函数调用结构图
RTMPDump (libRTMP)的整体的函数调用结构图如下图所示。
详细分析
已经连续写了一系列的博客了,其实大部分内容都是去年搞RTMP研究的时候积累的经验,回顾一下过去的知识,其实RTMPdump(libRTMP)主要的功能也都分析的差不多了,现在感觉还需要一些查漏补缺。主要就是它是如何处理各种消息(Message)的这方面还没有研究的特明白,在此需要详细研究一下。再来看一下RTMPdump(libRTMP)的“灵魂”函数RTMP_ClientPacket(),主要完成了各种消息的处理。
- //处理接收到的数据
- int
- RTMP_ClientPacket(RTMP *r, RTMPPacket *packet)
- {
- int bHasMediaPacket = 0;
- switch (packet->m_packetType)
- {
- //RTMP消息类型ID=1,设置块大小
- case 0x01:
- /* chunk size */
- //----------------
- r->dlg->AppendCInfo("处理收到的数据。消息 Set Chunk Size (typeID=1)。");
- //-----------------------------
- RTMP_LogPrintf("处理消息 Set Chunk Size (typeID=1)\n");
- HandleChangeChunkSize(r, packet);
- break;
- //RTMP消息类型ID=3,致谢
- case 0x03:
- /* bytes read report */
- RTMP_Log(RTMP_LOGDEBUG, "%s, received: bytes read report", __FUNCTION__);
- break;
- //RTMP消息类型ID=4,用户控制
- case 0x04:
- /* ctrl */
- //----------------
- r->dlg->AppendCInfo("处理收到的数据。消息 User Control (typeID=4)。");
- //-----------------------------
- RTMP_LogPrintf("处理消息 User Control (typeID=4)\n");
- HandleCtrl(r, packet);
- break;
- //RTMP消息类型ID=5
- case 0x05:
- /* server bw */
- //----------------
- r->dlg->AppendCInfo("处理收到的数据。消息 Window Acknowledgement Size (typeID=5)。");
- //-----------------------------
- RTMP_LogPrintf("处理消息 Window Acknowledgement Size (typeID=5)\n");
- HandleServerBW(r, packet);
- break;
- //RTMP消息类型ID=6
- case 0x06:
- /* client bw */
- //----------------
- r->dlg->AppendCInfo("处理收到的数据。消息 Set Peer Bandwidth (typeID=6)。");
- //-----------------------------
- RTMP_LogPrintf("处理消息 Set Peer Bandwidth (typeID=6)\n");
- HandleClientBW(r, packet);
- break;
- //RTMP消息类型ID=8,音频数据
- case 0x08:
- /* audio data */
- /*RTMP_Log(RTMP_LOGDEBUG, "%s, received: audio %lu bytes", __FUNCTION__, packet.m_nBodySize); */
- HandleAudio(r, packet);
- bHasMediaPacket = 1;
- if (!r->m_mediaChannel)
- r->m_mediaChannel = packet->m_nChannel;
- if (!r->m_pausing)
- r->m_mediaStamp = packet->m_nTimeStamp;
- break;
- //RTMP消息类型ID=9,视频数据
- case 0x09:
- /* video data */
- /*RTMP_Log(RTMP_LOGDEBUG, "%s, received: video %lu bytes", __FUNCTION__, packet.m_nBodySize); */
- HandleVideo(r, packet);
- bHasMediaPacket = 1;
- if (!r->m_mediaChannel)
- r->m_mediaChannel = packet->m_nChannel;
- if (!r->m_pausing)
- r->m_mediaStamp = packet->m_nTimeStamp;
- break;
- //RTMP消息类型ID=15,AMF3编码,忽略
- case 0x0F: /* flex stream send */
- RTMP_Log(RTMP_LOGDEBUG,
- "%s, flex stream send, size %lu bytes, not supported, ignoring",
- __FUNCTION__, packet->m_nBodySize);
- break;
- //RTMP消息类型ID=16,AMF3编码,忽略
- case 0x10: /* flex shared object */
- RTMP_Log(RTMP_LOGDEBUG,
- "%s, flex shared object, size %lu bytes, not supported, ignoring",
- __FUNCTION__, packet->m_nBodySize);
- break;
- //RTMP消息类型ID=17,AMF3编码,忽略
- case 0x11: /* flex message */
- {
- RTMP_Log(RTMP_LOGDEBUG,
- "%s, flex message, size %lu bytes, not fully supported",
- __FUNCTION__, packet->m_nBodySize);
- /*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
- /* some DEBUG code */
- #if 0
- RTMP_LIB_AMFObject obj;
- int nRes = obj.Decode(packet.m_body+1, packet.m_nBodySize-1);
- if(nRes < 0) {
- RTMP_Log(RTMP_LOGERROR, "%s, error decoding AMF3 packet", __FUNCTION__);
- /*return; */
- }
- obj.Dump();
- #endif
- if (HandleInvoke(r, packet->m_body + 1, packet->m_nBodySize - 1) == 1)
- bHasMediaPacket = 2;
- break;
- }
- //RTMP消息类型ID=18,AMF0编码,数据消息
- case 0x12:
- /* metadata (notify) */
- RTMP_Log(RTMP_LOGDEBUG, "%s, received: notify %lu bytes", __FUNCTION__,
- packet->m_nBodySize);
- //处理元数据,暂时注释
- /*
- if (HandleMetadata(r, packet->m_body, packet->m_nBodySize))
- bHasMediaPacket = 1;
- break;
- */
- //RTMP消息类型ID=19,AMF0编码,忽略
- case 0x13:
- RTMP_Log(RTMP_LOGDEBUG, "%s, shared object, not supported, ignoring",
- __FUNCTION__);
- break;
- //RTMP消息类型ID=20,AMF0编码,命令消息
- //处理命令消息!
- case 0x14:
- //----------------
- r->dlg->AppendCInfo("处理收到的数据。消息 命令 (AMF0编码) (typeID=20)。");
- //-----------------------------
- /* invoke */
- RTMP_Log(RTMP_LOGDEBUG, "%s, received: invoke %lu bytes", __FUNCTION__,
- packet->m_nBodySize);
- RTMP_LogPrintf("处理命令消息 (typeID=20,AMF0编码)\n");
- /*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
- if (HandleInvoke(r, packet->m_body, packet->m_nBodySize) == 1)
- bHasMediaPacket = 2;
- break;
- //RTMP消息类型ID=22
- case 0x16:
- {
- /* go through FLV packets and handle metadata packets */
- unsigned int pos = 0;
- uint32_t nTimeStamp = packet->m_nTimeStamp;
- while (pos + 11 < packet->m_nBodySize)
- {
- uint32_t dataSize = AMF_DecodeInt24(packet->m_body + pos + 1); /* size without header (11) and prevTagSize (4) */
- if (pos + 11 + dataSize + 4 > packet->m_nBodySize)
- {
- RTMP_Log(RTMP_LOGWARNING, "Stream corrupt?!");
- break;
- }
- if (packet->m_body[pos] == 0x12)
- {
- HandleMetadata(r, packet->m_body + pos + 11, dataSize);
- }
- else if (packet->m_body[pos] == 8 || packet->m_body[pos] == 9)
- {
- nTimeStamp = AMF_DecodeInt24(packet->m_body + pos + 4);
- nTimeStamp |= (packet->m_body[pos + 7] << 24);
- }
- pos += (11 + dataSize + 4);
- }
- if (!r->m_pausing)
- r->m_mediaStamp = nTimeStamp;
- /* FLV tag(s) */
- /*RTMP_Log(RTMP_LOGDEBUG, "%s, received: FLV tag(s) %lu bytes", __FUNCTION__, packet.m_nBodySize); */
- bHasMediaPacket = 1;
- break;
- }
- default:
- RTMP_Log(RTMP_LOGDEBUG, "%s, unknown packet type received: 0x%02x", __FUNCTION__,
- packet->m_packetType);
- #ifdef _DEBUG
- RTMP_LogHex(RTMP_LOGDEBUG, (const uint8_t *)packet->m_body, packet->m_nBodySize);
- #endif
- }
- return bHasMediaPacket;
- }
前文已经分析过当消息类型ID为0x14(20)的时候,即AMF0编码的命令消息的时候,会调用HandleInvoke()进行处理。
这里就不再对这种类型ID的消息进行分析了,分析一下其他类型的消息,毕竟从发起一个RTMP连接到接收视音频数据这个过程中是要处理很多消息的。
参考:RTMP流媒体播放过程
下面我们按照消息ID从小到大的顺序,看看接收到的各种消息都是如何处理的。
消息类型ID是0x01的消息功能是“设置块(Chunk)大小”,处理函数是HandleChangeChunkSize(),可见函数内容很简单。
- static void
- HandleChangeChunkSize(RTMP *r, const RTMPPacket *packet)
- {
- if (packet->m_nBodySize >= 4)
- {
- r->m_inChunkSize = AMF_DecodeInt32(packet->m_body);
- RTMP_Log(RTMP_LOGDEBUG, "%s, received: chunk size change to %d", __FUNCTION__,
- r->m_inChunkSize);
- }
- }
消息类型ID是0x03的消息功能是“致谢”,没有处理函数。
消息类型ID是0x04的消息功能是“用户控制(UserControl)”,处理函数是HandleCtrl(),这类的消息出现的频率非常高,函数体如下所示。具体用户控制消息的作用这里就不多说了,有相应的文档可以参考。
注:该函数中间有一段很长的英文注释,英语好的大神可以看一看
- //处理用户控制(UserControl)消息。用户控制消息是服务器端发出的。
- static void
- HandleCtrl(RTMP *r, const RTMPPacket *packet)
- {
- short nType = -1;
- unsigned int tmp;
- if (packet->m_body && packet->m_nBodySize >= 2)
- //事件类型(2B)
- nType = AMF_DecodeInt16(packet->m_body);
- RTMP_Log(RTMP_LOGDEBUG, "%s, received ctrl. type: %d, len: %d", __FUNCTION__, nType,
- packet->m_nBodySize);
- /*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
- if (packet->m_nBodySize >= 6)
- {
- //不同事件类型做不同处理
- switch (nType)
- {
- //流开始
- case 0:
- //流ID
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream Begin %d", __FUNCTION__, tmp);
- break;
- //流结束
- case 1:
- //流ID
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream EOF %d", __FUNCTION__, tmp);
- if (r->m_pausing == 1)
- r->m_pausing = 2;
- break;
- //流枯竭
- case 2:
- //流ID
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream Dry %d", __FUNCTION__, tmp);
- break;
- //是录制流
- case 4:
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream IsRecorded %d", __FUNCTION__, tmp);
- break;
- //Ping客户端
- case 6: /* server ping. reply with pong. */
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Ping %d", __FUNCTION__, tmp);
- RTMP_SendCtrl(r, 0x07, tmp, 0);
- break;
- /* FMS 3.5 servers send the following two controls to let the client
- * know when the server has sent a complete buffer. I.e., when the
- * server has sent an amount of data equal to m_nBufferMS in duration.
- * The server meters its output so that data arrives at the client
- * in realtime and no faster.
- *
- * The rtmpdump program tries to set m_nBufferMS as large as
- * possible, to force the server to send data as fast as possible.
- * In practice, the server appears to cap this at about 1 hour's
- * worth of data. After the server has sent a complete buffer, and
- * sends this BufferEmpty message, it will wait until the play
- * duration of that buffer has passed before sending a new buffer.
- * The BufferReady message will be sent when the new buffer starts.
- * (There is no BufferReady message for the very first buffer;
- * presumably the Stream Begin message is sufficient for that
- * purpose.)
- *
- * If the network speed is much faster than the data bitrate, then
- * there may be long delays between the end of one buffer and the
- * start of the next.
- *
- * Since usually the network allows data to be sent at
- * faster than realtime, and rtmpdump wants to download the data
- * as fast as possible, we use this RTMP_LF_BUFX hack: when we
- * get the BufferEmpty message, we send a Pause followed by an
- * Unpause. This causes the server to send the next buffer immediately
- * instead of waiting for the full duration to elapse. (That's
- * also the purpose of the ToggleStream function, which rtmpdump
- * calls if we get a read timeout.)
- *
- * Media player apps don't need this hack since they are just
- * going to play the data in realtime anyway. It also doesn't work
- * for live streams since they obviously can only be sent in
- * realtime. And it's all moot if the network speed is actually
- * slower than the media bitrate.
- */
- case 31:
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream BufferEmpty %d", __FUNCTION__, tmp);
- if (!(r->Link.lFlags & RTMP_LF_BUFX))
- break;
- if (!r->m_pausing)
- {
- r->m_pauseStamp = r->m_channelTimestamp[r->m_mediaChannel];
- RTMP_SendPause(r, TRUE, r->m_pauseStamp);
- r->m_pausing = 1;
- }
- else if (r->m_pausing == 2)
- {
- RTMP_SendPause(r, FALSE, r->m_pauseStamp);
- r->m_pausing = 3;
- }
- break;
- case 32:
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream BufferReady %d", __FUNCTION__, tmp);
- break;
- default:
- tmp = AMF_DecodeInt32(packet->m_body + 2);
- RTMP_Log(RTMP_LOGDEBUG, "%s, Stream xx %d", __FUNCTION__, tmp);
- break;
- }
- }
- if (nType == 0x1A)
- {
- RTMP_Log(RTMP_LOGDEBUG, "%s, SWFVerification ping received: ", __FUNCTION__);
- if (packet->m_nBodySize > 2 && packet->m_body[2] > 0x01)
- {
- RTMP_Log(RTMP_LOGERROR,
- "%s: SWFVerification Type %d request not supported! Patches welcome...",
- __FUNCTION__, packet->m_body[2]);
- }
- #ifdef CRYPTO
- /*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
- /* respond with HMAC SHA256 of decompressed SWF, key is the 30byte player key, also the last 30 bytes of the server handshake are applied */
- else if (r->Link.SWFSize)
- {
- RTMP_SendCtrl(r, 0x1B, 0, 0);
- }
- else
- {
- RTMP_Log(RTMP_LOGERROR,
- "%s: Ignoring SWFVerification request, use --swfVfy!",
- __FUNCTION__);
- }
- #else
- RTMP_Log(RTMP_LOGERROR,
- "%s: Ignoring SWFVerification request, no CRYPTO support!",
- __FUNCTION__);
- #endif
- }
- }
消息类型ID是0x05的消息功能是“窗口致谢大小(Window Acknowledgement Size,翻译的真是挺别扭)”,处理函数是HandleServerBW()。在这里注意一下,该消息在Adobe官方公开的文档中叫“Window Acknowledgement Size”,但是在Adobe公开协议规范之前,破解RTMP协议的组织一直管该协议叫“ServerBW”,只是个称呼,倒是也无所谓~处理代码很简单:
- static void
- HandleServerBW(RTMP *r, const RTMPPacket *packet)
- {
- r->m_nServerBW = AMF_DecodeInt32(packet->m_body);
- RTMP_Log(RTMP_LOGDEBUG, "%s: server BW = %d", __FUNCTION__, r->m_nServerBW);
- }
消息类型ID是0x06的消息功能是“设置对等端带宽(Set Peer Bandwidth)”,处理函数是HandleClientBW()。与上一种消息一样,该消息在Adobe官方公开的文档中叫“Set Peer Bandwidth”,但是在Adobe公开协议规范之前,破解RTMP协议的组织一直管该协议叫“ClientBW”。处理函数也不复杂:
- static void
- HandleClientBW(RTMP *r, const RTMPPacket *packet)
- {
- r->m_nClientBW = AMF_DecodeInt32(packet->m_body);
- if (packet->m_nBodySize > 4)
- r->m_nClientBW2 = packet->m_body[4];
- else
- r->m_nClientBW2 = -1;
- RTMP_Log(RTMP_LOGDEBUG, "%s: client BW = %d %d", __FUNCTION__, r->m_nClientBW,
- r->m_nClientBW2);
- }
消息类型ID是0x08的消息用于传输音频数据,在这里不处理。
消息类型ID是0x09的消息用于传输音频数据,在这里不处理。
消息类型ID是0x0F-11的消息用于传输AMF3编码的命令。
消息类型ID是0x12-14的消息用于传输AMF0编码的命令。
注:消息类型ID是0x14的消息很重要,用于传输AMF0编码的命令,已经做过分析。
rtmpdump源代码(Linux):http://download.csdn.net/detail/leixiaohua1020/6376561
rtmpdump源代码(VC 2005 工程):http://download.csdn.net/detail/leixiaohua1020/6563163