使用Librtmp收流，接收H264 + AAC（附代碼）

用librtmp接收H264+AAC的例子代碼網上參考資源比較少，這份代碼來自www.cnblogs.com的某個博客，但是已經找不到原鏈接了。把代碼附上，幫助各位網友在開發時少走些彎路。

//////////////////////////////////////////////////////////

#include "librtmp/rtmp_sys.h"
#include "librtmp/log.h"

 
#define HTON16(x)  ((x>>8&0xff)|(x<<8&0xff00))
#define HTON24(x)  ((x>>16&0xff)|(x<<16&0xff0000)|(x&0xff00))
#define HTON32(x)  ((x>>24&0xff)|(x>>8&0xff00)|(x<<8&0xff0000)|(x<<24&0xff000000))
#define HTONTIME(x) ((x>>16&0xff)|(x<<16&0xff0000)|(x&0xff00)|(x&0xff000000))

#pragma comment(lib, "librtmp.lib")

#pragma pack(1)
 
typedef struct _FLVHead
{
    char type[3];
    char version;
    char stream_info;
    int offset;
}FLVHead;
 
typedef struct _FLVTag
{
    int tag_size;
    char type;
    char length[3];
    char timecamp[3];
    char timecampex;
    char StreamsID[3];
}FLVTag;
 
#pragma pack()

char* ByteRead8(int *i8, char *buf) 
{
    memcpy(i8, buf, 1);
    return buf+1;
}
 
char* ByteRead16(int *i16, char *buf)
{ 
    memcpy(i16, buf, 2);
    *i16 = HTON16(*i16);
    return buf+2;
}
 
char* ByteRead32(int *i32, char *buf)
{
    memcpy(i32, buf, 4);
    *i32 = HTON32(*i32);
    return buf+4;
}

void UnPackH264(char *data,int len, char * outData, int & outLen)
{
    int h264space = 0x01000000;
    char *buf = data+1;
    char * pOutBuf = outData;

    int avctype = 0;
    buf = ByteRead8(&avctype,buf);
    buf += 3;
     
    int templength = 0;
    char *tempbuff = NULL;
    if (avctype == 0)
    {
        buf += 6;
         
        buf = ByteRead16(&templength,buf);
         
        tempbuff = buf;
       // fwrite(&h264space, 1, 4, h264file);
		memcpy(pOutBuf, &h264space, 4);
		pOutBuf += 4;
	
       // fwrite(tempbuff, 1, templength, h264file);
		memcpy(pOutBuf, tempbuff, templength);
		pOutBuf += templength;
        buf += templength;
         
        buf = ByteRead8(&templength, buf);
        buf = ByteRead16(&templength, buf);
         
        tempbuff = buf;
      //  fwrite(&h264space, 1, 4, h264file); 
		memcpy(pOutBuf, &h264space, 4);
		pOutBuf += 4;
       // fwrite(tempbuff, 1, templength, h264file);
		memcpy(pOutBuf, tempbuff, templength);
		pOutBuf += templength;
        buf += templength;
    }
    else
    {
        int countsize = 2 + 3; 
        while (countsize < len)
        {
            buf = ByteRead32(&templength, buf);
             
            tempbuff = buf;
            //fwrite(&h264space, 1, 4, h264file); 
			memcpy(pOutBuf, &h264space, 4);
			pOutBuf += 4;
           // fwrite(tempbuff, 1, templength, h264file);
			memcpy(pOutBuf, tempbuff, templength);
		    pOutBuf += templength;
            buf += templength;
            countsize += (templength + 4);
        } 
    }

   outLen = pOutBuf - outData;
}


int AAC_TO_ADTS(unsigned char * bufIn, int len,  int audioSamprate, unsigned char* pBufOut, const int nBufSize, int* pOutLen)
{
	//TRACE("AAC Header: %02x%02x%02x%02x \n", bufIn[0], bufIn[1], bufIn[2], bufIn[3]);

    unsigned char ADTS[] = {0xFF, 0xF9, 0x00, 0x00, 0x00, 0x00, 0xFC}; 
    int audioChannel = 2;//音頻聲道 1或2
    int audioBit = 16;//16位 固定
    //unsigned int framelen;
	const int header = 7; // header = sizeof(ADTS);
   
	unsigned char * pBufSrc = bufIn;
    unsigned char * pBufDest = pBufOut;

	*pOutLen = 0;

    switch(audioSamprate) //音頻採樣率
    {
	case  8000:
	    ADTS[2] = 0x6C;
		break;
    case  16000:
        ADTS[2] = 0x60;
        break;
    case  32000:
        ADTS[2] = 0x54;//0xD4
        break;
    case  44100:
        ADTS[2] = 0x50;
        break;
    case  48000:
        ADTS[2] = 0x4C;
        break;
    case  96000:
        ADTS[2] = 0x40;
        break;
	case 11025:
		ADTS[2] = 0x68;
		break;
	case 12000:
		ADTS[2] = 0xE4;
		break;
	case 22050:
		ADTS[2] = 0xDC;
		break;
    default:
        break;
    }
    ADTS[3] = (audioChannel==2)?0x80:0x40;

#if 0
	{
		unsigned short au_sizes_array[32] = {0};
		unsigned int framelen = 0;
	
		int au_size = pBufSrc[1]>>3;
		int au_num = au_size>>1;
		ASSERT(au_num < 32);

		pBufSrc+=2;

		for(int i=0; i<au_num; i++)
		{
		  framelen =  ((unsigned int)pBufSrc[0] << 5) & 0x1FE0;
		  framelen |= ((pBufSrc[1] >> 3) & 0x1f);
          au_sizes_array[i] = framelen;

		  pBufSrc += 2;
		}

		TRACE("Au_Num = %d \n", au_num);

		for(int i=0; i<au_num; i++)
		{
			framelen = au_sizes_array[i];

			ADTS[3] |= ((framelen+header) & 0x1800) >> 11;
			ADTS[4] = ((framelen+header) & 0x1FF8) >> 3;
			ADTS[5] = ((framelen+header) & 0x0007) << 5;
			ADTS[5] |= 0x1F;

			if (*pOutLen + framelen + header > nBufSize)
				break;

			memcpy(pBufDest, ADTS, sizeof(ADTS));
			memcpy(pBufDest+header, pBufSrc, framelen);


			pBufDest += framelen+header;
			*pOutLen += framelen+header;
			pBufSrc += framelen;
		}

	}
#else
	unsigned int framelen = 0;
	{

		framelen = len - 2;

		ADTS[3] |= ((framelen+header) & 0x1800) >> 11;
		ADTS[4] = ((framelen+header) & 0x1FF8) >> 3;
		ADTS[5] = ((framelen+header) & 0x0007) << 5;
		ADTS[5] |= 0x1F;

		memcpy(pBufDest, ADTS, sizeof(ADTS));
		memcpy(pBufDest+header, pBufSrc+2, framelen);
	
		pBufDest += framelen+header;
		*pOutLen += framelen+header;
	}
#endif


    return 1;
}


//定義變量
RTMP *  rtmp; 
string   m_InputUrl; //RTMP URL
bool    m_stop_status; //是否退出接收
char*   m_h264data;
BYTE *    m_aacADTSBuf;
bool    m_bSpsPpsGot; //是否已拿到SPS和PPS
BYTE    m_SpsPpsBuffer[2048];
int     m_nSpsPpsSize; 

#define ONE_AUDIO_FRAME_SIZE 192000

//初始化變量
BOOL Init()
{
	m_h264data = new char[1024*1000];
	m_aacADTSBuf = new BYTE[ONE_AUDIO_FRAME_SIZE];

	m_bSpsPpsGot = false;
	m_nSpsPpsSize = 0;
	memset(m_SpsPpsBuffer, 0, sizeof(m_SpsPpsBuffer));

    m_stop_status = false;

   //////////////////////////////////////////////
    bool bLiveStream = true;		

		/* set log level */
	//RTMP_LogLevel loglvl=RTMP_LOGDEBUG;
	//RTMP_LogSetLevel(loglvl);

	rtmp = RTMP_Alloc();
	RTMP_Init(rtmp);

	//set connection timeout,default 30s
	rtmp->Link.timeout=10;	

	// HKS's live URL
	if(!RTMP_SetupURL(rtmp, (char*)m_InputUrl.c_str()))
	{
		TRACE("RTMP_SetupURL Err\n");
		RTMP_Free(rtmp);
		return FALSE;
	}
	if (bLiveStream){
		rtmp->Link.lFlags|=RTMP_LF_LIVE;
	}


	RTMP_SetBufferMS(rtmp, 5*1000);		
	
	if(!RTMP_Connect(rtmp,NULL))
	{
		TRACE("RTMP_Connect Err\n");
		RTMP_Free(rtmp);
		rtmp=NULL;
		return FALSE;
	}

	if(!RTMP_ConnectStream(rtmp,0))
	{
		TRACE("ConnectStream Err\n");
		RTMP_Close(rtmp);
		RTMP_Free(rtmp);
		rtmp=NULL;
		return FALSE;
	}

	return TRUE;
}

//釋放對象
void UnInit()
{
   	if(rtmp)
	{
		RTMP_Close(rtmp);
		RTMP_Free(rtmp);
		rtmp=NULL;
	}

    if(m_h264data)
	{
		delete m_h264data;
		m_h264data = NULL;
	}

    if(m_aacADTSBuf != NULL)
	{
		delete m_aacADTSBuf;
		m_aacADTSBuf = NULL;
	}

}

//接收RTMP和解析
void readAndDecode()
{
	int nVideoFramesNum = 0;
	int64_t  first_pts_time = 0;

	DWORD start_time = GetTickCount(); 
	int64_t pts_time;

	long countbufsize=0;

	int nRead = 0;

	FLVHead  flvHead;
	FLVTag   flvTag;

	memset(&flvHead,0,sizeof(flvHead));

	nRead = RTMP_Read(rtmp,(char*)&flvHead,sizeof(FLVHead));

	if(nRead <= 0)
	{
		TRACE("RTMP_Read failed \n");
		return;
	}

	int bufsize = 1024*1024*1;			
	char *data =(char*)malloc(bufsize);
	memset(data,0,bufsize);

	flvHead.offset = HTON32(flvHead.offset);
	TRACE("文件類型: %c %c %c\n",flvHead.type[0],flvHead.type[1],flvHead.type[2]);
	TRACE("版本: %d\n",flvHead.version);
	TRACE("流信息: %d\n",flvHead.stream_info);
	TRACE("head長度: %d\n",flvHead.offset);

    AVCodecID audio_codec_id =  AV_CODEC_ID_AAC;
	
	memset(&flvTag,0,sizeof(FLVTag));

	while(1)
	{
		if(m_stop_status == true)
		{
			break;
		}

		//音視頻
		if( (nRead = RTMP_Read(rtmp,(char*)&flvTag,sizeof(FLVTag))) == sizeof(FLVTag))
		{
			flvTag.tag_size = HTON32(flvTag.tag_size);

			int length = 0;
			memcpy(&length,flvTag.length,3);
			length = HTON24(length);

			unsigned int time = 0;
			memcpy(&time,flvTag.timecamp,3);
			time = HTONTIME(time);

			//TRACE("\nTag大小: %d\n",flvTag.tag_size);
			//TRACE("Tag類型: %d, 長度：%d, 時間戳: %d\n",flvTag.type, length, time);

			pts_time = time*90; //轉成90KHz爲單位

			nRead = RTMP_Read(rtmp,data,length);
		    if(nRead > 0)
			{
				if(flvTag.type == 8) //音頻
				{
					if(audio_codec_id == AV_CODEC_ID_AAC)
					{
						int nOutAACLen = 0;

						AAC_TO_ADTS((unsigned char*)data, nRead, 44100, m_aacADTSBuf, ONE_AUDIO_FRAME_SIZE, &nOutAACLen);

						//DecodeAudio(m_aacADTSBuf, nOutAACLen, AV_CODEC_ID_AAC, TRUE);
					}
					else
					{
						//DecodeAudio(data, nRead, AV_CODEC_ID_AAC, TRUE);
					}
	
				}
				else if(flvTag.type == 9) 	//視頻
				{

					nVideoFramesNum++;

					if(!(data[0] == 0x0 && data[1] == 0x0 && data[2] == 0x0 && data[3] == 0x01))
					{
						//TRACE("Not H264 StartCode!\n");

						int outLen = 0;
						UnPackH264(data, nRead, m_h264data, outLen); //每個NALU前已經添加了四個字節的開始碼(0x0000001)

						if(outLen < 4)
						{
							TRACE("Video frame was too short! \n");
							continue;
						}

						int nalu_type = (m_h264data[4] & 0x1F);

						TRACE("FrameNo: %d, nalu_type: %d, size: %d \n", nVideoFramesNum, nalu_type, outLen);

						if(!m_bSpsPpsGot && nalu_type == 7)
						{			
				            ASSERT(outLen < sizeof(m_SpsPpsBuffer));

							memcpy(m_SpsPpsBuffer,m_h264data, outLen);
							m_nSpsPpsSize = outLen;
							m_bSpsPpsGot = true;

						}

						if(nalu_type == 5) //I frame
						{
							if(m_bSpsPpsGot) //在I幀前插入SPS和PPS
							{
								memmove(m_h264data + m_nSpsPpsSize, m_h264data, outLen);
								memcpy(m_h264data, m_SpsPpsBuffer, m_nSpsPpsSize);
								outLen += m_nSpsPpsSize;
							}
						}
	

						if(outLen > 0)
						{
							//DecodeVideo((BYTE*)m_h264data, outLen, pts_time);
						}
					}
					else
					{
						//DecodeVideo((BYTE*)data, nRead, pts_time);
					}

				}
			}

		}

		if(nRead < 0)
		{
			TRACE("RTMP_Read return: %d \n", nRead);
			break;
		}
		else if(nRead == 0)
		{
			Sleep(10);
			continue;
		}

	}//while


	free(data);

	if(rtmp != NULL)
	{
		RTMP_Close(rtmp);
		RTMP_Free(rtmp);
		rtmp = NULL;
	}
}

注意上面的代碼會自動插入SPS和PPS到每個I幀前面，你最好先考慮一下你的代碼是否需要這個功能。只要視頻開頭有一個I幀攜帶了PPS+PPS，視頻也是能播放的。如果不需要每個I幀前面都插入SPS+PPS（第一個I幀除外），可以屏蔽掉那些代碼。

如想了解更多librtmp使用上的問題，可參考我的一篇博文：使用librtmp接收要注意的問題