據塊長度是可變的，對原始幀加上ADTS頭進行ADTS?的封裝，就形成了ADTS幀。通常我們將得到的AAC原始幀進行封裝后寫入文件，用常用的播放器如千千靜聽即可播放，這是個驗證AAC數據是否正確的方法。???

???????進行封裝前，需要了解相關參數，如采樣率，聲道數，原始數據塊的長度等。下面把AAC原始數據幀加工為ADTS幀，據相關參數填寫組成7字節的ADTS頭。

???????The ADTS header is defined below -????

???????unsigned int obj_type = 0;????

???????unsigned int num_data_block = frame_length / 1024;?

???????// include the header length also?????

???????frame_length += 7;???

???????/* We want the same metadata */????

???????/* Generate ADTS header */????

???????if(adts_header == NULL) return;????

???????/* Sync point over a full byte */????

???????adts_header[0] = 0xFF;????

???????/* Sync point continued over first 4 bits + static 4 bits????

??????* (ID, layer, protection)*/????

???????adts_header[1] = 0xF9;????

???????/* Object type over first 2 bits */????

???????adts_header[2] = obj_type << 6;//????

???????/* rate index over next 4 bits */????

???????adts_header[2] |= (rate_idx << 2);????

???????/* channels over last 2 bits */????

???????adts_header[2] |= (channels & 0x4) >> 2;???

???????/* channels continued over next 2 bits + 4 bits at zero */????

???????adts_header[3] = (channels & 0x3) << 6;????

???????/* frame size over last 2 bits */????

???????adts_header[3] |= (frame_length & 0x1800) >> 11;????

???????/* frame size continued over full byte */????

???????adts_header[4] = (frame_length & 0x1FF8) >> 3;????

???????/* frame size continued first 3 bits */????

???????adts_header[5] = (frame_length & 0x7) << 5;????

???????/* buffer fullness (0x7FF for VBR) over 5 last bits*/????

???????adts_header[5] |= 0x1F;????

???????/* buffer fullness (0x7FF for VBR) continued over 6 first bits + 2 zeros?????

???????* number of raw data blocks */

???????adts_header[6] = 0xFC;// one raw data blocks .

???????adts_header[6] |= num_data_block & 0x03; //Set raw Data blocks.

?

???????在CMMB中，采用AAC音頻壓縮標準，默認狀況下，編碼參數如下：雙聲道，采樣率24KHZ，幀長變長，碼流可變碼率的碼流，一般采用的AAC profile為

AAC-LC。將從CMMB復用幀解析的一個AAC原始幀封裝為ADTS幀的方法如下：

uint8 aac_buf[ADTS_FRAME_SIZE]={0x0ff,0x0f9,0x058,0x80,0,0x1f,0xfc};

???????從上述7個字節分析音頻參數如下:

???????synword--0xfff

???????ID:0x1--- 1--- MPEG2 identifier,

???????LAYER--00

???????protection_absent ---01

???????profile--01???1 Low Complexity profile (LC) AAC-LC

???????smaping_freuency_index---0110-->0x06--->采樣率24KHZ?

???????channel_configuration --- aac_buf[3] = 0x08---->2---->雙聲道。。

???????adts_buffer_fullness--->0x7ff?碼率可變的碼流?

???????現插入長度參數?wDataLen;

???????void OnAudioAacFrame(byte* data, uint16 wDataLen)

???????{

??????????????unsigned int num_data_block = wDataLen / 1024;?

??????????????uint16 frame_Length;?

??????????????frame_Length = wDataLen + 7;?????

???????????????/* frame size over last 2 bits */?????????

??????????????aac_buf[3] |= (frame_length & 0x1800) >> 11;// the upper 2 bit?????????

??????????????/* frame size continued over full byte */?????????

??????????????aac_buf[4] = (frame_length & 0x1FF8) >> 3;// the middle 8 bit????????

??????????????/* frame size continued first 3 bits */?????????

??????????????aac_buf[5] |= (frame_length & 0x7) << 5;//the last 3 bit?

??????????????aac_bug[6] |= num_data_block & 0x03; //Set raw Data blocks.

??????????????emcpy(&aac_buf[7],data,wDataLen);

??????????????//形成一個ADTS幀寫入文件。

??????????????fwrite(aac_buf,wDataLen+7,sizeof(byte),f_audio);

???????}

三 LATM格

???????LATM?的全稱為“Low-overhead MPEG-4 Audio TransportMultiplex”（低開銷音頻傳輸復用），是MPEG-4 AAC制定的一種高效率的碼流傳輸方式，MPEG-2 TS?流也采用LATM?作為AAC?音頻碼流的封裝格式之?LATM格式也以幀為單位，主要由AudioSpecificConfig（音頻特定配置單元）與音頻負載組成。

AudioSpecificConfig?描述了一個LATM?幀的信息，音頻負載主要由PayloadLengthInfo（負載長度信息）和PayloadMux（負載凈荷）組成。

???????AudioSpecificConfig?信息可以是帶內傳，也可以是帶外傳。所謂帶內傳，就是指每一個LATM?幀，都含有一個AudioSpecificConfig?信息；而帶外傳，

則每一個LATM幀都不含有AudioSpecificConfig?信息，而通過其他方式把AudioSpecificConfig信息發送到解碼端，由于AudioSpecificConfig?信息一般是不變的，所以只需發送一次即可。由此可見，AudioSpecificConfig?信息采用帶內傳輸可適應音頻編碼信息不斷變化的情況，而采用帶外傳輸，可以節省音頻傳輸碼率。帶內或帶外傳，由muxconfigPresent?標志位決定。例如流媒體應用中，muxconfigPresent?可設置為0，這樣LATM幀中將不含有AudioSpecificConfig信息，LATM幀通過RTP包發送出去，AudioSpecificConfig?可通過SDP文件一次性傳送到解碼端。

???????AudioSpecificConfig?主要參數

???????numSubFrames??子幀的數目

???????numProgram????復用的節目數

???????numLayer??????復用的層數

???????frameLengthType??負載的幀長度類型，包括固定長度與可變長度

???????audioObjectType???音頻對象類型

???????samplingFrequency???采樣率

???????channelConfiguration???聲道配置

???????音頻負載由若干子幀組成，每個子幀由PayloadLengthInfo和PayloadMux組成，與ADTS幀凈荷一樣，音頻負載主要包含原始幀數據。?

???????AAC打包成TS流通常有兩種方式，分別是先打包成ADTS或LATM。ADTS的每一幀都有個幀頭，在每個幀頭信息都一樣的狀況下，會有很大的冗余。LATM格式具有很大的靈活性，每幀的音頻配置單元既可以帶內傳輸，有可以帶外傳輸。正因為如此，LATM不僅適用于流傳輸還可以用于RTP傳輸，RTP傳輸時，若音頻數據配置信息是保持不變，可以先通過SDP會話先傳輸StreamMuxConfig（AudioSpecificConfig）信息，由于LATM流由一個包含了一個或多個音頻幀的audioMuxElements序列組成。一個完整或部分完整的audioMuxElement可直接映射到一個RTP負載上。?

???????下面是一個audoMuxEmlemt

???????AudioMuxElement(muxConfigPresent)

???????{?

??????????????if (muxConfigPresent)??

??????????????{?

?????????????????????useSameStreamMux;????

?????????????????????if (!useSameStreamMux)???????

????????????????????????????StreamMuxConfig();

?????????????}

?

??????????????if (audioMuxVersionA == 0)?

??????????????{?

?????????????????????for (i = 0; i <= numSubFrames; i++)????

?????????????????????{

?????????????????????????????PayloadLengthInfo();???????

????????????????????????????PayloadMux();???

????????????????????}.??

??????????????}

???????}

??

???????可以很簡單的把ADTS幀轉換為LATM幀，根據ADTS頭的信息，生成StreamMuxConfig，將ADTS中的原始幀提取出來，前面加上PayloadLengthInfo做為LATM的音頻幀。按照上述格式打包生成AudioMuxElement，作為RTP的負載傳輸.

?

四、 CMMB中的LATM?

?

???????當CMMB中音頻壓縮標準為AAC時，默認采用LATM封裝。StreamMuxConfig采用帶外傳輸。StreamMuxConifg中的若干默認參數如下：

???????audioMuxVersion:0標志流語法版本號為0

???????allStreamSameTiemFraming：1，標志復用到PayLoadMux()中的所有負載共享一個共同的時基

???????umSubFrames:0??表示只有一個音頻子幀.?

???????audioObjectType:2 AAC-LC freameLengthType:0?幀長度是可變的

???????latmBufferFullness:0xFF?碼率可變的碼流

?

參考：

[1]ISO/IEC 13818-7(2003 MPEG-2 AAC, Second Edition)

[2] ISO13818-7（2006 Fourth edition AAC）

[3] RFC 3016 (rfc3016) - RTP Payload Format for MPEG-4 Audio-Visual Streams

[4] AAC音頻壓縮編碼標準的ADTS與LATM格式分析

[5] GYZ 234-2008：?CMMB復用實施指南

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