[Live-devel] Once again, SDP support for Live555 & interaction with FFMpeg
1.First, create a “MediaSession” object, by calling
“MediaSession::createNew()”, with the SDP description (string) as

2.Then, go through each of this object’s ‘subsessions’ (in this case,
there’ll be just one, for “video”), and call
“MediaSubsession::initiate()” on it.

3.Then, you can create an appropriate ‘sink’ object (e.g.,
encapsulating your decoder), and then call “startPlaying()” on it,
passing the subsession’s “readSource()” as parameter.

See the “openRTSP” code (specifically, “testProgs/playCommon.cpp”)
for an example of how this is done.





This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA

// Copyright (c) 1996-2013, Live Networks, Inc.  All rights reserved
// A demo application, showing how to create and run a RTSP client (that can potentially receive multiple streams concurrently).
// NOTE: This code - although it builds a running application - is intended only to illustrate how to develop your own RTSP
// client application.  For a full-featured RTSP client application - with much more functionality, and many options - see
// "openRTSP":

#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"

// Forward function definitions:
class RTPClient;
// RTSP 'response handlers':
void startSetupSubSession(RTPClient* rtpClient);
void rtpClientStartPlay(RTPClient* rtpClient);
void initMediaSession(RTPClient* rtpClient);

// Other event handler functions:
void subsessionAfterPlaying(void* clientData); // called when a stream's subsession (e.g., audio or video substream) ends
void subsessionByeHandler(void* clientData); // called when a RTCP "BYE" is received for a subsession
void streamTimerHandler(void* clientData);
// called at the end of a stream's expected duration (if the stream has not already signaled its end using a RTCP "BYE")

// The main streaming routine (for each "rtsp://" URL):
void openURL(UsageEnvironment& env, char const* progName, char const* szSDP);

// Used to iterate through each stream's 'subsessions', setting up each one:
void setupNextSubsession(RTPClient* rtpClient);

// Used to shut down and close a stream (including its "RTPClient" object):
void shutdownStream(RTPClient* rtpClient, int exitCode = 1);

// A function that outputs a string that identifies each stream (for debugging output).  Modify this if you wish:
//UsageEnvironment& operator<<(UsageEnvironment& env, const RTPClient& rtpClient) {
//return env << "[URL:\"" << rtpClient.url() << "\"]: ";

// A function that outputs a string that identifies each subsession (for debugging output).  Modify this if you wish:
UsageEnvironment& operator<<(UsageEnvironment& env, const MediaSubsession& subsession) {
    return env << subsession.mediumName() << "/" << subsession.codecName();

void usage(UsageEnvironment& env, char const* progName) {
    env << "Usage: " << progName <<  "c=IN IP4\r\nm=video 5000 RTP/AVP 105\r\na=rtpmap:105 H264\r\n"; 

char eventLoopWatchVariable = 0;

int main(int argc, char** argv) {
    // Begin by setting up our usage environment:
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

    // We need at least one SDP argument, like:
       //"c=IN IP4\r\nm=video 5000 RTP/AVP 105\r\na=rtpmap:105 H264\r\n“
       //注意SDP参数的含义: c字段为RTP数据源的IP, m字段video通道的本地rtp端口为5000,payload type为105

    if (argc < 2) {
        usage(*env, argv[0]);
        return 1;

    // There are argc-1 URLs: argv[1] through argv[argc-1].  Open and start streaming each one:
    for (int i = 1; i <= argc-1; ++i) {
        openURL(*env, argv[0], argv[i]);

    // All subsequent activity takes place within the event loop:
    // This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero.

    return 0;

    // If you choose to continue the application past this point (i.e., if you comment out the "return 0;" statement above),
    // and if you don't intend to do anything more with the "TaskScheduler" and "UsageEnvironment" objects,
    // then you can also reclaim the (small) memory used by these objects by uncommenting the following code:
    env->reclaim(); env = NULL;
    delete scheduler; scheduler = NULL;


// Define a class to hold per-stream state that we maintain throughout each stream's lifetime:

class StreamClientState {
    virtual ~StreamClientState();

    MediaSubsessionIterator* iter;
    MediaSession* session;
    MediaSubsession* subsession;
    TaskToken streamTimerTask;
    double duration;

// Define a data sink (a subclass of "MediaSink") to receive the data for each subsession (i.e., each audio or video 'substream').
// In practice, this might be a class (or a chain of classes) that decodes and then renders the incoming audio or video.
// Or it might be a "FileSink", for outputting the received data into a file (as is done by the "openRTSP" application).
// In this example code, however, we define a simple 'dummy' sink that receives incoming data, but does nothing with it.

class DummySink: public MediaSink {
    static DummySink* createNew(UsageEnvironment& env,
        MediaSubsession& subsession, // identifies the kind of data that's being received
        char const* streamId = NULL); // identifies the stream itself (optional)

    DummySink(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId);
    // called only by "createNew()"
    virtual ~DummySink();

    static void afterGettingFrame(void* clientData, unsigned frameSize,
        unsigned numTruncatedBytes,
    struct timeval presentationTime,
        unsigned durationInMicroseconds);
    void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
    struct timeval presentationTime, unsigned durationInMicroseconds);

    // redefined virtual functions:
    virtual Boolean continuePlaying();

    u_int8_t* fReceiveBuffer;
    MediaSubsession& fSubsession;
    char* fStreamId;

class RTPClient {
    RTPClient(UsageEnvironment& e, MediaSession *ms) : mediaSession(ms), env(e)

    virtual ~RTPClient()

    UsageEnvironment&  envir()
        return env;
    MediaSession* mediaSession;
    UsageEnvironment& env;
    StreamClientState scs;


#define RTSP_CLIENT_VERBOSITY_LEVEL 1 // by default, print verbose output from each "RTPClient"

static unsigned rtspClientCount = 0; // Counts how many streams (i.e., "RTPClient"s) are currently in use.

void openURL(UsageEnvironment& env, char const* progName, char const* szSDP) {
    // Begin by creating a "RTPClient" object.  Note that there is a separate "RTPClient" object for each stream that we wish
    // to receive (even if more than stream uses the same "rtsp://" URL).
    MediaSession* mediaSession = MediaSession::createNew(env,  szSDP);
    if (mediaSession == NULL){
        env << "Failed to create a RTP Client for SDP \"" << szSDP << "\": " << env.getResultMsg() << "\n";
        return ;

    RTPClient *client = new RTPClient(env, mediaSession);
    if(client == NULL)
        return ;


    // Next, send a RTSP "DESCRIBE" command, to get a SDP description for the stream.
    // Note that this command - like all RTSP commands - is sent asynchronously; we do not block, waiting for a response.
    // Instead, the following function call returns immediately, and we handle the RTSP response later, from within the event loop:

// Implementation of the RTSP 'response handlers':

void initMediaSession(RTPClient* rtpClient)
    do {
        UsageEnvironment& env = rtpClient->envir(); // alias
        StreamClientState& scs = ((RTPClient*)rtpClient)->scs; // alias
        if (!scs.session->hasSubsessions())
            env << "This session has no media subsessions (i.e., no \"m=\" lines)\n";
            //env << *rtpClient << "Failed to get a SDP description: " << resultString << "\n";
            //delete[] resultString;

        // Then, create and set up our data source objects for the session.  We do this by iterating over the session's 'subsessions',
        // calling "MediaSubsession::initiate()", and then sending a RTSP "SETUP" command, on each one.
        // (Each 'subsession' will have its own data source.)
        scs.iter = new MediaSubsessionIterator(*scs.session);
    } while (0);

    // An unrecoverable error occurred with this stream.

// By default, we request that the server stream its data using RTP/UDP.
// If, instead, you want to request that the server stream via RTP-over-TCP, change the following to True:

void setupNextSubsession(RTPClient* rtpClient) {
    UsageEnvironment& env = rtpClient->envir(); // alias
    StreamClientState& scs = ((RTPClient*)rtpClient)->scs; // alias

    scs.subsession = scs.iter->next();
    if (scs.subsession != NULL) {
        if (!scs.subsession->initiate()) {
            env << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n";
            setupNextSubsession(rtpClient); // give up on this subsession; go to the next one
        } else {
            env << "Initiated the \"" << *scs.subsession
                << "\" subsession (client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << ")\n";


    // We've finished setting up all of the subsessions.  Now,  start the streaming:

void startSetupSubSession(RTPClient* rtpClient) {
    bool success = true;
    do {
        UsageEnvironment& env = rtpClient->envir(); // alias
        StreamClientState& scs = ((RTPClient*)rtpClient)->scs; // alias

        env << "Set up the \"" << *scs.subsession
            << "\" subsession (client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << ")\n";

        // Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
        // (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
        // after we've sent a RTSP "PLAY" command.)

        scs.subsession->sink = DummySink::createNew(env, *scs.subsession, "test");
        // perhaps use your own custom "MediaSink" subclass instead
        if (scs.subsession->sink == NULL) {
            env << "Failed to create a data sink for the \"" << *scs.subsession
                << "\" subsession: " << env.getResultMsg() << "\n";
            success = false;

        env << "Created a data sink for the \"" << *scs.subsession << "\" subsession\n";
        scs.subsession->miscPtr = rtpClient; // a hack to let subsession handle functions get the "RTPClient" from the subsession
            subsessionAfterPlaying, scs.subsession);
        // Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
        if (scs.subsession->rtcpInstance() != NULL) {
            scs.subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, scs.subsession);
    } while (0);

    // Set up the next subsession, if any:

void rtpClientStartPlay(RTPClient* rtpClient) {
    Boolean success = False;

    do {
        UsageEnvironment& env = rtpClient->envir(); // alias
        StreamClientState& scs = ((RTPClient*)rtpClient)->scs; // alias

        // Set a timer to be handled at the end of the stream's expected duration (if the stream does not already signal its end
        // using a RTCP "BYE").  This is optional.  If, instead, you want to keep the stream active - e.g., so you can later
        // 'seek' back within it and do another RTSP "PLAY" - then you can omit this code.
        // (Alternatively, if you don't want to receive the entire stream, you could set this timer for some shorter value.)
        if (scs.duration > 0) {
            unsigned const delaySlop = 2; // number of seconds extra to delay, after the stream's expected duration.  (This is optional.)
            scs.duration += delaySlop;
            unsigned uSecsToDelay = (unsigned)(scs.duration*1000000);
            scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtpClient);

        env << "Started playing session";
        if (scs.duration > 0) {
            env << " (for up to " << scs.duration << " seconds)";
        env << "...\n";

        success = True;
    } while (0);

    if (!success) {
        // An unrecoverable error occurred with this stream.

// Implementation of the other event handlers:

void subsessionAfterPlaying(void* clientData) {
    MediaSubsession* subsession = (MediaSubsession*)clientData;
    RTPClient* rtpClient = (RTPClient*)(subsession->miscPtr);

    // Begin by closing this subsession's stream:
    subsession->sink = NULL;

    // Next, check whether *all* subsessions' streams have now been closed:
    MediaSession& session = subsession->parentSession();
    MediaSubsessionIterator iter(session);
    while ((subsession = != NULL) {
        if (subsession->sink != NULL) return; // this subsession is still active

    // All subsessions' streams have now been closed, so shutdown the client:

void subsessionByeHandler(void* clientData) {
    MediaSubsession* subsession = (MediaSubsession*)clientData;
    RTPClient* rtpClient = (RTPClient*)subsession->miscPtr;
    UsageEnvironment& env = rtpClient->envir(); // alias

    env << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession\n";

    // Now act as if the subsession had closed:

void streamTimerHandler(void* clientData) {
    RTPClient* rtpClient = (RTPClient*)clientData;
    StreamClientState& scs = rtpClient->scs; // alias

    scs.streamTimerTask = NULL;

    // Shut down the stream:

void shutdownStream(RTPClient* rtpClient) {
    UsageEnvironment& env = rtpClient->envir(); // alias
    StreamClientState& scs = ((RTPClient*)rtpClient)->scs; // alias

    // First, check whether any subsessions have still to be closed:
    if (scs.session != NULL) {
        Boolean someSubsessionsWereActive = False;
        MediaSubsessionIterator iter(*scs.session);
        MediaSubsession* subsession;

        while ((subsession = != NULL) {
            if (subsession->sink != NULL) {
                subsession->sink = NULL;

                if (subsession->rtcpInstance() != NULL) {
                    subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"

                someSubsessionsWereActive = True;

        if (someSubsessionsWereActive) {
            // Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
            // Don't bother handling the response to the "TEARDOWN".
            //rtpClient->sendTeardownCommand(*scs.session, NULL);

    env << "Closing the stream.\n";
    delete rtpClient;
    // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.


// Implementation of "StreamClientState":

: iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0) {

StreamClientState::~StreamClientState() {
    delete iter;
    if (session != NULL) {
        // We also need to delete "session", and unschedule "streamTimerTask" (if set)
        UsageEnvironment& env = session->envir(); // alias


// Implementation of "DummySink":

// Even though we're not going to be doing anything with the incoming data, we still need to receive it.
// Define the size of the buffer that we'll use:

DummySink* DummySink::createNew(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId) {
    return new DummySink(env, subsession, streamId);

DummySink::DummySink(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId)
: MediaSink(env),
fSubsession(subsession) {
    fStreamId = strDup(streamId);
    fReceiveBuffer = new u_int8_t[DUMMY_SINK_RECEIVE_BUFFER_SIZE];

DummySink::~DummySink() {
    delete[] fReceiveBuffer;
    delete[] fStreamId;

void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned durationInMicroseconds) {
    DummySink* sink = (DummySink*)clientData;
    sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);

// If you don't want to see debugging output for each received frame, then comment out the following line:

void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
    // We've just received a frame of data.  (Optionally) print out information about it:
    if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
    envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
    if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
    char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
    sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
    envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
    if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
        envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
    envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
    envir() << "\n";

    // Then continue, to request the next frame of data:

Boolean DummySink::continuePlaying() {
    if (fSource == NULL) return False; // sanity check (should not happen)

    // Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:
    fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,
        afterGettingFrame, this,
        onSourceClosure, this);
    return True;


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