To verify if the handle leak I've noticed is in my code or in liveMedia I wrote a really simple program that just start and stops the event loop. I took the latest testRTSPClient.cpp and just replaced main() with a loop that initiates a thread which starts the event loop - StartEventLoop(). The main thread sleeps for 2s and then sets eventLoopWatchVariable making the thread exit. No streams are started, it's just the event loop going up and down.
Code attached (testLiveMediaUpDown.cpp). I compile it using VC++; threading probably needs to be adjusted if using other compiler/OS. If I run this and monitor handles used I can see that the handles are incremented by one each time the event loop runs. This can e.g. be done using SysInternal's "Process Explorer" (handle column not on by default). Looking at the handle type I can also see it's the "\Device\Afd" handles that are increasing. I believe "Afd" stands for Auxiliary Function Driver and that it's related to sockets, but "Afd" is new to me as I've have never come across it before. Any idea where that handle leak originates? Could it be Windows specific? /Claes From: Erlandsson, Claes P (CERLANDS) Sent: Friday, July 19, 2013 11:20 AM To: 'live-devel@lists.live555.com' Subject: Seeing small handle leak when repeatedly starting and stopping event loop When I start and stop the liveMedia event loop I see a small handle leak in my Windows test client. When looking in Process Explorer I can see the amount of file handles with the name "\Device\Afd" growing. I don't have to stream anything for it to occur, i.e. I just start the event loop and then bring it down by setting the flag to 1. It is very possible it's due to some needed cleanup I'm missing. After leaving doEventLoop I delete my trigger(s), call env->reclaim() and delete the TaskScheduler. Is there anything else that should be done? /Claes
/********** 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 <http://www.gnu.org/copyleft/lesser.html>.) 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": http://www.live555.com/openRTSP/ #include "liveMedia.hh" #include "BasicUsageEnvironment.hh" #include <iostream> #include <process.h> using namespace std; // Forward function definitions: // RTSP 'response handlers': void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString); void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString); void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString); // 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* rtspURL); // Used to iterate through each stream's 'subsessions', setting up each one: void setupNextSubsession(RTSPClient* rtspClient); // Used to shut down and close a stream (including its "RTSPClient" object): void shutdownStream(RTSPClient* rtspClient, 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 RTSPClient& rtspClient) { return env << "[URL:\"" << rtspClient.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 << " <rtsp-url-1> ... <rtsp-url-N>\n"; env << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)\n"; } char eventLoopWatchVariable = 0; void StartEventLoop(void *param) { cout << "Starting event loop" << endl; // Begin by setting up our usage environment: TaskScheduler* scheduler = BasicTaskScheduler::createNew(); UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); // All subsequent activity takes place within the event loop: env->taskScheduler().doEventLoop(&eventLoopWatchVariable); // This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero. // 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; cout << "Leaving event loop" << endl; _endthread(); } int main(int argc, char** argv) { HANDLE hThread; int counter = 0; for (int i=0; i <1000; i++) { cout << "Starting thread " << ++counter << endl; eventLoopWatchVariable = 0; hThread = (HANDLE)_beginthread(StartEventLoop, 0, NULL); Sleep(2000); eventLoopWatchVariable = 1; WaitForSingleObject( hThread, INFINITE ); cout << "Thread \"synced\"" << endl << endl; } cout << "Main ends" << endl; return 0; } // Define a class to hold per-stream state that we maintain throughout each stream's lifetime: class StreamClientState { public: StreamClientState(); virtual ~StreamClientState(); public: MediaSubsessionIterator* iter; MediaSession* session; MediaSubsession* subsession; TaskToken streamTimerTask; double duration; }; // If you're streaming just a single stream (i.e., just from a single URL, once), then you can define and use just a single // "StreamClientState" structure, as a global variable in your application. However, because - in this demo application - we're // showing how to play multiple streams, concurrently, we can't do that. Instead, we have to have a separate "StreamClientState" // structure for each "RTSPClient". To do this, we subclass "RTSPClient", and add a "StreamClientState" field to the subclass: class ourRTSPClient: public RTSPClient { public: static ourRTSPClient* createNew(UsageEnvironment& env, char const* rtspURL, int verbosityLevel = 0, char const* applicationName = NULL, portNumBits tunnelOverHTTPPortNum = 0); protected: ourRTSPClient(UsageEnvironment& env, char const* rtspURL, int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum); // called only by createNew(); virtual ~ourRTSPClient(); public: StreamClientState scs; }; // 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 { public: 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) private: 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); private: // redefined virtual functions: virtual Boolean continuePlaying(); private: u_int8_t* fReceiveBuffer; MediaSubsession& fSubsession; char* fStreamId; }; #define RTSP_CLIENT_VERBOSITY_LEVEL 1 // by default, print verbose output from each "RTSPClient" static unsigned rtspClientCount = 0; // Counts how many streams (i.e., "RTSPClient"s) are currently in use. void openURL(UsageEnvironment& env, char const* progName, char const* rtspURL) { // Begin by creating a "RTSPClient" object. Note that there is a separate "RTSPClient" object for each stream that we wish // to receive (even if more than stream uses the same "rtsp://" URL). RTSPClient* rtspClient = ourRTSPClient::createNew(env, rtspURL, RTSP_CLIENT_VERBOSITY_LEVEL, progName); if (rtspClient == NULL) { env << "Failed to create a RTSP client for URL \"" << rtspURL << "\": " << env.getResultMsg() << "\n"; return; } ++rtspClientCount; // 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: rtspClient->sendDescribeCommand(continueAfterDESCRIBE); } // Implementation of the RTSP 'response handlers': void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString) { do { UsageEnvironment& env = rtspClient->envir(); // alias StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias if (resultCode != 0) { env << *rtspClient << "Failed to get a SDP description: " << resultString << "\n"; delete[] resultString; break; } char* const sdpDescription = resultString; env << *rtspClient << "Got a SDP description:\n" << sdpDescription << "\n"; // Create a media session object from this SDP description: scs.session = MediaSession::createNew(env, sdpDescription); delete[] sdpDescription; // because we don't need it anymore if (scs.session == NULL) { env << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << "\n"; break; } else if (!scs.session->hasSubsessions()) { env << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)\n"; break; } // 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); setupNextSubsession(rtspClient); return; } while (0); // An unrecoverable error occurred with this stream. shutdownStream(rtspClient); } // 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: #define REQUEST_STREAMING_OVER_TCP False void setupNextSubsession(RTSPClient* rtspClient) { UsageEnvironment& env = rtspClient->envir(); // alias StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias scs.subsession = scs.iter->next(); if (scs.subsession != NULL) { if (!scs.subsession->initiate()) { env << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n"; setupNextSubsession(rtspClient); // give up on this subsession; go to the next one } else { env << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << ")\n"; // Continue setting up this subsession, by sending a RTSP "SETUP" command: rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP); } return; } // We've finished setting up all of the subsessions. Now, send a RTSP "PLAY" command to start the streaming: if (scs.session->absStartTime() != NULL) { // Special case: The stream is indexed by 'absolute' time, so send an appropriate "PLAY" command: rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY, scs.session->absStartTime(), scs.session->absEndTime()); } else { scs.duration = scs.session->playEndTime() - scs.session->playStartTime(); rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY); } } void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString) { do { UsageEnvironment& env = rtspClient->envir(); // alias StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias if (resultCode != 0) { env << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << "\n"; break; } env << *rtspClient << "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, rtspClient->url()); // perhaps use your own custom "MediaSink" subclass instead if (scs.subsession->sink == NULL) { env << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n"; break; } env << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession\n"; scs.subsession->miscPtr = rtspClient; // a hack to let subsession handle functions get the "RTSPClient" from the subsession scs.subsession->sink->startPlaying(*(scs.subsession->readSource()), 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); delete[] resultString; // Set up the next subsession, if any: setupNextSubsession(rtspClient); } void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString) { Boolean success = False; do { UsageEnvironment& env = rtspClient->envir(); // alias StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias if (resultCode != 0) { env << *rtspClient << "Failed to start playing session: " << resultString << "\n"; break; } // 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, rtspClient); } env << *rtspClient << "Started playing session"; if (scs.duration > 0) { env << " (for up to " << scs.duration << " seconds)"; } env << "...\n"; success = True; } while (0); delete[] resultString; if (!success) { // An unrecoverable error occurred with this stream. shutdownStream(rtspClient); } } // Implementation of the other event handlers: void subsessionAfterPlaying(void* clientData) { MediaSubsession* subsession = (MediaSubsession*)clientData; RTSPClient* rtspClient = (RTSPClient*)(subsession->miscPtr); // Begin by closing this subsession's stream: Medium::close(subsession->sink); subsession->sink = NULL; // Next, check whether *all* subsessions' streams have now been closed: MediaSession& session = subsession->parentSession(); MediaSubsessionIterator iter(session); while ((subsession = iter.next()) != NULL) { if (subsession->sink != NULL) return; // this subsession is still active } // All subsessions' streams have now been closed, so shutdown the client: shutdownStream(rtspClient); } void subsessionByeHandler(void* clientData) { MediaSubsession* subsession = (MediaSubsession*)clientData; RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr; UsageEnvironment& env = rtspClient->envir(); // alias env << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession\n"; // Now act as if the subsession had closed: subsessionAfterPlaying(subsession); } void streamTimerHandler(void* clientData) { ourRTSPClient* rtspClient = (ourRTSPClient*)clientData; StreamClientState& scs = rtspClient->scs; // alias scs.streamTimerTask = NULL; // Shut down the stream: shutdownStream(rtspClient); } void shutdownStream(RTSPClient* rtspClient, int exitCode) { UsageEnvironment& env = rtspClient->envir(); // alias StreamClientState& scs = ((ourRTSPClient*)rtspClient)->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 = iter.next()) != NULL) { if (subsession->sink != NULL) { Medium::close(subsession->sink); 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". rtspClient->sendTeardownCommand(*scs.session, NULL); } } env << *rtspClient << "Closing the stream.\n"; Medium::close(rtspClient); // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed. if (--rtspClientCount == 0) { // The final stream has ended, so exit the application now. // (Of course, if you're embedding this code into your own application, you might want to comment this out, // and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".) exit(exitCode); } } // Implementation of "ourRTSPClient": ourRTSPClient* ourRTSPClient::createNew(UsageEnvironment& env, char const* rtspURL, int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum) { return new ourRTSPClient(env, rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum); } ourRTSPClient::ourRTSPClient(UsageEnvironment& env, char const* rtspURL, int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum) : RTSPClient(env,rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum, -1) { } ourRTSPClient::~ourRTSPClient() { } // Implementation of "StreamClientState": StreamClientState::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 env.taskScheduler().unscheduleDelayedTask(streamTimerTask); Medium::close(session); } } // 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: #define DUMMY_SINK_RECEIVE_BUFFER_SIZE 100000 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: #define DEBUG_PRINT_EACH_RECEIVED_FRAME 1 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: #ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME 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 } #ifdef DEBUG_PRINT_NPT envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime); #endif envir() << "\n"; #endif // Then continue, to request the next frame of data: continuePlaying(); } 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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