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wordpress加菜单,seo教学网站,如何做销售网站,东莞智通人才网首页librtmp是一个RTMP的开源库，很多地方用它来做推流、拉流。它是RTMPDump开源软件里的一部分，librtmp的下载地址：RTMPDump，目前最新版是V2.3。本文重点介绍librtmp优化。 1、调整网络输出块大小。 RTMP_Connect0函数中LibRTMP是关…

librtmp是一个RTMP的开源库，很多地方用它来做推流、拉流。它是RTMPDump开源软件里的一部分，librtmp的下载地址：RTMPDump，目前最新版是V2.3。本文重点介绍librtmp优化。

1、调整网络输出块大小。

RTMP_Connect0函数中LibRTMP是关闭了Nagle算法这个TCP选项的，为了实时性这样做是好的，但是要注意到LibRTMP的结构体RTMP的成员是有m_outChunkSize，并且在RTMP_Init函数中被初始化了默认值128，然后整个LibRTMP代码没有改变m_outChunkSize的接口函数，内部也没有改变m_outChunkSize的实现逻辑，也没有发送改变块大小的消息给流媒体服务器的代码逻辑，关闭Nagle加如此小的块大小会导致很多小包，而以太网的MTU是1500，这样如果用在播放客户端由于主要是接收媒体流到也没有什么，但是如果用在发布媒体流的推流客户端网络效率就太低了，并且IP小包太多还会引起流媒体的服务器软中断升高，导致内核占用的CPU过高。m_outChunkSize在发送给流媒体服务器消息会用于分块，所以从这个方面来说LibRTMP还是部分支持改变块大小的，这部分逻辑实现不需要任何改变。

调整输出块大小的函数

static int
ChangeChunkSize(RTMP *r,int outChunkSize)
{
RTMPPacket packet;
char pbuf[RTMP_MAX_HEADER_SIZE + 4];packet.m_nBytesRead = 0;
packet.m_body = pbuf + RTMP_MAX_HEADER_SIZE;packet.m_packetType = RTMP_PACKET_TYPE_CHUNK_SIZE;
packet.m_nChannel = 0x04; 
packet.m_headerType = RTMP_PACKET_SIZE_LARGE;
packet.m_nTimeStamp = 0;
packet.m_nInfoField2 = 0;
packet.m_hasAbsTimestamp = 0;
packet.m_nBodySize = 4;
r->m_outChunkSize = outChunkSize;r->m_outChunkSize = htonl(r->m_outChunkSize);memcpy(packet.m_body, &r->m_outChunkSize, 4);r->m_outChunkSize = ntohl(r->m_outChunkSize);return RTMP_SendPacket(r, &packet, TRUE);
}

注1：RTMP协议的消息类型01（RTMP_PACKET_TYPE_CHUNK_SIZE宏的值）就是用于改变输出块大小的消息类型，结合MTU

注2：outChunkSize大小可以选择1500-20(IP头)-20（TCP头）=1460，考虑到IP头、TCP头有扩展选项，加之PPPoE，为保证起见可选为1360，也可以设为大于MTU的其它值，不过这样的话就会出现IP分片了，也不是好习惯。

注3：每当调用本函数后就顺便修改了RTMP的成员变量m_outChunkSize，以保持与服务器收到的一致。

调整网络输出块大小的函数的时机

随时可以调整，只不过在调用ChangeChunkSize函数后，要注意到这个函数内部已经改变了RTMP的成员变量m_outChunkSize，这样在调用这个函数之后的所有发给流媒体服务器的消息要以这个块大小来分块，由于TCP的有序性，服务器在收到该改变块大小的消息后也会以此块大小来解析后序的所有消息，由于播放客户端主要是拉流，播放端需要传给服务器的数据不多，可以不修改，基于此可以在收到connect的响应后的处理逻辑中调用ChangeChunkSize函数，具体如下（HandleInvoke函数中部分代码）：

if (r->Link.protocol & RTMP_FEATURE_WRITE)
{
ChangeChunkSize(r, 1360);//若不改拉流时的输出块大小在这里调用ChangeChunkSize
SendReleaseStream(r);
SendFCPublish(r);
}
else
{
RTMP_SendServerBW(r);
RTMP_SendCtrl(r, 3, 0, 300);
}

2、AMF_GetProp函数bug

librtmp\amf.c文件里有个函数（大概是1124行）：

AMFObjectProperty *
AMF_GetProp(AMFObject *obj, const AVal *name, int nIndex)
{if (nIndex >= 0){if (nIndex <= obj->o_num) //这里有问题return &obj->o_props[nIndex];}else{int n;for (n = 0; n < obj->o_num; n++){if (AVMATCH(&obj->o_props[n].p_name, name))return &obj->o_props[n];}}return (AMFObjectProperty *)&AMFProp_Invalid;
}

3、librtmp发送阻塞

在librtmp的RTMP_Connect0( )中加入了send超时设置。

/* set timeout */
{struct timeval timeout;timeout.tv_sec = 5;timeout.tv_usec = 0;SET_RCVTIMEO(tv, r->Link.timeout);if (setsockopt(r->m_sb.sb_socket, SOL_SOCKET, SO_RCVTIMEO, (char *)&tv, sizeof(tv))){RTMP_Log(RTMP_LOGERROR, "%s, Setting socket timeout to %ds failed!", __FUNCTION__, r->Link.timeout);}if(setsockopt(r->m_sb.sb_socket, SOL_SOCKET, SO_SNDTIMEO, (char*)&timeout, sizeof(timeout)) == -1){RTMP_Log(RTMP_LOGERROR, "%s, Setting socket timeout to %ds failed!", __FUNCTION__, timeout.tv_sec);}else{RTMP_Log(RTMP_LOGDEBUG, "%s, Setting socket timeout to %ds success!", __FUNCTION__, timeout.tv_sec);}
}

4、librtmp 推流网络断开导致整个应用程序crash的错误，崩溃堆栈出现在WriteN函数中。
解决方法，修改rtmp.c文件中WriteN函数：

static int
WriteN(RTMP *r, const char *buffer, int n)
{const char *ptr = buffer;
#ifdef CRYPTOchar *encrypted = 0;char buf[RTMP_BUFFER_CACHE_SIZE];if (r->Link.rc4keyOut){if (n > sizeof(buf))encrypted = (char *)malloc(n);elseencrypted = (char *)buf;ptr = encrypted;RC4_encrypt2(r->Link.rc4keyOut, n, buffer, ptr);}
#endifwhile (n > 0){int nBytes;if (r->Link.protocol & RTMP_FEATURE_HTTP)nBytes = HTTP_Post(r, RTMPT_SEND, ptr, n);elsenBytes = RTMPSockBuf_Send(&r->m_sb, ptr, n);/*RTMP_Log(RTMP_LOGDEBUG, "%s: %d\n", __FUNCTION__, nBytes); */if (nBytes < 0){int sockerr = GetSockError();RTMP_Log(RTMP_LOGERROR, "%s, RTMP send error %d (%d bytes)", __FUNCTION__,sockerr, n);if (sockerr == EINTR && !RTMP_ctrlC)continue;if(WSAECONNABORTED  == sockerr || WSAECONNRESET  == sockerr){//连接被对方断开RTMP_CloseNoSendAnyData(r);}else{RTMP_Close(r);}n = 1;break;}if (nBytes == 0)break;n -= nBytes;ptr += nBytes;}#ifdef CRYPTOif (encrypted && encrypted != buf)free(encrypted);
#endifreturn n == 0;
}

//新增RTMP_CloseNoSendAnyData函数

void
RTMP_CloseNoSendAnyData(RTMP *r)
{int i;if (RTMP_IsConnected(r)){if (r->m_stream_id > 0){if (r->m_clientID.av_val){HTTP_Post(r, RTMPT_CLOSE, "", 1);free(r->m_clientID.av_val);r->m_clientID.av_val = NULL;r->m_clientID.av_len = 0;}RTMPSockBuf_Close(&r->m_sb);}r->m_stream_id = -1;r->m_sb.sb_socket = -1;r->m_nBWCheckCounter = 0;r->m_nBytesIn = 0;r->m_nBytesInSent = 0;if (r->m_read.flags & RTMP_READ_HEADER) {free(r->m_read.buf);r->m_read.buf = NULL;}r->m_read.dataType = 0;r->m_read.flags = 0;r->m_read.status = 0;r->m_read.nResumeTS = 0;r->m_read.nIgnoredFrameCounter = 0;r->m_read.nIgnoredFlvFrameCounter = 0;r->m_write.m_nBytesRead = 0;RTMPPacket_Free(&r->m_write);for (i = 0; i < RTMP_CHANNELS; i++){if (r->m_vecChannelsIn[i]){RTMPPacket_Free(r->m_vecChannelsIn[i]);free(r->m_vecChannelsIn[i]);r->m_vecChannelsIn[i] = NULL;}if (r->m_vecChannelsOut[i]){free(r->m_vecChannelsOut[i]);r->m_vecChannelsOut[i] = NULL;}}AV_clear(r->m_methodCalls, r->m_numCalls);r->m_methodCalls = NULL;r->m_numCalls = 0;r->m_numInvokes = 0;r->m_bPlaying = FALSE;r->m_sb.sb_size = 0;r->m_msgCounter = 0;r->m_resplen = 0;r->m_unackd = 0;free(r->Link.playpath0.av_val);r->Link.playpath0.av_val = NULL;if (r->Link.lFlags & RTMP_LF_FTCU){free(r->Link.tcUrl.av_val);r->Link.tcUrl.av_val = NULL;r->Link.lFlags ^= RTMP_LF_FTCU;}#ifdef CRYPTOif (r->Link.dh){MDH_free(r->Link.dh);r->Link.dh = NULL;}if (r->Link.rc4keyIn){RC4_free(r->Link.rc4keyIn);r->Link.rc4keyIn = NULL;}if (r->Link.rc4keyOut){RC4_free(r->Link.rc4keyOut);r->Link.rc4keyOut = NULL;}
#endif
}

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