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RTMP协议相关：
【流媒体】RTMP协议概述
【流媒体】RTMP协议的数据格式
【流媒体】RTMP协议的消息类型
【流媒体】RTMPDump—主流程简单分析
【流媒体】RTMPDump—RTMP_Connect函数（握手、网络连接）

参考雷博的系列文章（可以从一篇链接到其他文章）：
RTMPdump 源代码分析 1： main()函数

1. RTMP_ConnectStream函数

RTMP_ConnectStream()的作用是建立流连接，先回顾一下RTMP标准文档当中是如何进行流的连接的，以client向server发送play命令为例，流程图如下所示。从流程中看，在进行了握手和RTMP连接之后，由client向server发送一个命令 “createStream”，随后由server返回一个命令消息 _result，表示对这个 “createStream” 的反馈。随后进行play命令

RTMP实现 “createStream” 这条命令的函数为RTMP_ConnectStream()，这个函数的实现比较简单，主要有两个步骤：
（1）读取packet（RTMP_ReadPacket）
（2）解析packet（RTMP_ClientPacket）

int
RTMP_ConnectStream(RTMP * r, int seekTime)
{RTMPPacket packet = { 0 };/* seekTime was already set by SetupStream / SetupURL.* This is only needed by ReconnectStream.*/if (seekTime > 0)r->Link.seekTime = seekTime;r->m_mediaChannel = 0;// 1.读取packetwhile (!r->m_bPlaying && RTMP_IsConnected(r) && RTMP_ReadPacket(r, &packet)){if (RTMPPacket_IsReady(&packet)){if (!packet.m_nBodySize)continue;if ((packet.m_packetType == RTMP_PACKET_TYPE_AUDIO) ||(packet.m_packetType == RTMP_PACKET_TYPE_VIDEO) ||(packet.m_packetType == RTMP_PACKET_TYPE_INFO)){RTMP_Log(RTMP_LOGWARNING, "Received FLV packet before play()! Ignoring.");RTMPPacket_Free(&packet);continue;}// 2.解析packetRTMP_ClientPacket(r, &packet);RTMPPacket_Free(&packet);}}return r->m_bPlaying;
}

1.1 读取packet（RTMP_ReadPacket）

RTMP_ReadPacket()函数的实现如下

int
RTMP_ReadPacket(RTMP * r, RTMPPacket * packet)
{uint8_t hbuf[RTMP_MAX_HEADER_SIZE] = { 0 };char* header = (char*)hbuf;int nSize, hSize, nToRead, nChunk;int didAlloc = FALSE;int extendedTimestamp;RTMP_Log(RTMP_LOGDEBUG2, "%s: fd=%d", __FUNCTION__, r->m_sb.sb_socket);// 读取packet的第1个字节，即basic headerif (ReadN(r, (char*)hbuf, 1) == 0){RTMP_Log(RTMP_LOGERROR, "%s, failed to read RTMP packet header", __FUNCTION__);return FALSE;}// fmtpacket->m_headerType = (hbuf[0] & 0xc0) >> 6;// chunk stream id (cs_id)packet->m_nChannel = (hbuf[0] & 0x3f);header++;// 第1字节后6位为0，说明basic header size为2字节if (packet->m_nChannel == 0){if (ReadN(r, (char*)& hbuf[1], 1) != 1){RTMP_Log(RTMP_LOGERROR, "%s, failed to read RTMP packet header 2nd byte",__FUNCTION__);return FALSE;}packet->m_nChannel = hbuf[1];packet->m_nChannel += 64;header++;}else if (packet->m_nChannel == 1) // 第1字节后6位为1，说明basic header size为3字节{int tmp;if (ReadN(r, (char*)& hbuf[1], 2) != 2){RTMP_Log(RTMP_LOGERROR, "%s, failed to read RTMP packet header 3nd byte",__FUNCTION__);return FALSE;}tmp = (hbuf[2] << 8) + hbuf[1];packet->m_nChannel = tmp + 64; // 计算cs_idRTMP_Log(RTMP_LOGDEBUG, "%s, m_nChannel: %0x", __FUNCTION__, packet->m_nChannel);header += 2;}// 计算message header sizenSize = packetSize[packet->m_headerType];// cs_id大于已分配的，需要进行重新分配if (packet->m_nChannel >= r->m_channelsAllocatedIn){int n = packet->m_nChannel + 10;int* timestamp = realloc(r->m_channelTimestamp, sizeof(int) * n);RTMPPacket** packets = realloc(r->m_vecChannelsIn, sizeof(RTMPPacket*) * n);if (!timestamp)free(r->m_channelTimestamp);if (!packets)free(r->m_vecChannelsIn);r->m_channelTimestamp = timestamp;r->m_vecChannelsIn = packets;if (!timestamp || !packets) {r->m_channelsAllocatedIn = 0;return FALSE;}memset(r->m_channelTimestamp + r->m_channelsAllocatedIn, 0, sizeof(int) * (n - r->m_channelsAllocatedIn));memset(r->m_vecChannelsIn + r->m_channelsAllocatedIn, 0, sizeof(RTMPPacket*) * (n - r->m_channelsAllocatedIn));r->m_channelsAllocatedIn = n;}// 如果获取到整个header信息，timestamp是绝对值if (nSize == RTMP_LARGE_HEADER_SIZE)	/* if we get a full header the timestamp is absolute */packet->m_hasAbsTimestamp = TRUE;else if (nSize < RTMP_LARGE_HEADER_SIZE){				/* using values from the last message of this channel */if (r->m_vecChannelsIn[packet->m_nChannel])memcpy(packet, r->m_vecChannelsIn[packet->m_nChannel],sizeof(RTMPPacket));}nSize--; // {11, 7, 3, 0}// 读取RTMP的message headerif (nSize > 0 && ReadN(r, header, nSize) != nSize){RTMP_Log(RTMP_LOGERROR, "%s, failed to read RTMP packet header. type: %x",__FUNCTION__, (unsigned int)hbuf[0]);return FALSE;}hSize = nSize + (header - (char*)hbuf);// 下面根据不同格式的message header来解析字段if (nSize >= 3){// 解析timestampacket->m_nTimeStamp = AMF_DecodeInt24(header);/*RTMP_Log(RTMP_LOGDEBUG, "%s, reading RTMP packet chunk on channel %x, headersz %i, timestamp %i, abs timestamp %i", __FUNCTION__, packet.m_nChannel, nSize, packet.m_nTimeStamp, packet.m_hasAbsTimestamp); */if (nSize >= 6){// 解析message lengthpacket->m_nBodySize = AMF_DecodeInt24(header + 3);packet->m_nBytesRead = 0;if (nSize > 6){// 解析message type idpacket->m_packetType = header[6];if (nSize == 11) // 解析message stream idpacket->m_nInfoField2 = DecodeInt32LE(header + 7);}}}// 检查是否有扩展时间戳，如果有则读取extendedTimestamp = packet->m_nTimeStamp == 0xffffff;if (extendedTimestamp){if (ReadN(r, header + nSize, 4) != 4){RTMP_Log(RTMP_LOGERROR, "%s, failed to read extended timestamp",__FUNCTION__);return FALSE;}packet->m_nTimeStamp = AMF_DecodeInt32(header + nSize);hSize += 4;}RTMP_LogHexString(RTMP_LOGDEBUG2, (uint8_t*)hbuf, hSize);if (packet->m_nBodySize > 0 && packet->m_body == NULL){if (!RTMPPacket_Alloc(packet, packet->m_nBodySize)){RTMP_Log(RTMP_LOGDEBUG, "%s, failed to allocate packet", __FUNCTION__);return FALSE;}didAlloc = TRUE;packet->m_headerType = (hbuf[0] & 0xc0) >> 6;}// 剩余需要读取的字节数nToRead = packet->m_nBodySize - packet->m_nBytesRead;nChunk = r->m_inChunkSize;if (nToRead < nChunk)nChunk = nToRead;// 是否需要将原始chunk拷贝/* Does the caller want the raw chunk? */if (packet->m_chunk){packet->m_chunk->c_headerSize = hSize;memcpy(packet->m_chunk->c_header, hbuf, hSize);packet->m_chunk->c_chunk = packet->m_body + packet->m_nBytesRead;packet->m_chunk->c_chunkSize = nChunk;}// 获取body的信息if (ReadN(r, packet->m_body + packet->m_nBytesRead, nChunk) != nChunk){RTMP_Log(RTMP_LOGERROR, "%s, failed to read RTMP packet body. len: %u",__FUNCTION__, packet->m_nBodySize);return FALSE;}RTMP_LogHexString(RTMP_LOGDEBUG2, (uint8_t*)packet->m_body + packet->m_nBytesRead, nChunk);packet->m_nBytesRead += nChunk;// 保留该数据包作为该通道上其他数据包的参考/* keep the packet as ref for other packets on this channel */if (!r->m_vecChannelsIn[packet->m_nChannel])r->m_vecChannelsIn[packet->m_nChannel] = malloc(sizeof(RTMPPacket));memcpy(r->m_vecChannelsIn[packet->m_nChannel], packet, sizeof(RTMPPacket));if (extendedTimestamp){r->m_vecChannelsIn[packet->m_nChannel]->m_nTimeStamp = 0xffffff;}// 当前packet所有信息都读取到了，拷贝时间戳并且将当前packet重置if (RTMPPacket_IsReady(packet)){/* make packet's timestamp absolute */if (!packet->m_hasAbsTimestamp)packet->m_nTimeStamp += r->m_channelTimestamp[packet->m_nChannel];	/* timestamps seem to be always relative!! */r->m_channelTimestamp[packet->m_nChannel] = packet->m_nTimeStamp;/* reset the data from the stored packet. we keep the header since we may use it later if a new packet for this channel *//* arrives and requests to re-use some info (small packet header) */r->m_vecChannelsIn[packet->m_nChannel]->m_body = NULL;r->m_vecChannelsIn[packet->m_nChannel]->m_nBytesRead = 0;r->m_vecChannelsIn[packet->m_nChannel]->m_hasAbsTimestamp = FALSE;	/* can only be false if we reuse header */}else{packet->m_body = NULL;	/* so it won't be erased on free */}return TRUE;
}

1.2 解析packet（RTMP_ClientPacket）

该函数的主要作用是解析接收到的数据报，根据数据报的类型进行相应的操作。这些操作包括：
（1）RTMP_PACKET_TYPE_CHUNK_SIZE
设置chunk size

（2）RTMP_PACKET_TYPE_BYTES_READ_REPORT
应答消息，表示已经接收到了传输过来的数据报，返回的是已读取的比特数

（3）RTMP_PACKET_TYPE_CONTROL
用户控制信息

（4）RTMP_PACKET_TYPE_SERVER_BW
设置服务器带宽

（5）RTMP_PACKET_TYPE_CLIENT_BW
设置用户带宽

（6）RTMP_PACKET_TYPE_AUDIO
音频数据

（7）RTMP_PACKET_TYPE_VIDEO
视频数据

（8）RTMP_PACKET_TYPE_FLEX_STREAM_SEND
数据消息，发送元数据或任何用户数据到对端，AMF3 = 15

（9）RTMP_PACKET_TYPE_FLEX_SHARED_OBJECT
共享对象消息, AMF3 = 16

（10）RTMP_PACKET_TYPE_FLEX_MESSAGE
传递AMF编码命令，AMF3 = 17

（11）RTMP_PACKET_TYPE_INFO
数据消息，发送元数据或任何用户数据到对端，AFM0 = 18

（12）RTMP_PACKET_TYPE_SHARED_OBJECT
共享对象消息，AMF0 = 19

（13）RTMP_PACKET_TYPE_INVOKE
传递AMF编码命令，AMF0 = 20

（14）RTMP_PACKET_TYPE_FLASH_VIDEO
聚合消息，一个单一的包含一系列的RTMP子消息的消息；FLV视频

int
RTMP_ClientPacket(RTMP * r, RTMPPacket * packet)
{int bHasMediaPacket = 0;switch (packet->m_packetType){case RTMP_PACKET_TYPE_CHUNK_SIZE:	// 设置chunk size/* chunk size */HandleChangeChunkSize(r, packet);break;case RTMP_PACKET_TYPE_BYTES_READ_REPORT:	// 应答消息，表示已经接收到了传输过来的数据报，返回的是已读取的比特数/* bytes read report */RTMP_Log(RTMP_LOGDEBUG, "%s, received: bytes read report", __FUNCTION__);break;case RTMP_PACKET_TYPE_CONTROL:	// 控制命令/* ctrl */HandleCtrl(r, packet);break;case RTMP_PACKET_TYPE_SERVER_BW:	// 设置服务器带宽/* server bw */HandleServerBW(r, packet);break;case RTMP_PACKET_TYPE_CLIENT_BW:	// 设置用户带宽/* client bw */HandleClientBW(r, packet);break;case RTMP_PACKET_TYPE_AUDIO:		// 音频数据/* 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;case RTMP_PACKET_TYPE_VIDEO:		// 视频数据/* 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;case RTMP_PACKET_TYPE_FLEX_STREAM_SEND:	// 数据消息，发送元数据或任何用户数据到对端，AMF3 = 15/* flex stream send */RTMP_Log(RTMP_LOGDEBUG,"%s, flex stream send, size %u bytes, not supported, ignoring",__FUNCTION__, packet->m_nBodySize);break;case RTMP_PACKET_TYPE_FLEX_SHARED_OBJECT:	// 共享对象消息, AMF3 = 16/* flex shared object */RTMP_Log(RTMP_LOGDEBUG,"%s, flex shared object, size %u bytes, not supported, ignoring",__FUNCTION__, packet->m_nBodySize);break;case RTMP_PACKET_TYPE_FLEX_MESSAGE:		// 传递AMF编码命令，AMF3 = 17/* flex message */{RTMP_Log(RTMP_LOGDEBUG,"%s, flex message, size %u bytes, not fully supported",__FUNCTION__, packet->m_nBodySize);/*RTMP_LogHex(packet.m_body, packet.m_nBodySize); *//* some DEBUG code */
#if 0RTMP_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();
#endifif (HandleInvoke(r, packet->m_body + 1, packet->m_nBodySize - 1) == 1)bHasMediaPacket = 2;break;}case RTMP_PACKET_TYPE_INFO:	// 数据消息，发送元数据或任何用户数据到对端，AFM0 = 18/* metadata (notify) */RTMP_Log(RTMP_LOGDEBUG, "%s, received: notify %u bytes", __FUNCTION__,packet->m_nBodySize);if (HandleMetadata(r, packet->m_body, packet->m_nBodySize))bHasMediaPacket = 1;break;case RTMP_PACKET_TYPE_SHARED_OBJECT:	// 共享对象消息, AMF3 = 16RTMP_Log(RTMP_LOGDEBUG, "%s, shared object, not supported, ignoring",__FUNCTION__);break;case RTMP_PACKET_TYPE_INVOKE:	// 传递AMF编码命令，AMF0 = 20/* invoke */RTMP_Log(RTMP_LOGDEBUG, "%s, received: invoke %u bytes", __FUNCTION__,packet->m_nBodySize);/*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */if (HandleInvoke(r, packet->m_body, packet->m_nBodySize) == 1)bHasMediaPacket = 2;break;case RTMP_PACKET_TYPE_FLASH_VIDEO:	// 聚合消息，一个单一的包含一系列的RTMP子消息的消息{// FLV视频现在使用量比较少，这里就不分析了/* 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 _DEBUGRTMP_LogHex(RTMP_LOGDEBUG, packet->m_body, packet->m_nBodySize);
#endif}return bHasMediaPacket;
}

1.2.1 设置Chunk Size（HandleChangeChunkSize）

static void
HandleChangeChunkSize(RTMP * r, const RTMPPacket * packet)
{if (packet->m_nBodySize >= 4){// 解码4字节AMF编码的信息r->m_inChunkSize = AMF_DecodeInt32(packet->m_body);RTMP_Log(RTMP_LOGDEBUG, "%s, received: chunk size change to %d", __FUNCTION__,r->m_inChunkSize);}
}

1.2.2 用户控制信息（HandleCtrl）

static void
HandleCtrl(RTMP * r, const RTMPPacket * packet)
{short nType = -1;unsigned int tmp;if (packet->m_body && packet->m_nBodySize >= 2)nType = AMF_DecodeInt16(packet->m_body); // 前2个字节为Event typeRTMP_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){switch (nType){case 0: // Stream Begintmp = AMF_DecodeInt32(packet->m_body + 2);RTMP_Log(RTMP_LOGDEBUG, "%s, Stream Begin %d", __FUNCTION__, tmp);break;case 1: // Stream EOFtmp = 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;case 2: // Stream Drytmp = AMF_DecodeInt32(packet->m_body + 2);RTMP_Log(RTMP_LOGDEBUG, "%s, Stream Dry %d", __FUNCTION__, tmp);break;case 4: // Stream IsRecordedtmp = AMF_DecodeInt32(packet->m_body + 2);RTMP_Log(RTMP_LOGDEBUG, "%s, Stream IsRecorded %d", __FUNCTION__, tmp);break;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.*//*1. 由于网络通常允许以比实时更快的速度发送数据，并且rtmpdump希望尽可能快地下载数据，因此我们使用RTMP_LF_BUFX hack:当我们获得BufferEmpty消息时，我们发送一个Pause，然后发送一个Unpause这将导致服务器立即发送下一个缓冲区，而不是等待整个持续时间结束。(这也是ToggleStream函数的目的，rtmpdump在读取超时时调用该函数2. 媒体播放器应用程序不需要这个hack，因为它们只是要实时播放数据。它也不适用于直播流，因为它们显然只能实时发送。如果网络速度实际上比媒体比特率慢，那么这一切都没有意义*/case 31: // Stream BufferEmptytmp = 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_mediaChannel < r->m_channelsAllocatedIn ?r->m_channelTimestamp[r->m_mediaChannel] : 0;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: // Stream BufferReadytmp = AMF_DecodeInt32(packet->m_body + 2);RTMP_Log(RTMP_LOGDEBUG, "%s, Stream BufferReady %d", __FUNCTION__, tmp);break;default: // Stream xxtmp = 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__);}
#elseRTMP_Log(RTMP_LOGERROR,"%s: Ignoring SWFVerification request, no CRYPTO support!",__FUNCTION__);
#endif}
}

1.2.3 设置应答窗口大小（HandleServerBW）

从RTMPDump代码中看，这条命令消息通常由client发出到server，用于设置应答窗口大小

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);
}

1.2.4 设置对端带宽（HandleClientBW）

从RTMPDump代码中看，这条命令通常由server发送给client，用于设置client发送带宽

static void
HandleClientBW(RTMP * r, const RTMPPacket * packet)
{// 解析带宽r->m_nClientBW = AMF_DecodeInt32(packet->m_body);// m_nClientBW2表示limit type/*1）Limit type = 0 （Hard Limit）硬限制，对端应该限制其输出带宽到指示的窗口大小（2）Limit type = 1 （Soft Limit）对端应该限制其输出带宽到知识的窗口大小，或者已经有限制在其作用的话就取两者之间的较小值（3）Limit type = 2（Dynamic Limit）如果先前的限制类型为 Hard，处理这个消息就好像它被标记为 Hard，否则的话忽略这个消息*/if (packet->m_nBodySize > 4)r->m_nClientBW2 = packet->m_body[4];elser->m_nClientBW2 = -1;RTMP_Log(RTMP_LOGDEBUG, "%s: client BW = %d %d", __FUNCTION__, r->m_nClientBW,r->m_nClientBW2);
}

1.2.5 音频数据（HandleAudio）

这个函数没有在RTMPDump中实现

static void
HandleAudio(RTMP * r, const RTMPPacket * packet)
{
}

1.2.6 视频数据（HandleVideo）

这个函数没有在RTMPDump中实现

static void
HandleVideo(RTMP * r, const RTMPPacket * packet)
{
}

1.2.7 元数据（HandleMetadata）

static int
HandleMetadata(RTMP * r, char* body, unsigned int len)
{/* allright we get some info here, so parse it and print it *//* also keep duration or filesize to make a nice progress bar */AMFObject obj;AVal metastring;int ret = FALSE;int nRes = AMF_Decode(&obj, body, len, FALSE);if (nRes < 0){RTMP_Log(RTMP_LOGERROR, "%s, error decoding meta data packet", __FUNCTION__);return FALSE;}AMF_Dump(&obj);AMFProp_GetString(AMF_GetProp(&obj, NULL, 0), &metastring);if (AVMATCH(&metastring, &av_onMetaData)){AMFObjectProperty prop;/* Show metadata */RTMP_Log(RTMP_LOGINFO, "Metadata:");DumpMetaData(&obj); // 输出metadata格式if (RTMP_FindFirstMatchingProperty(&obj, &av_duration, &prop)){r->m_fDuration = prop.p_vu.p_number;/*RTMP_Log(RTMP_LOGDEBUG, "Set duration: %.2f", m_fDuration); */}// 寻找音频或视频标记/* Search for audio or video tags */if (RTMP_FindPrefixProperty(&obj, &av_video, &prop))r->m_read.dataType |= 1;if (RTMP_FindPrefixProperty(&obj, &av_audio, &prop))r->m_read.dataType |= 4;ret = TRUE;}AMF_Reset(&obj);return ret;
}

1.2.8 命令消息（HandleInvoke）

在RTMPDump中，该函数主要被用于处理server返回过来的命令消息

/* Returns 0 for OK/Failed/error, 1 for 'Stop or Complete' */
static int
HandleInvoke(RTMP * r, const char* body, unsigned int nBodySize)
{AMFObject obj;AVal method;double txn;int ret = 0, nRes;if (body[0] != 0x02)		/* make sure it is a string method name we start with */{RTMP_Log(RTMP_LOGWARNING, "%s, Sanity failed. no string method in invoke packet",__FUNCTION__);return 0;}nRes = AMF_Decode(&obj, body, nBodySize, FALSE);if (nRes < 0){RTMP_Log(RTMP_LOGERROR, "%s, error decoding invoke packet", __FUNCTION__);return 0;}AMF_Dump(&obj);AMFProp_GetString(AMF_GetProp(&obj, NULL, 0), &method);txn = AMFProp_GetNumber(AMF_GetProp(&obj, NULL, 1));RTMP_Log(RTMP_LOGDEBUG, "%s, server invoking <%s>", __FUNCTION__, method.av_val);if (AVMATCH(&method, &av__result))	// 检查是否是av__result命令{AVal methodInvoked = { 0 };int i;for (i = 0; i < r->m_numCalls; i++) {if (r->m_methodCalls[i].num == (int)txn) {methodInvoked = r->m_methodCalls[i].name;AV_erase(r->m_methodCalls, &r->m_numCalls, i, FALSE);break;}}if (!methodInvoked.av_val) {RTMP_Log(RTMP_LOGDEBUG, "%s, received result id %f without matching request",__FUNCTION__, txn);goto leave;}RTMP_Log(RTMP_LOGDEBUG, "%s, received result for method call <%s>", __FUNCTION__,methodInvoked.av_val);// 检查是否是av_connect命令/*我理解这里的意思应该是，从server返回了一个result，并且是client发送出去av_connect的result*/if (AVMATCH(&methodInvoked, &av_connect)){if (r->Link.token.av_len){AMFObjectProperty p;if (RTMP_FindFirstMatchingProperty(&obj, &av_secureToken, &p)){DecodeTEA(&r->Link.token, &p.p_vu.p_aval);SendSecureTokenResponse(r, &p.p_vu.p_aval);}}if (r->Link.protocol & RTMP_FEATURE_WRITE){SendReleaseStream(r);SendFCPublish(r);}else{RTMP_SendServerBW(r);RTMP_SendCtrl(r, 3, 0, 300);}// 前面发送的connect已经成功了，现在可以发送申请创建流的命令RTMP_SendCreateStream(r);if (!(r->Link.protocol & RTMP_FEATURE_WRITE)){/* Authenticate on Justin.tv legacy servers before sending FCSubscribe */if (r->Link.usherToken.av_len)SendUsherToken(r, &r->Link.usherToken);/* Send the FCSubscribe if live stream or if subscribepath is set */if (r->Link.subscribepath.av_len)SendFCSubscribe(r, &r->Link.subscribepath);else if (r->Link.lFlags & RTMP_LF_LIVE)SendFCSubscribe(r, &r->Link.playpath);}}else if (AVMATCH(&methodInvoked, &av_createStream)) // 检查是否是av_createStream命令{r->m_stream_id = (int)AMFProp_GetNumber(AMF_GetProp(&obj, NULL, 3));if (r->Link.protocol & RTMP_FEATURE_WRITE){SendPublish(r);}else{if (r->Link.lFlags & RTMP_LF_PLST)SendPlaylist(r);// 前面发送的av_createStream命令成功了，现在可以发送play和control的命令SendPlay(r);RTMP_SendCtrl(r, 3, r->m_stream_id, r->m_nBufferMS);}}else if (AVMATCH(&methodInvoked, &av_play) ||AVMATCH(&methodInvoked, &av_publish)) // 检查是否是av_play或av_publish命令{r->m_bPlaying = TRUE;}free(methodInvoked.av_val);}else if (AVMATCH(&method, &av_onBWDone)) // 检查是否是av_onBWDone命令{if (!r->m_nBWCheckCounter)SendCheckBW(r);}else if (AVMATCH(&method, &av_onFCSubscribe)) // 检查是否是av_onFCSubscribe命令{/* SendOnFCSubscribe(); */}else if (AVMATCH(&method, &av_onFCUnsubscribe)) // 检查是否是av_onFCUnsubscribe命令{RTMP_Close(r);ret = 1;}else if (AVMATCH(&method, &av_ping)) // 检查是否是av_ping命令{SendPong(r, txn);}else if (AVMATCH(&method, &av__onbwcheck)) // 检查是否是av__onbwcheck命令{SendCheckBWResult(r, txn);}else if (AVMATCH(&method, &av__onbwdone)) // 检查是否是av__onbwdone命令{int i;for (i = 0; i < r->m_numCalls; i++)if (AVMATCH(&r->m_methodCalls[i].name, &av__checkbw)){AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);break;}}else if (AVMATCH(&method, &av__error)) // 检查是否是av__error命令{
#ifdef CRYPTOAVal methodInvoked = { 0 };int i;if (r->Link.protocol & RTMP_FEATURE_WRITE){for (i = 0; i < r->m_numCalls; i++){if (r->m_methodCalls[i].num == txn){methodInvoked = r->m_methodCalls[i].name;AV_erase(r->m_methodCalls, &r->m_numCalls, i, FALSE);break;}}if (!methodInvoked.av_val){RTMP_Log(RTMP_LOGDEBUG, "%s, received result id %f without matching request",__FUNCTION__, txn);goto leave;}RTMP_Log(RTMP_LOGDEBUG, "%s, received error for method call <%s>", __FUNCTION__,methodInvoked.av_val);if (AVMATCH(&methodInvoked, &av_connect)){AMFObject obj2;AVal code, level, description;AMFProp_GetObject(AMF_GetProp(&obj, NULL, 3), &obj2);AMFProp_GetString(AMF_GetProp(&obj2, &av_code, -1), &code);AMFProp_GetString(AMF_GetProp(&obj2, &av_level, -1), &level);AMFProp_GetString(AMF_GetProp(&obj2, &av_description, -1), &description);RTMP_Log(RTMP_LOGDEBUG, "%s, error description: %s", __FUNCTION__, description.av_val);/* if PublisherAuth returns 1, then reconnect */if (PublisherAuth(r, &description) == 1){CloseInternal(r, 1);if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0))goto leave;}}}else{RTMP_Log(RTMP_LOGERROR, "rtmp server sent error");}free(methodInvoked.av_val);
#elseRTMP_Log(RTMP_LOGERROR, "rtmp server sent error");
#endif}else if (AVMATCH(&method, &av_close)) // 检查是否是av_close命令{RTMP_Log(RTMP_LOGERROR, "rtmp server requested close");RTMP_Close(r);}else if (AVMATCH(&method, &av_onStatus)) // 检查是否是av_onStatus命令{	// server使用“onStatus”命令向client发送NetStream状态更新AMFObject obj2;AVal code, level;AMFProp_GetObject(AMF_GetProp(&obj, NULL, 3), &obj2);AMFProp_GetString(AMF_GetProp(&obj2, &av_code, -1), &code);AMFProp_GetString(AMF_GetProp(&obj2, &av_level, -1), &level);RTMP_Log(RTMP_LOGDEBUG, "%s, onStatus: %s", __FUNCTION__, code.av_val);if (AVMATCH(&code, &av_NetStream_Failed)|| AVMATCH(&code, &av_NetStream_Play_Failed)|| AVMATCH(&code, &av_NetStream_Play_StreamNotFound)|| AVMATCH(&code, &av_NetConnection_Connect_InvalidApp)){r->m_stream_id = -1;RTMP_Close(r);RTMP_Log(RTMP_LOGERROR, "Closing connection: %s", code.av_val);}else if (AVMATCH(&code, &av_NetStream_Play_Start)|| AVMATCH(&code, &av_NetStream_Play_PublishNotify)){int i;r->m_bPlaying = TRUE;for (i = 0; i < r->m_numCalls; i++){if (AVMATCH(&r->m_methodCalls[i].name, &av_play)){AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);break;}}}else if (AVMATCH(&code, &av_NetStream_Publish_Start)){int i;r->m_bPlaying = TRUE;for (i = 0; i < r->m_numCalls; i++){if (AVMATCH(&r->m_methodCalls[i].name, &av_publish)){AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);break;}}}/* Return 1 if this is a Play.Complete or Play.Stop */else if (AVMATCH(&code, &av_NetStream_Play_Complete)|| AVMATCH(&code, &av_NetStream_Play_Stop)|| AVMATCH(&code, &av_NetStream_Play_UnpublishNotify)){RTMP_Close(r);ret = 1;}else if (AVMATCH(&code, &av_NetStream_Seek_Notify)){r->m_read.flags &= ~RTMP_READ_SEEKING;}else if (AVMATCH(&code, &av_NetStream_Pause_Notify)){if (r->m_pausing == 1 || r->m_pausing == 2){RTMP_SendPause(r, FALSE, r->m_pauseStamp);r->m_pausing = 3;}}}else if (AVMATCH(&method, &av_playlist_ready)){int i;for (i = 0; i < r->m_numCalls; i++){if (AVMATCH(&r->m_methodCalls[i].name, &av_set_playlist)){AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);break;}}}else{}
leave:AMF_Reset(&obj);return ret;
}

现在假设状态为client向server发送了av_connect命令，server会给予一个反馈，client会根据这个反馈去进行下一步的操作，如果server告诉client，connect成功了，现在就可以调用RTMP_SendCreateStream()函数发送av_createStream命令，RTMP_SendCreateStream()函数定义如下

int
RTMP_SendCreateStream(RTMP * r)
{RTMPPacket packet;char pbuf[256], * pend = pbuf + sizeof(pbuf);char* enc;packet.m_nChannel = 0x03;	/* control channel (invoke) */packet.m_headerType = RTMP_PACKET_SIZE_MEDIUM;packet.m_packetType = RTMP_PACKET_TYPE_INVOKE;packet.m_nTimeStamp = 0;packet.m_nInfoField2 = 0;packet.m_hasAbsTimestamp = 0;packet.m_body = pbuf + RTMP_MAX_HEADER_SIZE;enc = packet.m_body;enc = AMF_EncodeString(enc, pend, &av_createStream); // 写入av_createStream命令enc = AMF_EncodeNumber(enc, pend, ++r->m_numInvokes);*enc++ = AMF_NULL;		/* NULL */packet.m_nBodySize = enc - packet.m_body;return RTMP_SendPacket(r, &packet, TRUE);
}

2.小结

本文记录了使用RTMP进行流连接的过程，主要内容包括：
（1）读取server反馈的packet
（2）解析packet。前面client已经发送了av_connect命令，这里会解析这条命令是否成功，如果成功则可以使用RTMP_SendCreateStream()来发送av_createStream命令，申请创建流连接