感谢作者：http://blog.csdn.net/byxdaz/article/details/53993791
libRTMP 库从 RTMP 直播服务器不断地获取 RTMP 包，从RTMP包中获取音频和视频数据。RTMP只是一个应用层协议，传输的数据格式都是基于FLV格式的。
我们在推送音、视频包之前，会首先向服务器推送一个音、视频同步包，该包包含了 AAC 音频帧以及 H264 码流的解码信息。因此当我们首次向服务器请求 RTMP 包之后，服务器会下发给我们对应的音、视频同步包，这将决定我们如何解析并重组音、视频。
一、首先要了解FLV格式。
flv文件主要由两部分组成：header和body。
header部分记录了flv的类型、版本等信息，是flv的开头，一般都差不多，占9bytes。具体格式如下：
文件类型
3 bytes
“FLV”
版本
1 byte
一般为0x01
流信息
1 byte
倒数第一位是1表示有视频，倒数第三位是1表示有音频，倒数第二、四位必须为0
header长度
4 bytes
整个header的长度，一般为9；大于9表示下面还有扩展信息
body部分由一个个Tag组成，每个Tag的下面有一块4bytes的空间，用来记录这个tag的长度，这个后置用于逆向读取处理，他们的关系如下图：
2.1.Tag
每个Tag由也是由两部分组成的：Tag Header和Tag Data。Tag Header里存放的是当前Tag的类型、数据区（Tag Data）长度等信息，具体如下：
名称
长度
介绍
Tag类型
1 bytes
8：音频
9：视频
18：脚本
其他：保留
数据区长度
3 bytes
在数据区的长度
时间戳
3 bytes
整数，单位是毫秒。对于脚本型的tag总是0
时间戳扩展
1 bytes
将时间戳扩展为4bytes，代表高8位。很少用到
StreamsID
3 bytes
总是0
数据区(data)
由数据区长度决定
数据实体
2.2.Tag Data
数据区根据Tag类型的不同可分为三种，音频数据、视频数据和脚本数据。
2.2.1.音频数据
音频数据包括音频同步包和音频raw数据。
音频同步包格式
名称
二进制值
Hex
介绍
音频格式
4 bits
[AAC]1010
0xA
0 = Linear PCM, platform endian
1 = ADPCM
2 = MP3
3 = Linear PCM, little endian
4 = Nellymoser 16-kHz mono
5 = Nellymoser 8-kHz mono
6 = Nellymoser
7 = G.711 A-law logarithmic PCM
8 = G.711 mu-law logarithmic PCM
9 = reserved
10 = AAC
11 = Speex
14 = MP3 8-Khz
15 = Device-specific sound
采样率
2 bits
[44kHZ]11
0 = 5.5-kHz
1 = 11-kHz
2 = 22-kHz
3 = 44-kHz
对于AAC总是3
采样的长度
1 bit
0 = snd8Bit
1 = snd16Bit
压缩过的音频都是16bit
音频类型
1 bit
0 = sndMono，单声道
1 = sndStereo，立体声
对于AAC总是1
ACCPacketType
8bit
00000000
只有SoundFormat == 10时才有此8bi的字段
0x00,表示音频同步包；0x01，表示音频raw数据。
AudioObjectType
5bits
[AAC LC]00010
SampleRateIndex
4bits
[44100]0100
ChannelConfig
4bits [Stereo]0010
FrameLengthFlag
1bit
0
dependOnCoreCoder
1bit
0
extensionFlag
1bit
0
共4个字节（2bytes AACDecoderSpecificInfo和2bytesAudioSpecificConfig）
注：speex数据
每个音频包（相当于FLV中的audiotag）中，第一个字节的前四位表示编码格式，等于11说明为speex编码。后4个字节分别表示编码采样率、单声道or立体声、每个sample大小8位or16位。但采用speex编码时，它们是固定不变的，协议中的为无效数据。编码采样率恒为16khz，单声道，16bit/sample。 剩余的数据为音频帧数据。
AACsequence header存放的是AudioSpecificConfig结构，该结构则在“ISO-14496-3Audio”中描述。AudioSpecificConfig结构的描述非常复杂，这里我做一下简化，事先设定要将要编码的音频格式，其中，选择"AAC-LC"为音频编码，音频采样率为44100，于是AudioSpecificConfig简化为下表：
音频raw数据格式
名称
二进制值
Hex
介绍
音频格式
4 bits
[AAC]1010
0xA
0 = Linear PCM, platform endian
1 = ADPCM
2 = MP3
3 = Linear PCM, little endian
4 = Nellymoser 16-kHz mono
5 = Nellymoser 8-kHz mono
6 = Nellymoser
7 = G.711 A-law logarithmic PCM
8 = G.711 mu-law logarithmic PCM
9 = reserved
10 = AAC
11 = Speex
14 = MP3 8-Khz
15 = Device-specific sound
采样率
2 bits
[44kHZ]11
0 = 5.5-kHz
1 = 11-kHz
2 = 22-kHz
3 = 44-kHz
对于AAC总是3
采样的长度
1 bit
0 = snd8Bit
1 = snd16Bit
压缩过的音频都是16bit
音频类型
1 bit
0 = sndMono，单声道
1 = sndStereo，立体声
对于AAC总是1
ACCPacketType
8bit
只有SoundFormat == 10时才有此8bi的字段
0x00,表示音频同步包；0x01，表示音频raw数据。
Raw data
音频raw数据
AudioSpecificConfig部分参数含义见：
https://wiki.multimedia.cx/index.php?title=MPEG-4_Audio
2.2.2.视频数据
视频同步包
Field
Type
Comment
Frame Type
UB [4]
Type of video frame. The following values are defined:
1 = key frame (for AVC, a seekable frame)
2 = inter frame (for AVC, a non-seekable frame)
3 = disposable inter frame (H.263 only)
4 = generated key frame (reserved for server use only)
5 = video info/command frame
CodecID
UB [4]
Codec Identifier. The following values are defined:
2 = Sorenson H.263
3 = Screen video
4 = On2 VP6
5 = On2 VP6 with alpha channel
6 = Screen video version 2
7 = AVC
AVCPacketType
IF CodecID == 7
UI8
The following values are defined:
0 = AVC sequence header
1 = AVC NALU
2 = AVC end of sequence (lower level NALU sequence ender is not required or supported)
CompositionTime
IF CodecID == 7
SI24
IF AVCPacketType == 1
Composition time offset
ELSE
0
See ISO 14496-12, 8.15.3 for an explanation of composition
times. The offset in an FLV file is always in milliseconds.
VideoTagHeader的头1个字节，也就是接跟着StreamID的1个字节包含着视频帧类型及视频CodecID最基本信息.表里列的十分清楚.
VideoTagHeader之后跟着的就是VIDEODATA数据了，也就是video payload.当然就像音频AAC一样，这里也有特例就是如果视频的格式是AVC（H.264）的话，VideoTagHeader会多出4个字节的信息.
AVCPacketType和 CompositionTime。AVCPacketType表示接下来 VIDEODATA（AVCVIDEOPACKET）的内容：
IF AVCPacketType == 0 AVCDecoderConfigurationRecord（AVC sequence header）
IF AVCPacketType == 1 One or more NALUs (Full frames are required)
AVCDecoderConfigurationRecord.包含着是H.264解码相关比较重要的sps和pps信息，再给AVC解码器送数据流之前一定要把sps和pps信息送出，否则的话解码器不能正常解码。而且在解码器stop之后再次start之前，如seek、快进快退状态切换等，都需要重新送一遍sps和pps的信息.AVCDecoderConfigurationRecord在FLV文件中一般情况也是出现1次，也就是第一个video tag.
AVC sequence header就是AVCDecoderConfigurationRecord结构，该结构在标准文档“ISO-14496-15AVC file format”中有详细说明。
2.2.3脚本数据
脚本Tag一般只有一个，是flv的第一个Tag，用于存放flv的信息，比如duration、audiodatarate、creator、width等。
首先介绍下脚本的数据类型。所有数据都是以数据类型+（数据长度）+数据的格式出现的，数据类型占1byte，数据长度看数据类型是否存在，后面才是数据。
其中数据类型的种类有：
- 0 = Number type
- 1 = Boolean type
- 2 = String type
- 3 = Object type
- 4 = MovieClip type
- 5 = Null type
- 6 = Undefined type
- 7 = Reference type
- 8 = ECMA array type
- 10 = Strict array type
- 11 = Date type
- 12 = Long string type
如果类型为String，后面的2bytes为字符串的长度（LongString是4bytes），再后面才是字符串数据；如果是Number类型，后面的8bytes为Double类型的数据；Boolean类型，后面1byte为Bool类型。
知道了这些后再来看看flv中的脚本，一般开头是0x02，表示String类型，后面的2bytes为字符串长度，一般是0x000a（“onMetaData”的长度），再后面就是字符串“onMetaData”。好像flv格式的文件都有onMetaData标记，在运行ActionScript的时候会用到它。后面跟的是0x08，表示ECMA Array类型，这个和Map比较相似，一个键跟着一个值。键都是String类型的，所以开头的0x02被省略了，直接跟着的是字符串的长度，然后是字符串，再是值的类型，也就是上面介绍的那些了。
第一个AMF包：第1个字节表示AMF包类型。
第二个AMF包：
第1个字节表示AMF包类型，一般总是0x08，表示数组。第2-5个字节为UI32类型值，表示数组元素的个数。后面即为各数组元素的封装，数组元素为元素名称和值组成的对。常见的数组元素如下表：
一、 通过RTMP_ReadPacket获取一包数据，在里面获取视频、音频、scripttag等。
视频H264获取过程:
1、获取第一视频同步包之后，解析到sps、pps数据。视频头增加H264边界分割符（00 00 00 01 或000001分隔符）。
2、其他视频包，解析出nalu数据。视频头增加H264边界分割符。
音频AAC获取过程:
1、获取第一音频同步包之后，解析到AACDecoderSpecificInfo和AudioSpecificConfig。AudioSpecificConfig用于生成ADST。
2、其他音频包，解析出原始音频数据（即es）。
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ADTSheader。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
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：MPEGVersion: 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-2AAC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。
Thereare 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。
代码
-
-
-
- #include <stdio.h>
- #include "librtmp/rtmp_sys.h"
- #include "librtmp/log.h"
-
-
- typedef struct tagAACDecoderSpecific
- {
- unsigned char nAudioFortmatType;
- unsigned char nAudioSampleType;
- unsigned char nAudioSizeType;
- unsigned char nAudioStereo;
- unsigned char nAccPacketType;
- }AACDecoderSpecific;
-
- typedef struct tagAudioSpecificConfig
- {
- unsigned char nAudioObjectType;
- unsigned char nSampleFrequencyIndex;
- unsigned char nChannels;
- unsigned char nFrameLengthFlag;
- unsigned char nDependOnCoreCoder;
- unsigned char nExtensionFlag;
- }AudioSpecificConfig;
-
- int InitSockets()
- {
- #ifdef WIN32
- WORD version;
- WSADATA wsaData;
- version = MAKEWORD(1, 1);
- return (WSAStartup(version, &wsaData) == 0);
- #endif
- }
-
- void CleanupSockets()
- {
- #ifdef WIN32
- WSACleanup();
- #endif
- }
-
-
- void ParseScriptTag(char *pBuffer, int nBufferLength, int &nVideoCodecId, int &nVideoWidth, int &nVideoHeight, int &nFrameRate, int &nAudioCodecId, int &nAudioSampleRate, int &nAudioSampleSize, bool &bStereo, int &nFileSize)
- {
- AMFObject obj;
- AVal val;
- AMFObjectProperty * property;
- AMFObject subObject;
- int nRes = AMF_Decode(&obj, pBuffer, nBufferLength, FALSE);
- if (nRes < 0)
- {
-
- return;
- }
-
- AMF_Dump(&obj);
- for (int n = 0; n < obj.o_num; n++)
- {
- property = AMF_GetProp(&obj, NULL, n);
- if (property != NULL)
- {
- if (property->p_type == AMF_OBJECT)
- {
- AMFProp_GetObject(property, &subObject);
- for (int m = 0; m < subObject.o_num; m++)
- {
- property = AMF_GetProp(&subObject, NULL, m);
- if (property != NULL)
- {
- if (property->p_type == AMF_OBJECT)
- {
-
- }
- else if (property->p_type == AMF_BOOLEAN)
- {
- int bVal = AMFProp_GetBoolean(property);
- if (strncmp("stereo", property->p_name.av_val, property->p_name.av_len) == 0)
- {
- bStereo = bVal > 0 ? true : false;
- }
- }
- else if (property->p_type == AMF_NUMBER)
- {
- double dVal = AMFProp_GetNumber(property);
- if (strncmp("width", property->p_name.av_val, property->p_name.av_len) == 0)
- {
- nVideoWidth = (int)dVal;
- }
- else if (stricmp("height", property->p_name.av_val) == 0)
- {
- nVideoHeight = (int)dVal;
- }
- else if (stricmp("framerate", property->p_name.av_val) == 0)
- {
- nFrameRate = (int)dVal;
- }
- else if (stricmp("videocodecid", property->p_name.av_val) == 0)
- {
- nVideoCodecId = (int)dVal;
- }
- else if (stricmp("audiosamplerate", property->p_name.av_val) == 0)
- {
- nAudioSampleRate = (int)dVal;
- }
- else if (stricmp("audiosamplesize", property->p_name.av_val) == 0)
- {
- nAudioSampleSize = (int)dVal;
- }
- else if (stricmp("audiocodecid", property->p_name.av_val) == 0)
- {
- nAudioCodecId = (int)dVal;
- }
- else if (stricmp("filesize", property->p_name.av_val) == 0)
- {
- nFileSize = (int)dVal;
- }
- }
- else if (property->p_type == AMF_STRING)
- {
- AMFProp_GetString(property, &val);
- }
-
- }
- }
- }
- else
- {
- AMFProp_GetString(property, &val);
- }
- }
- }
- }
-
-
- int CreateADTS(AudioSpecificConfig ascAudioSpecificConfig, unsigned short nAudioFrameLength, char *pADTS)
- {
- if (pADTS == NULL)
- return -1;
-
- unsigned char adts[7] = { 0 };
- int chanCfg = ascAudioSpecificConfig.nChannels;
-
- adts[0] = (byte)0xFF;
- adts[1] = (byte)0xF9;
- adts[2] = (byte)(((ascAudioSpecificConfig.nAudioObjectType - 1) << 6) + (ascAudioSpecificConfig.nSampleFrequencyIndex << 2) + (chanCfg >> 2));
- adts[3] = (byte)(((chanCfg & 3) << 6) + (nAudioFrameLength >> 11));
- adts[4] = (byte)((nAudioFrameLength & 0x7FF) >> 3);
- adts[5] = (byte)(((nAudioFrameLength & 7) << 5) + 0x1F);
- adts[6] = (byte)0xFC;
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
- if (pADTS != NULL)
- {
- memcpy(pADTS, adts, sizeof(adts));
- }
- return 0;
- }
-
- int main(int argc, char* argv[])
- {
- InitSockets();
-
- double duration=-1;
- int nRead;
-
- bool bLiveStream=true;
-
-
- int bufsize=1024*1024*10;
- char *buf=(char*)malloc(bufsize);
- memset(buf,0,bufsize);
- long countbufsize=0;
-
-
- if (0 == 1)
- {
- FILE *fp = fopen("receive.flv", "wb");
- if (!fp){
- RTMP_LogPrintf("Open File Error.\n");
- CleanupSockets();
- return -1;
- }
-
-
-
-
-
- RTMP *rtmp = RTMP_Alloc();
- RTMP_Init(rtmp);
-
- rtmp->Link.timeout = 30;
-
- if(!RTMP_SetupURL(rtmp,"rtmp://live.hkstv.hk.lxdns.com/live/hks"))
-
-
-
- {
- RTMP_Log(RTMP_LOGERROR, "SetupURL Err\n");
- RTMP_Free(rtmp);
- CleanupSockets();
- return -1;
- }
-
-
- RTMP_SetBufferMS(rtmp, 3600 * 1000);
-
- if (!RTMP_Connect(rtmp, NULL)){
- RTMP_Log(RTMP_LOGERROR, "Connect Err\n");
- RTMP_Free(rtmp);
- CleanupSockets();
- return -1;
- }
-
- if (!RTMP_ConnectStream(rtmp, 0)){
- RTMP_Log(RTMP_LOGERROR, "ConnectStream Err\n");
- RTMP_Close(rtmp);
- RTMP_Free(rtmp);
- CleanupSockets();
- return -1;
- }
-
-
- while(nRead=RTMP_Read(rtmp,buf,bufsize)){
- fwrite(buf,1,nRead,fp);
- countbufsize+=nRead;
- RTMP_LogPrintf("Receive: %5dByte, Total: %5.2fkB\n",nRead,countbufsize*1.0/1024);
- }
- if (fp)
- fclose(fp);
-
- if (buf){
- free(buf);
- }
-
- if (rtmp){
- RTMP_Close(rtmp);
- RTMP_Free(rtmp);
- CleanupSockets();
- rtmp = NULL;
- }
- return 0;
- }
-
-
- if (0 == 0)
- {
- FILE *fp = fopen("receive.h264", "wb");
- if (!fp){
- RTMP_LogPrintf("Open File Error.\n");
- CleanupSockets();
- return -1;
- }
- FILE *fpAAC = fopen("receive.m4a", "wb");
- if(!fpAAC){
- RTMP_LogPrintf("Open File Error.\n");
- CleanupSockets();
- return -1;
- }
-
-
-
-
-
- RTMP *rtmp = RTMP_Alloc();
- RTMP_Init(rtmp);
-
- rtmp->Link.timeout = 30;
-
- if(!RTMP_SetupURL(rtmp,"rtmp://live.hkstv.hk.lxdns.com/live/hks"))
-
-
- {
- RTMP_Log(RTMP_LOGERROR, "SetupURL Err\n");
- RTMP_Free(rtmp);
- CleanupSockets();
- return -1;
- }
- if (bLiveStream){
- rtmp->Link.lFlags |= RTMP_LF_LIVE;
- }
-
-
- RTMP_SetBufferMS(rtmp, 3600 * 1000);
-
- if (!RTMP_Connect(rtmp, NULL)){
- RTMP_Log(RTMP_LOGERROR, "Connect Err\n");
- RTMP_Free(rtmp);
- CleanupSockets();
- return -1;
- }
-
- if (!RTMP_ConnectStream(rtmp, 0)){
- RTMP_Log(RTMP_LOGERROR, "ConnectStream Err\n");
- RTMP_Close(rtmp);
- RTMP_Free(rtmp);
- CleanupSockets();
- return -1;
- }
-
- RTMPPacket pc = { 0 };
- bool bVideoFirst = true;
- bool bAudioFirst = true;
- unsigned int nFrameType = 0;
- unsigned char result = 0;
- char *data = NULL;
-
- static const char flvHeader[] = { 'F', 'L', 'V', 0x01,
- 0x00,
- 0x00, 0x00, 0x00, 0x09,
- 0x00, 0x00, 0x00, 0x00
- };
-
- int jjj = 0;
- AudioSpecificConfig ascAudioSpecificConfig = { 0 };
- while (RTMP_ReadPacket(rtmp, &pc))
- {
- if (RTMPPacket_IsReady(&pc))
- {
- if (!pc.m_nBodySize)
- continue;
- if (pc.m_packetType == RTMP_PACKET_TYPE_VIDEO && RTMP_ClientPacket(rtmp, &pc))
- {
- jjj++;
- if (jjj > 1000)
- break;
- result = pc.m_body[0];
- data = pc.m_body;
- bool bIsKeyFrame = false;
- if (result == 0x17)
- {
- bIsKeyFrame = true;
- }
- else if (result == 0x27)
- {
- bIsKeyFrame = false;
- }
- static unsigned char const start_code[4] = { 0x00, 0x00, 0x00, 0x01 };
-
-
- if (bVideoFirst)
- {
- fwrite(start_code, 1, 4, fp);
-
-
-
- int spsnum = data[10] & 0x1f;
- int number_sps = 11;
- int count_sps = 1;
- while (count_sps <= spsnum){
- int spslen = (data[number_sps] & 0x000000FF) << 8 | (data[number_sps + 1] & 0x000000FF);
- number_sps += 2;
- fwrite(data + number_sps, 1, spslen, fp);
- fwrite(start_code, 1, 4, fp);
- number_sps += spslen;
- count_sps++;
- }
-
- int ppsnum = data[number_sps] & 0x1f;
- int number_pps = number_sps + 1;
- int count_pps = 1;
- while (count_pps <= ppsnum){
- int ppslen = (data[number_pps] & 0x000000FF) << 8 | data[number_pps + 1] & 0x000000FF;
- number_pps += 2;
- fwrite(data + number_pps, 1, ppslen, fp);
- fwrite(start_code, 1, 4, fp);
- number_pps += ppslen;
- count_pps++;
- }
- bVideoFirst = false;
- }
- else
- {
-
- int len = 0;
- int num = 5;
- while (num<pc.m_nBodySize)
- {
- len = (data[num] & 0x000000FF) << 24 | (data[num + 1] & 0x000000FF) << 16 | (data[num + 2] & 0x000000FF) << 8 | data[num + 3] & 0x000000FF;
- num = num + 4;
- fwrite(data + num, 1, len, fp);
- fwrite(start_code, 1, 4, fp);
- num = num + len;
- }
- }
- }
- else if (pc.m_packetType == RTMP_PACKET_TYPE_AUDIO && RTMP_ClientPacket(rtmp, &pc))
- {
- data = pc.m_body;
- if (bAudioFirst)
- {
-
- AACDecoderSpecific adsAACDecoderSpecific = { 0 };
- adsAACDecoderSpecific.nAudioFortmatType = (data[0] & 0xf0) >> 4;
- adsAACDecoderSpecific.nAudioSampleType = (data[0] & 0x0c) >> 2;
- adsAACDecoderSpecific.nAudioSizeType = (data[0] & 0x02) >> 1;
- adsAACDecoderSpecific.nAudioStereo = data[0] & 0x01;
- if(adsAACDecoderSpecific.nAudioFortmatType == 10)
- {
-
-
-
- adsAACDecoderSpecific.nAccPacketType = data[1];
- unsigned short audioSpecificConfig = 0;
- audioSpecificConfig = (data[2] & 0xff) << 8;
- audioSpecificConfig += 0x00ff & data[3];
- ascAudioSpecificConfig.nAudioObjectType = (audioSpecificConfig & 0xF800) >> 11;
- ascAudioSpecificConfig.nSampleFrequencyIndex = (audioSpecificConfig & 0x0780) >> 7;
- ascAudioSpecificConfig.nChannels = (audioSpecificConfig & 0x78) >> 3;
- ascAudioSpecificConfig.nFrameLengthFlag = (audioSpecificConfig & 0x04) >> 2;
- ascAudioSpecificConfig.nDependOnCoreCoder = (audioSpecificConfig & 0x02) >> 1;
- ascAudioSpecificConfig.nExtensionFlag = audioSpecificConfig & 0x01;
- }
- else if(adsAACDecoderSpecific.nAudioFortmatType == 11)
- {
-
- adsAACDecoderSpecific.nAudioStereo = 0;
- adsAACDecoderSpecific.nAudioSizeType = 1;
- adsAACDecoderSpecific.nAudioSampleType = 4;
- }
-
- bAudioFirst = false;
- }
- else
- {
- if(pc.m_nBodySize > 2)
- {
- if(data[1] == 1)
- {
-
-
- char szADTSTemp[8] = { 0 };
- CreateADTS(ascAudioSpecificConfig, pc.m_nBodySize - 2 + 7, szADTSTemp);
- fwrite(szADTSTemp, 7, 1, fpAAC);
-
- fwrite(pc.m_body + 2, pc.m_nBodySize - 2, 1, fpAAC);
- }
- else
- {
- int kkkkk = 0;
- }
- }
- else
- {
- int kkkk = 0;
- }
- }
-
-
- }
- else if (pc.m_packetType == RTMP_PACKET_TYPE_INFO && RTMP_ClientPacket(rtmp, &pc))
- {
- int nVideoCodecId = 0;
- int nVideoWidth = 255;
- int nVideoHeight = 188;
- int nVideoFrameRate = 25;
- int nAudioCodecId = 0;
- int nAudioSampleRate = 0;
- int nAudioSampleSize = 0;
- bool bStereo = false;
- int nFileSize = 0;
- ParseScriptTag(pc.m_body, pc.m_nBodySize, nVideoCodecId, nVideoWidth, nVideoHeight, nVideoFrameRate, nAudioCodecId, nAudioSampleRate, nAudioSampleSize, bStereo, nFileSize);
- int k = 0;
- }
- RTMPPacket_Free(&pc);
- }
- }
-
- if (fp)
- fclose(fp);
-
- if (fpAAC)
- fclose(fpAAC);
-
- if (buf){
- free(buf);
- }
-
- if (rtmp){
- RTMP_Close(rtmp);
- RTMP_Free(rtmp);
- CleanupSockets();
- rtmp = NULL;
- }
- }
-
- return 0;
- }
注：H.264视频分析软件(Elecard.Streameye.Tools)分析H264数据，使用暴风影音播放m4a文件中的aac音频。
代码下载：点击链接
参考资料：
http://blog.csdn.net/leixiaohua1020/article/details/42104893
http://blog.chinaunix.net/uid-15063109-id-4273162.html
http://blog.csdn.net/bsplover/article/details/7426511
http://blog.csdn.net/zqf_office/article/details/50868520
