注：此前写了一些列的分析RTMPdump（libRTMP）源代码的文章，在此列一个列表：
RTMPdump 源代码分析 1： main()函数
RTMPDump（libRTMP）源代码分析 2：解析RTMP地址——RTMP_ParseURL()
RTMPdump（libRTMP） 源代码分析 3： AMF编码
RTMPdump（libRTMP）源代码分析 4： 连接第一步——握手（Hand Shake）
RTMPdump（libRTMP） 源代码分析 5： 建立一个流媒体连接 （NetConnection部分）
RTMPdump（libRTMP） 源代码分析 6： 建立一个流媒体连接 （NetStream部分 1）
RTMPdump（libRTMP） 源代码分析 7： 建立一个流媒体连接 （NetStream部分 2）
RTMPdump（libRTMP） 源代码分析 8： 发送消息（Message）
RTMPdump（libRTMP） 源代码分析 9： 接收消息（Message）（接收视音频数据）
RTMPdump（libRTMP） 源代码分析 10： 处理各种消息（Message）
===============================
rtmpdump 是一个用来处理 RTMP 流媒体的工具包，支持 rtmp://, rtmpt://, rtmpe://, rtmpte://, and rtmps:// 等。之前在学习RTMP协议的时候，发现没有讲它源代码的，只好自己分析，现在打算把自己学习的成果写出来，可能结果不一定都对，先暂且记录一下。
使用RTMPdump下载一个流媒体的大致流程是这样的：
RTMP_Init();//初始化结构体 InitSockets();//初始化Socket RTMP_ParseURL();//解析输入URL RTMP_SetupStream();//一些设置 fopen();//打开文件，准备写入 RTMP_Connect();//建立NetConnection RTMP_ConnectStream()//建立NetStream Download();//下载函数 RTMP_Close();//关闭连接 fclose();//关闭文件 CleanupSockets();//清理Socket
其中Download()主要是使用RTMP_Read()进行下载的。
注：可以参考：RTMP流媒体播放过程
下面贴上自己注释的RTMPDump源代码。注意以下几点：
1.此RTMPDump已经被移植进VC 2010 的 MFC的工程，所以main()函数已经被改名为rtmpdump()，而且参数也改了，传进来一个MFC窗口的句柄。不过功能没怎么改（控制台程序移植到MFC以后，main()就不是程序的入口了，所以main()名字改成什么是无所谓的）
2.里面有很多提取信息的代码形如：rtmp.dlg->AppendCInfo("开始初始化Socket...");这些代码是我为了获取RTMP信息而自己加的，并不影响程序的执行。
int rtmpdump(LPVOID lpParam,int argc,char **argv)
{
extern char *optarg;
//一定要设置，否则只能运行一次
extern int optind;
optind=0;
int nStatus = RD_SUCCESS;
double percent = 0;
double duration = 0.0;
int nSkipKeyFrames = DEF_SKIPFRM; // skip this number of keyframes when resuming
int bOverrideBufferTime = FALSE; // if the user specifies a buffer time override this is true
int bStdoutMode = TRUE; // if true print the stream directly to stdout, messages go to stderr
int bResume = FALSE; // true in resume mode
uint32_t dSeek = 0; // seek position in resume mode, 0 otherwise
uint32_t bufferTime = DEF_BUFTIME;
// meta header and initial frame for the resume mode (they are read from the file and compared with
// the stream we are trying to continue
char *metaHeader = 0;
uint32_t nMetaHeaderSize = 0;
// video keyframe for matching
char *initialFrame = 0;
uint32_t nInitialFrameSize = 0;
int initialFrameType = 0; // tye: audio or video
AVal hostname = { 0, 0 };
AVal playpath = { 0, 0 };
AVal subscribepath = { 0, 0 };
int port = -1;
int protocol = RTMP_PROTOCOL_UNDEFINED;
int retries = 0;
int bLiveStream = FALSE; // 是直播流吗? then we can't seek/resume
int bHashes = FALSE; // display byte counters not hashes by default
long int timeout = DEF_TIMEOUT; // timeout connection after 120 seconds
uint32_t dStartOffset = 0; // 非直播流搜寻点seek position in non-live mode
uint32_t dStopOffset = 0;
RTMP rtmp = { 0 };
AVal swfUrl = { 0, 0 };
AVal tcUrl = { 0, 0 };
AVal pageUrl = { 0, 0 };
AVal app = { 0, 0 };
AVal auth = { 0, 0 };
AVal swfHash = { 0, 0 };
uint32_t swfSize = 0;
AVal flashVer = { 0, 0 };
AVal sockshost = { 0, 0 };
#ifdef CRYPTO
int swfAge = 30; /* 30 days for SWF cache by default */
int swfVfy = 0;
unsigned char hash[RTMP_SWF_HASHLEN];
#endif
char *flvFile = 0;
signal(SIGINT, sigIntHandler);
signal(SIGTERM, sigIntHandler);
#ifndef WIN32
signal(SIGHUP, sigIntHandler);
signal(SIGPIPE, sigIntHandler);
signal(SIGQUIT, sigIntHandler);
#endif
RTMP_debuglevel = RTMP_LOGINFO;
//首先搜寻“ --quiet”选项
int index = 0;
while (index < argc)
{
if (strcmp(argv[index], "--quiet") == 0
|| strcmp(argv[index], "-q") == 0)
RTMP_debuglevel = RTMP_LOGCRIT;
index++;
}
#define RTMPDUMP_VERSION "1.0"
RTMP_LogPrintf("RTMP流媒体下载 %s\n", RTMPDUMP_VERSION);
RTMP_LogPrintf
("2012 雷霄骅 中国传媒大学/信息工程学院/通信与信息系统/数字电视技术\n");
//RTMP_LogPrintf("输入 -h 获取命令选项\n");
RTMP_Init(&rtmp);
//句柄-----------------------------
rtmp.dlg=(CSpecialPRTMPDlg *)lpParam;
//---------------------------------
//----------------------
rtmp.dlg->AppendCInfo("开始初始化Socket...");
//-----------------------------
if (!InitSockets())
{
//----------------------
rtmp.dlg->AppendCInfo("初始化Socket失败！");
//-----------------------------
RTMP_Log(RTMP_LOGERROR,
"Couldn't load sockets support on your platform, exiting!");
return RD_FAILED;
}
//----------------------
rtmp.dlg->AppendCInfo("成功初始化Socket");
//-----------------------------
/* sleep(30); */
int opt;
/* struct option longopts[] = {
{"help", 0, NULL, 'h'},
{"host", 1, NULL, 'n'},
{"port", 1, NULL, 'c'},
{"socks", 1, NULL, 'S'},
{"protocol", 1, NULL, 'l'},
{"playpath", 1, NULL, 'y'},
{"playlist", 0, NULL, 'Y'},
{"rtmp", 1, NULL, 'r'},
{"swfUrl", 1, NULL, 's'},
{"tcUrl", 1, NULL, 't'},
{"pageUrl", 1, NULL, 'p'},
{"app", 1, NULL, 'a'},
{"auth", 1, NULL, 'u'},
{"conn", 1, NULL, 'C'},
#ifdef CRYPTO
{"swfhash", 1, NULL, 'w'},
{"swfsize", 1, NULL, 'x'},
{"swfVfy", 1, NULL, 'W'},
{"swfAge", 1, NULL, 'X'},
#endif
{"flashVer", 1, NULL, 'f'},
{"live", 0, NULL, 'v'},
{"flv", 1, NULL, 'o'},
{"resume", 0, NULL, 'e'},
{"timeout", 1, NULL, 'm'},
{"buffer", 1, NULL, 'b'},
{"skip", 1, NULL, 'k'},
{"subscribe", 1, NULL, 'd'},
{"start", 1, NULL, 'A'},
{"stop", 1, NULL, 'B'},
{"token", 1, NULL, 'T'},
{"hashes", 0, NULL, '#'},
{"debug", 0, NULL, 'z'},
{"quiet", 0, NULL, 'q'},
{"verbose", 0, NULL, 'V'},
{0, 0, 0, 0}
};*/
//分析命令行参数，注意用法。
//选项都是一个字母，后面有冒号的代表该选项还有相关参数
//一直循环直到获取所有的opt
while ((opt =
getopt/*_long*/(argc, argv,
"hVveqzr:s:t:p:a:b:f:o:u:C:n:c:l:y:Ym:k:d:A:B:T:w:x:W:X:S:#"/*,
longopts, NULL*/)) != -1)
{
//不同的选项做不同的处理
switch (opt)
{
case 'h':
usage(argv[0]);
return RD_SUCCESS;
#ifdef CRYPTO
case 'w':
{
int res = hex2bin(optarg, &swfHash.av_val);
if (res != RTMP_SWF_HASHLEN)
{
swfHash.av_val = NULL;
RTMP_Log(RTMP_LOGWARNING,
"Couldn't parse swf hash hex string, not hexstring or not %d bytes, ignoring!", RTMP_SWF_HASHLEN);
}
swfHash.av_len = RTMP_SWF_HASHLEN;
break;
}
case 'x':
{
int size = atoi(optarg);
if (size <= 0)
{
RTMP_Log(RTMP_LOGERROR, "SWF Size must be at least 1, ignoring\n");
}
else
{
swfSize = size;
}
break;
}
case 'W':
STR2AVAL(swfUrl, optarg);
swfVfy = 1;
break;
case 'X':
{
int num = atoi(optarg);
if (num < 0)
{
RTMP_Log(RTMP_LOGERROR, "SWF Age must be non-negative, ignoring\n");
}
else
{
swfAge = num;
}
}
break;
#endif
case 'k':
nSkipKeyFrames = atoi(optarg);
if (nSkipKeyFrames < 0)
{
RTMP_Log(RTMP_LOGERROR,
"Number of keyframes skipped must be greater or equal zero, using zero!");
nSkipKeyFrames = 0;
}
else
{
RTMP_Log(RTMP_LOGDEBUG, "Number of skipped key frames for resume: %d",
nSkipKeyFrames);
}
break;
case 'b':
{
int32_t bt = atol(optarg);
if (bt < 0)
{
RTMP_Log(RTMP_LOGERROR,
"Buffer time must be greater than zero, ignoring the specified value %d!",
bt);
}
else
{
bufferTime = bt;
bOverrideBufferTime = TRUE;
}
break;
}
//直播流
case 'v':
//----------------
rtmp.dlg->AppendCInfo("该RTMP的URL是一个直播流");
//----------------
bLiveStream = TRUE; // no seeking or resuming possible!
break;
case 'd':
STR2AVAL(subscribepath, optarg);
break;
case 'n':
STR2AVAL(hostname, optarg);
break;
case 'c':
port = atoi(optarg);
break;
case 'l':
protocol = atoi(optarg);
if (protocol < RTMP_PROTOCOL_RTMP || protocol > RTMP_PROTOCOL_RTMPTS)
{
RTMP_Log(RTMP_LOGERROR, "Unknown protocol specified: %d", protocol);
return RD_FAILED;
}
break;
case 'y':
STR2AVAL(playpath, optarg);
break;
case 'Y':
RTMP_SetOpt(&rtmp, &av_playlist, (AVal *)&av_true);
break;
//路径参数-r
case 'r':
{
AVal parsedHost, parsedApp, parsedPlaypath;
unsigned int parsedPort = 0;
int parsedProtocol = RTMP_PROTOCOL_UNDEFINED;
//解析URL。注optarg指向参数（URL）
RTMP_LogPrintf("RTMP URL : %s\n",optarg);
//----------------
rtmp.dlg->AppendCInfo("解析RTMP的URL...");
//----------------
if (!RTMP_ParseURL
(optarg, &parsedProtocol, &parsedHost, &parsedPort,
&parsedPlaypath, &parsedApp))
{
//----------------
rtmp.dlg->AppendCInfo("解析RTMP的URL失败！");
//----------------
RTMP_Log(RTMP_LOGWARNING, "无法解析 url (%s)!",
optarg);
}
else
{
//----------------
rtmp.dlg->AppendCInfo("解析RTMP的URL成功");
//----------------
//把解析出来的数据赋值
if (!hostname.av_len)
hostname = parsedHost;
if (port == -1)
port = parsedPort;
if (playpath.av_len == 0 && parsedPlaypath.av_len)
{
playpath = parsedPlaypath;
}
if (protocol == RTMP_PROTOCOL_UNDEFINED)
protocol = parsedProtocol;
if (app.av_len == 0 && parsedApp.av_len)
{
app = parsedApp;
}
}
break;
}
case 's':
STR2AVAL(swfUrl, optarg);
break;
case 't':
STR2AVAL(tcUrl, optarg);
break;
case 'p':
STR2AVAL(pageUrl, optarg);
break;
case 'a':
STR2AVAL(app, optarg);
break;
case 'f':
STR2AVAL(flashVer, optarg);
break;
//指定输出文件
case 'o':
flvFile = optarg;
if (strcmp(flvFile, "-"))
bStdoutMode = FALSE;
break;
case 'e':
bResume = TRUE;
break;
case 'u':
STR2AVAL(auth, optarg);
break;
case 'C': {
AVal av;
STR2AVAL(av, optarg);
if (!RTMP_SetOpt(&rtmp, &av_conn, &av))
{
RTMP_Log(RTMP_LOGERROR, "Invalid AMF parameter: %s", optarg);
return RD_FAILED;
}
}
break;
case 'm':
timeout = atoi(optarg);
break;
case 'A':
dStartOffset = (int) (atof(optarg) * 1000.0);
break;
case 'B':
dStopOffset = (int) (atof(optarg) * 1000.0);
break;
case 'T': {
AVal token;
STR2AVAL(token, optarg);
RTMP_SetOpt(&rtmp, &av_token, &token);
}
break;
case '#':
bHashes = TRUE;
break;
case 'q':
RTMP_debuglevel = RTMP_LOGCRIT;
break;
case 'V':
RTMP_debuglevel = RTMP_LOGDEBUG;
break;
case 'z':
RTMP_debuglevel = RTMP_LOGALL;
break;
case 'S':
STR2AVAL(sockshost, optarg);
break;
default:
RTMP_LogPrintf("unknown option: %c\n", opt);
usage(argv[0]);
return RD_FAILED;
break;
}
}
if (!hostname.av_len)
{
RTMP_Log(RTMP_LOGERROR,
"您必须指定 主机名(hostname) (--host) 或 url (-r \"rtmp://host[:port]/playpath\") 包含 a hostname");
return RD_FAILED;
}
if (playpath.av_len == 0)
{
RTMP_Log(RTMP_LOGERROR,
"您必须指定 播放路径(playpath) (--playpath) 或 url (-r \"rtmp://host[:port]/playpath\") 包含 a playpath");
return RD_FAILED;
}
if (protocol == RTMP_PROTOCOL_UNDEFINED)
{
RTMP_Log(RTMP_LOGWARNING,
"您没有指定 协议(protocol) (--protocol) 或 rtmp url (-r), 默认协议 RTMP");
protocol = RTMP_PROTOCOL_RTMP;
}
if (port == -1)
{
RTMP_Log(RTMP_LOGWARNING,
"您没有指定 端口(port) (--port) 或 rtmp url (-r), 默认端口 1935");
port = 0;
}
if (port == 0)
{
if (protocol & RTMP_FEATURE_SSL)
port = 443;
else if (protocol & RTMP_FEATURE_HTTP)
port = 80;
else
port = 1935;
}
if (flvFile == 0)
{
RTMP_Log(RTMP_LOGWARNING,
"请指定一个输出文件 (-o filename), using stdout");
bStdoutMode = TRUE;
}
if (bStdoutMode && bResume)
{
RTMP_Log(RTMP_LOGWARNING,
"Can't resume in stdout mode, ignoring --resume option");
bResume = FALSE;
}
if (bLiveStream && bResume)
{
RTMP_Log(RTMP_LOGWARNING, "Can't resume live stream, ignoring --resume option");
bResume = FALSE;
}
#ifdef CRYPTO
if (swfVfy)
{
if (RTMP_HashSWF(swfUrl.av_val, (unsigned int *)&swfSize, hash, swfAge) == 0)
{
swfHash.av_val = (char *)hash;
swfHash.av_len = RTMP_SWF_HASHLEN;
}
}
if (swfHash.av_len == 0 && swfSize > 0)
{
RTMP_Log(RTMP_LOGWARNING,
"Ignoring SWF size, supply also the hash with --swfhash");
swfSize = 0;
}
if (swfHash.av_len != 0 && swfSize == 0)
{
RTMP_Log(RTMP_LOGWARNING,
"Ignoring SWF hash, supply also the swf size with --swfsize");
swfHash.av_len = 0;
swfHash.av_val = NULL;
}
#endif
if (tcUrl.av_len == 0)
{
char str[512] = { 0 };
tcUrl.av_len = snprintf(str, 511, "%s://%.*s:%d/%.*s",
RTMPProtocolStringsLower[protocol], hostname.av_len,
hostname.av_val, port, app.av_len, app.av_val);
tcUrl.av_val = (char *) malloc(tcUrl.av_len + 1);
strcpy(tcUrl.av_val, str);
}
int first = 1;
// User defined seek offset
if (dStartOffset > 0)
{
//直播流
if (bLiveStream)
{
RTMP_Log(RTMP_LOGWARNING,
"Can't seek in a live stream, ignoring --start option");
dStartOffset = 0;
}
}
//----------------
rtmp.dlg->AppendCInfo("开始初始化RTMP连接的参数...");
//----------------
//设置
RTMP_SetupStream(&rtmp, protocol, &hostname, port, &sockshost, &playpath,
&tcUrl, &swfUrl, &pageUrl, &app, &auth, &swfHash, swfSize,
&flashVer, &subscribepath, dSeek, dStopOffset, bLiveStream, timeout);
//此处设置参数-----------------
rtmp.dlg->AppendCInfo("成功初始化RTMP连接的参数");
//-----------------------------
char *temp=(char *)malloc(MAX_URL_LENGTH);
memcpy(temp,rtmp.Link.hostname.av_val,rtmp.Link.hostname.av_len);
temp[rtmp.Link.hostname.av_len]='\0';
rtmp.dlg->AppendB_R_L_Info("主机名",temp);
itoa(rtmp.Link.port,temp,10);
rtmp.dlg->AppendB_R_L_Info("端口号",temp);
memcpy(temp,rtmp.Link.app.av_val,rtmp.Link.app.av_len);
temp[rtmp.Link.app.av_len]='\0';
rtmp.dlg->AppendB_R_L_Info("应用程序",temp);
memcpy(temp,rtmp.Link.playpath.av_val,rtmp.Link.playpath.av_len);
temp[rtmp.Link.playpath.av_len]='\0';
rtmp.dlg->AppendB_R_L_Info("路径",temp);
//-----------------------------
/* Try to keep the stream moving if it pauses on us */
if (!bLiveStream && !(protocol & RTMP_FEATURE_HTTP))
rtmp.Link.lFlags |= RTMP_LF_BUFX;
off_t size = 0;
// ok,我们必须获得timestamp of the last keyframe (only keyframes are seekable) / last audio frame (audio only streams)
if (bResume)
{
//打开文件，输出的文件(Resume)
nStatus =
OpenResumeFile(flvFile, &file, &size, &metaHeader, &nMetaHeaderSize,
&duration);
if (nStatus == RD_FAILED)
goto clean;
if (!file)
{
// file does not exist, so go back into normal mode
bResume = FALSE; // we are back in fresh file mode (otherwise finalizing file won't be done)
}
else
{
//获取最后一个关键帧
nStatus = GetLastKeyframe(file, nSkipKeyFrames,
&dSeek, &initialFrame,
&initialFrameType, &nInitialFrameSize);
if (nStatus == RD_FAILED)
{
RTMP_Log(RTMP_LOGDEBUG, "Failed to get last keyframe.");
goto clean;
}
if (dSeek == 0)
{
RTMP_Log(RTMP_LOGDEBUG,
"Last keyframe is first frame in stream, switching from resume to normal mode!");
bResume = FALSE;
}
}
}
//如果输出文件不存在
if (!file)
{
if (bStdoutMode)
{
//直接输出到stdout
file = stdout;
SET_BINMODE(file);
}
else
{
//打开一个文件
//w+b 读写打开或建立一个二进制文件，允许读和写。
//-----------------
rtmp.dlg->AppendCInfo("创建输出文件...");
//-----------------------------
file = fopen(flvFile, "w+b");
if (file == 0)
{
//-----------------
rtmp.dlg->AppendCInfo("创建输出文件失败！");
//-----------------------------
RTMP_LogPrintf("Failed to open file! %s\n", flvFile);
return RD_FAILED;
}
rtmp.dlg->AppendCInfo("成功创建输出文件");
}
}
#ifdef _DEBUG
netstackdump = fopen("netstackdump", "wb");
netstackdump_read = fopen("netstackdump_read", "wb");
#endif
while (!RTMP_ctrlC)
{
RTMP_Log(RTMP_LOGDEBUG, "Setting buffer time to: %dms", bufferTime);
//设置Buffer时间
//-----------------
rtmp.dlg->AppendCInfo("设置缓冲(Buffer)的时间");
//-----------------------------
RTMP_SetBufferMS(&rtmp, bufferTime);
//第一次执行
if (first)
{
first = 0;
RTMP_LogPrintf("开始建立连接！\n");
//-----------------
rtmp.dlg->AppendCInfo("开始建立连接（NetConnection）...");
//-----------------------------
//建立连接(Connect)
if (!RTMP_Connect(&rtmp, NULL))
{
//-----------------
rtmp.dlg->AppendCInfo("建立连接（NetConnection）失败！");
//-----------------------------
nStatus = RD_FAILED;
break;
}
//-----------------
rtmp.dlg->AppendCInfo("成功建立连接（NetConnection）");
//-----------------------------
//RTMP_Log(RTMP_LOGINFO, "已链接...");
// User defined seek offset
if (dStartOffset > 0)
{
// Don't need the start offset if resuming an existing file
if (bResume)
{
RTMP_Log(RTMP_LOGWARNING,
"Can't seek a resumed stream, ignoring --start option");
dStartOffset = 0;
}
else
{
dSeek = dStartOffset;
}
}
// Calculate the length of the stream to still play
if (dStopOffset > 0)
{
// Quit if start seek is past required stop offset
if (dStopOffset <= dSeek)
{
RTMP_LogPrintf("Already Completed\n");
nStatus = RD_SUCCESS;
break;
}
}
//创建流(Stream)（发送connect命令消息后处理传来的数据）
itoa(rtmp.m_inChunkSize,temp,10);
rtmp.dlg->AppendB_R_Info("输入Chunk大小",temp);
itoa(rtmp.m_outChunkSize,temp,10);
rtmp.dlg->AppendB_R_Info("输出Chunk大小",temp);
itoa(rtmp.m_stream_id,temp,10);
rtmp.dlg->AppendB_R_Info("Stream ID",temp);
itoa(rtmp.m_nBufferMS,temp,10);
rtmp.dlg->AppendB_R_Info("Buffer时长（ms）",temp);
itoa(rtmp.m_nServerBW,temp,10);
rtmp.dlg->AppendB_R_Info("ServerBW",temp);
itoa(rtmp.m_nClientBW,temp,10);
rtmp.dlg->AppendB_R_Info("ClientBW",temp);
itoa((int)rtmp.m_fEncoding,temp,10);
rtmp.dlg->AppendB_R_Info("命令消息编码方法",temp);
itoa((int)rtmp.m_fDuration,temp,10);
rtmp.dlg->AppendB_R_Info("时长（s）",temp);
rtmp.dlg->ShowBInfo();
free(temp);
//-----------------
rtmp.dlg->AppendCInfo("开始建立网络流（NetStream）");
//-----------------------------
if (!RTMP_ConnectStream(&rtmp, dSeek))
{
//-----------------
rtmp.dlg->AppendCInfo("建立网络流（NetStream）失败！");
//-----------------
nStatus = RD_FAILED;
break;
}
//-----------------
rtmp.dlg->AppendCInfo("成功建立网络流（NetStream）！");
//-----------------
}
else
{
nInitialFrameSize = 0;
if (retries)
{
RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n");
if (!RTMP_IsTimedout(&rtmp))
nStatus = RD_FAILED;
else
nStatus = RD_INCOMPLETE;
break;
}
RTMP_Log(RTMP_LOGINFO, "Connection timed out, trying to resume.\n\n");
/* Did we already try pausing, and it still didn't work? */
if (rtmp.m_pausing == 3)
{
/* Only one try at reconnecting... */
retries = 1;
dSeek = rtmp.m_pauseStamp;
if (dStopOffset > 0)
{
if (dStopOffset <= dSeek)
{
RTMP_LogPrintf("Already Completed\n");
nStatus = RD_SUCCESS;
break;
}
}
if (!RTMP_ReconnectStream(&rtmp, dSeek))
{
RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n");
if (!RTMP_IsTimedout(&rtmp))
nStatus = RD_FAILED;
else
nStatus = RD_INCOMPLETE;
break;
}
}
else if (!RTMP_ToggleStream(&rtmp))
{
RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n");
if (!RTMP_IsTimedout(&rtmp))
nStatus = RD_FAILED;
else
nStatus = RD_INCOMPLETE;
break;
}
bResume = TRUE;
}
//-----------------
//-----------------
rtmp.dlg->AppendCInfo("开始将媒体数据写入文件");
//-----------------
//下载,写入文件
nStatus = Download(&rtmp, file, dSeek, dStopOffset, duration, bResume,
metaHeader, nMetaHeaderSize, initialFrame,
initialFrameType, nInitialFrameSize,
nSkipKeyFrames, bStdoutMode, bLiveStream, bHashes,
bOverrideBufferTime, bufferTime, &percent);
free(initialFrame);
initialFrame = NULL;
/* If we succeeded, we're done.
*/
if (nStatus != RD_INCOMPLETE || !RTMP_IsTimedout(&rtmp) || bLiveStream)
break;
}
//当下载完的时候
if (nStatus == RD_SUCCESS)
{
//-----------------
rtmp.dlg->AppendCInfo("写入文件完成");
//-----------------
RTMP_LogPrintf("Download complete\n");
}
//没下载完的时候
else if (nStatus == RD_INCOMPLETE)
{
//-----------------
rtmp.dlg->AppendCInfo("写入文件可能不完整");
//-----------------
RTMP_LogPrintf
("Download may be incomplete (downloaded about %.2f%%), try resuming\n",
percent);
}
//后续清理工作
clean:
//-----------------
rtmp.dlg->AppendCInfo("关闭连接");
//-----------------
RTMP_Log(RTMP_LOGDEBUG, "Closing connection.\n");
RTMP_Close(&rtmp);
rtmp.dlg->AppendCInfo("关闭文件");
if (file != 0)
fclose(file);
rtmp.dlg->AppendCInfo("关闭Socket");
CleanupSockets();
#ifdef _DEBUG
if (netstackdump != 0)
fclose(netstackdump);
if (netstackdump_read != 0)
fclose(netstackdump_read);
#endif
return nStatus;
}
其中InitSocket()代码很简单，初始化了Socket，如下：
// 初始化 sockets
int
InitSockets()
{
#ifdef WIN32
WORD version;
WSADATA wsaData;
version = MAKEWORD(1, 1);
return (WSAStartup(version, &wsaData) == 0);
#else
return TRUE;
#endif
}
CleanupSockets()则更简单：
inline void
CleanupSockets()
{
#ifdef WIN32
WSACleanup();
#endif
}
Download()函数则比较复杂：
int
Download(RTMP * rtmp, // connected RTMP object
FILE * file, uint32_t dSeek, uint32_t dStopOffset, double duration, int bResume, char *metaHeader, uint32_t nMetaHeaderSize, char *initialFrame, int initialFrameType, uint32_t nInitialFrameSize, int nSkipKeyFrames, int bStdoutMode, int bLiveStream, int bHashes, int bOverrideBufferTime, uint32_t bufferTime, double *percent) // percentage downloaded [out]
{
int32_t now, lastUpdate;
int bufferSize = 64 * 1024;
char *buffer = (char *) malloc(bufferSize);
int nRead = 0;
//long ftell(FILE *stream);
//返回当前文件指针
RTMP_LogPrintf("开始下载！\n");
off_t size = ftello(file);
unsigned long lastPercent = 0;
//时间戳
rtmp->m_read.timestamp = dSeek;
*percent = 0.0;
if (rtmp->m_read.timestamp)
{
RTMP_Log(RTMP_LOGDEBUG, "Continuing at TS: %d ms\n", rtmp->m_read.timestamp);
}
//是直播
if (bLiveStream)
{
RTMP_LogPrintf("直播流\n");
}
else
{
// print initial status
// Workaround to exit with 0 if the file is fully (> 99.9%) downloaded
if (duration > 0)
{
if ((double) rtmp->m_read.timestamp >= (double) duration * 999.0)
{
RTMP_LogPrintf("Already Completed at: %.3f sec Duration=%.3f sec\n",
(double) rtmp->m_read.timestamp / 1000.0,
(double) duration / 1000.0);
return RD_SUCCESS;
}
else
{
*percent = ((double) rtmp->m_read.timestamp) / (duration * 1000.0) * 100.0;
*percent = ((double) (int) (*percent * 10.0)) / 10.0;
RTMP_LogPrintf("%s download at: %.3f kB / %.3f sec (%.1f%%)\n",
bResume ? "Resuming" : "Starting",
(double) size / 1024.0, (double) rtmp->m_read.timestamp / 1000.0,
*percent);
}
}
else
{
RTMP_LogPrintf("%s download at: %.3f kB\n",
bResume ? "Resuming" : "Starting",
(double) size / 1024.0);
}
}
if (dStopOffset > 0)
RTMP_LogPrintf("For duration: %.3f sec\n", (double) (dStopOffset - dSeek) / 1000.0);
//各种设置参数到rtmp连接
if (bResume && nInitialFrameSize > 0)
rtmp->m_read.flags |= RTMP_READ_RESUME;
rtmp->m_read.initialFrameType = initialFrameType;
rtmp->m_read.nResumeTS = dSeek;
rtmp->m_read.metaHeader = metaHeader;
rtmp->m_read.initialFrame = initialFrame;
rtmp->m_read.nMetaHeaderSize = nMetaHeaderSize;
rtmp->m_read.nInitialFrameSize = nInitialFrameSize;
now = RTMP_GetTime();
lastUpdate = now - 1000;
do
{
//从rtmp中把bufferSize（64k）个数据读入buffer
nRead = RTMP_Read(rtmp, buffer, bufferSize);
//RTMP_LogPrintf("nRead: %d\n", nRead);
if (nRead > 0)
{
//函数：size_t fwrite(const void* buffer,size_t size,size_t count,FILE* stream);
//向文件读入写入一个数据块。返回值：返回实际写入的数据块数目
//（1）buffer：是一个指针，对fwrite来说，是要输出数据的地址。
//（2）size：要写入内容的单字节数；
//（3）count:要进行写入size字节的数据项的个数；
//（4）stream:目标文件指针。
//（5）返回实际写入的数据项个数count。
//关键。把buffer里面的数据写成文件
if (fwrite(buffer, sizeof(unsigned char), nRead, file) !=
(size_t) nRead)
{
RTMP_Log(RTMP_LOGERROR, "%s: Failed writing, exiting!", __FUNCTION__);
free(buffer);
return RD_FAILED;
}
//记录已经写入的字节数
size += nRead;
//RTMP_LogPrintf("write %dbytes (%.1f kB)\n", nRead, nRead/1024.0);
if (duration <= 0) // if duration unknown try to get it from the stream (onMetaData)
duration = RTMP_GetDuration(rtmp);
if (duration > 0)
{
// make sure we claim to have enough buffer time!
if (!bOverrideBufferTime && bufferTime < (duration * 1000.0))
{
bufferTime = (uint32_t) (duration * 1000.0) + 5000; // 再加5s以确保buffertime足够长
RTMP_Log(RTMP_LOGDEBUG,
"Detected that buffer time is less than duration, resetting to: %dms",
bufferTime);
//重设Buffer长度
RTMP_SetBufferMS(rtmp, bufferTime);
//给服务器发送UserControl消息通知Buffer改变
RTMP_UpdateBufferMS(rtmp);
}
//计算百分比
*percent = ((double) rtmp->m_read.timestamp) / (duration * 1000.0) * 100.0;
*percent = ((double) (int) (*percent * 10.0)) / 10.0;
if (bHashes)
{
if (lastPercent + 1 <= *percent)
{
RTMP_LogStatus("#");
lastPercent = (unsigned long) *percent;
}
}
else
{
//设置显示数据的更新间隔200ms
now = RTMP_GetTime();
if (abs(now - lastUpdate) > 200)
{
RTMP_LogStatus("\r%.3f kB / %.2f sec (%.1f%%)",
(double) size / 1024.0,
(double) (rtmp->m_read.timestamp) / 1000.0, *percent);
lastUpdate = now;
}
}
}
else
{
//现在距离开机的毫秒数
now = RTMP_GetTime();
//每间隔200ms刷新一次数据
if (abs(now - lastUpdate) > 200)
{
if (bHashes)
RTMP_LogStatus("#");
else
//size为已写入文件的字节数
RTMP_LogStatus("\r%.3f kB / %.2f sec", (double) size / 1024.0,
(double) (rtmp->m_read.timestamp) / 1000.0);
lastUpdate = now;
}
}
}
#ifdef _DEBUG
else
{
RTMP_Log(RTMP_LOGDEBUG, "zero read!");
}
#endif
}
while (!RTMP_ctrlC && nRead > -1 && RTMP_IsConnected(rtmp) && !RTMP_IsTimedout(rtmp));
free(buffer);
if (nRead < 0)
//nRead是读取情况
nRead = rtmp->m_read.status;
/* Final status update */
if (!bHashes)
{
if (duration > 0)
{
*percent = ((double) rtmp->m_read.timestamp) / (duration * 1000.0) * 100.0;
*percent = ((double) (int) (*percent * 10.0)) / 10.0;
//输出
RTMP_LogStatus("\r%.3f kB / %.2f sec (%.1f%%)",
(double) size / 1024.0,
(double) (rtmp->m_read.timestamp) / 1000.0, *percent);
}
else
{
RTMP_LogStatus("\r%.3f kB / %.2f sec", (double) size / 1024.0,
(double) (rtmp->m_read.timestamp) / 1000.0);
}
}
RTMP_Log(RTMP_LOGDEBUG, "RTMP_Read returned: %d", nRead);
//读取错误
if (bResume && nRead == -2)
{
RTMP_LogPrintf("Couldn't resume FLV file, try --skip %d\n\n",
nSkipKeyFrames + 1);
return RD_FAILED;
}
//读取正确
if (nRead == -3)
return RD_SUCCESS;
//没读完...
if ((duration > 0 && *percent < 99.9) || RTMP_ctrlC || nRead < 0
|| RTMP_IsTimedout(rtmp))
{
return RD_INCOMPLETE;
}
return RD_SUCCESS;
}
以上内容是我能理解到的rtmpdump.c里面的内容。
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
相关推荐
这个压缩包“rtmpdump源代码.rar”包含rtmpdump的源代码，使得开发者可以深入理解其工作原理，进行二次开发或者调试。 **RTMP协议** RTMP是一种用于实时数据传输的协议，常用于在线视频直播和点播服务。它由Adobe ...
1. **获取源代码**: 解压rtmpdump-2.4压缩包。 2. **依赖安装**: 安装必要的编译依赖，例如`libssl-dev`和`librtmp-dev`。 3. **编译源码**: 使用`./configure`进行配置，`make`进行编译，`sudo make install`进行...
这个压缩包“rtmpdump.zip”包含了在Visual Studio环境中编译rtmpdump项目所需的所有关键文件，包括头文件和源代码文件。RTMP是一种广泛用于在线直播和视频分发的协议，由Adobe Systems开发。 首先，让我们深入了解...
1. rtmpdump源代码：这是工具的源代码，程序员可以查看并理解其工作原理，或者根据需求进行定制。 2. librtmp源代码：提供了库的源代码，开发者可以修改、优化或扩展功能。 3. 静态库（.lib文件）：适用于那些希望将...
这个压缩包“rtmpdump-2.2e.tar.gz”包含了rtmpdump的源代码，允许用户深入理解其工作原理并可能进行自定义修改。下面将详细讨论RTMP协议、rtmpdump工具、以及它在IT领域的应用。 RTMP（Real Time Messaging ...
2. **获取源代码**：从rtmpdump项目的官方仓库或第三方源获取源代码。 3. **配置项目**：在VC6.0中创建新项目，将源代码文件添加到项目中，配置相应的编译选项和链接器设置。 4. **解决依赖**：如果项目依赖于特定库...
"bin"目录通常存放可执行文件，而"librtmp_src"则表示RTMP库的源代码，这是rtmpdump的核心部分，用于处理与RTMP服务器的通信。 编译rtmpdump时，你需要确保你有正确的开发环境，例如Visual Studio 2010，因为它是一...
在压缩包中的"rtmpdump-2.3"可能是rtmpdump的源代码或者编译好的二进制文件，用户可以进一步查看源码理解其工作原理，或者直接在Android设备上使用编译后的二进制文件进行流媒体操作。对于开发者来说，这为自定义...
1. **源代码**：这是rtmpdump-2.3的核心部分，由C或C++编写，包含了实现RTMP协议解析和交互的所有函数和结构。开发者可以查看源码，理解其工作原理，甚至进行自定义修改。 2. **编译好的动态库**：预编译的动态库...
2. **下载源代码**：从官方网站或者GitHub仓库获取rtmpdump-2.3的源代码包。 3. **解压与配置**：使用`tar`命令解压缩源代码包，然后进入目录并使用`./configure`进行配置，根据提示检查依赖是否齐全。 4. **编译与...
- 添加源文件：将rtmpdump和librtmp的源代码添加到项目中。 - 适应平台：修改源代码中的编译器特定宏和包含路径，以适应VS2008环境。 - 链接库设置：确保正确配置了所有依赖库，如zlib等。 - 编译和调试：编译...
1. **流媒体下载**：rtmpdump支持从RTMP服务器上抓取视频流并将其保存到本地文件，这对于离线观看或者备份在线视频非常有用。 2. **支持多种平台**：rtmpdump不仅可以在常见的桌面操作系统（如Windows、Linux和macOS...
**rtmpdump包详解** RTMP（Real Time Messaging ...请注意，rtmpdump是一个开源工具，其源代码可供开发者参考和扩展。然而，不合法的使用可能会引起法律问题，因此在使用rtmpdump时，务必尊重内容提供商的版权规定。
rtmpdump is a toolkit for RTMP streams. All forms of RTMP are supported, including rtmp://, rtmpt://, rtmpe://, rtmpte://, and rtmps://. License: GPLv2 Copyright (C) 2009 Andrej Stepanchuk ...
源代码可以让你深入理解RTMP协议的工作原理，以及rtmpdump如何实现数据的捕获和处理。这对于想要自定义或扩展rtmpdump功能的人来说尤其重要。 rtmpdump.exe是rtmpdump的可执行文件，你可以用它来从RTMP服务器上下载...
1. 源代码文件：包含了实现RTMP协议解析和操作的C++源代码。 2. 头文件：定义了类、函数和常量等接口，供其他源文件引用。 3. 编译脚本或配置文件：用于设置编译选项和链接库，确保在VS2005环境下成功编译。 4. ...
rtmpdump-v1.6.tar.gz是rtmpdump的一个特定版本，通过解压这个tar.gz压缩包，我们可以获取到rtmpdump的源代码和其他相关文件。在这个压缩包中，唯一列出的子文件名是"rtmpdump"，这通常指的是rtmpdump的可执行程序...
1. rtmpdump源码：这是rtmpdump项目的完整源代码，包含了C或C++语言编写的函数和结构，开发者可以通过阅读源码了解其工作原理，或者根据需求进行修改和定制。 2. openssl源码和编译结果：openssl库的源代码和编译好...
在编译rtmpdump时，需要将librtmp_src文件夹中的源代码与其他依赖库合并，并配置合适的编译选项。对于Windows环境，这通常涉及设置项目属性、链接库路径和包含目录。完成编译后，将生成rtmpdump可执行文件，可用于与...
同时，开发者也可以利用rtmpdump的源代码，学习如何实现RTMP协议的相关功能，为自己的项目提供支持。 然而，随着HLS (HTTP Live Streaming) 和DASH (Dynamic Adaptive Streaming over HTTP) 等适应性更强的流媒体...