[Live-devel] Strange crashing "bug" (maybe?)

[Ben Rush]

Resolved. This is NOT a bug with Live555 but simply a misunderstanding of
how to use the Groupsocks. This is MY fault, NOT the Live555 code.

I should change the InitializeRoomClientAudioChannel method to use
Groupsock2 pointers instead of passing the address from the stack-based
ones. These addresses must be held onto internally, and obviously when the
stack unwinds the addresses are blown away. So the values of Groupsock
aren't copied by SimpleRTPSource::createNew, but the addresses are held
onto. Which is perfectly fine.

Sorry for the unnecessary post.

On Mon, Jun 27, 2016 at 10:57 AM Ben Rush <ben at ben-rush.net> wrote:

> This has happened to me a couple times while coding up things and I'd like
> to see if it's something stupid I'm doing, or whether I've uncovered a bug
> in Live555.
>
> I've got a class called "RoomClientStreamingServer" which does RTSP + RTP
> audio. Here is the header file for it:
>
> class RoomClientStreamingServer : public IRoomClient
> {
> public:
>     RoomClientStreamingServer();
>     ~RoomClientStreamingServer();
>     void InitializeServer();
>     bool InitializeRoomClientAudioChannel(int roomClientAudioListenPort);
>     bool InitializeRoomClientRTSPChannel(int rtspPort, std::string
> streamName, bool doAudio);
>     void StartServer();
>     virtual void StopServer() override;
> private:
>     TaskScheduler* _taskScheduler;
>     BasicUsageEnvironment* _usageEnvironment;
>
> private://audio channel
>     SpeakerSink* _speakerSink;
>     RTPSource* _audioRTPSource;
>     RTCPInstance* _audioRTCPSource;
>
> private://RTSP channel
>     RTSPServer* _rtspServer;
>     ServerMediaSession* _sms;
>     H264LiveServerMediaSession *_videoSubSession;
>     WindowsAudioMediaSession* _audioSubSession;
>     char _volatile;
> };
>
> I use it like this:
>
>     RoomClientStreamingServer* roomClient = new
> RoomClientStreamingServer();
>     roomClient->InitializeServer();
>
> roomClient->InitializeRoomClientAudioChannel(roomClientAudioListenPort);
>     roomClient->InitializeRoomClientRTSPChannel(rtspPort, streamName,
> doAudio);
>
>     roomClient->StartServer();
>
> The weirdness happens when calling StartServer. StartServer has a very
> simple implementation:
>
> void RoomClientStreamingServer::StartServer()
> {
>     _taskScheduler->doEventLoop(&_volatile);
> }
>
> This bombs with the following error:
>
> "BasicTaskScheduler::SingleStep(): select() fails: No error
> socket numbers used in the select() call: 812(r) 816(r)"
>
> It appears to be a problem with the InitializeRoomClientAudioChannel()
> call because I can duplicate this error if I comment out the
> InitializeRoomClientRTSPChannel().
>
> Here is the InitializeRoomClientAudioChannel() implementation:
>
>     _speakerSink = SpeakerSink::createNew(*_usageEnvironment, true,
> nullptr, false);
>
>     unsigned int rtpPortNum = roomClientAudioListenPort; //
> StreamingOptions::RoomClientListenPort;
>     unsigned int rtcpPortNum = rtpPortNum + 1;
>     char* ipAddress = "0.0.0.0";
>     //char* ipAddress = "239.255.42.42";
>
>     struct in_addr address;
>     address.S_un.S_addr = our_inet_addr(ipAddress);
>     const Port rtpPort(rtpPortNum);
>     const Port rtcpPort(rtcpPortNum);
>
>     Groupsock2 rtpGroupSock(*_usageEnvironment, address, rtpPort, 1);
>     Groupsock2 rtcpGroupSock(*_usageEnvironment, address, rtcpPort, 1);
>
>     _speakerSink->SetGroupSocks(&rtpGroupSock, &rtcpGroupSock);
>
>     //RTPSource* rtpSource = WaveFormDataStreamer::createNew(*environment,
> &rtpGroupSock);
>     int payloadFormatCode = 11;
>     const char* mimeType = "L16";
>     int fSamplingFrequency = 44100;
>     int fNumChannels = 1;
>     _audioRTPSource = SimpleRTPSource::createNew(
>         *_usageEnvironment, (Groupsock*)&rtpGroupSock, payloadFormatCode,
>         fSamplingFrequency, "audio/L16", 0, False /*no 'M' bit*/);
>
>     const unsigned maxCNAMElen = 100;
>     unsigned char CNAME[maxCNAMElen + 1];
>     gethostname((char*)CNAME, maxCNAMElen);
>     CNAME[maxCNAMElen] = '\0'; // just in case
>
>     _audioRTCPSource =
>         RTCPInstance::createNew(*_usageEnvironment,
> (Groupsock*)&rtcpGroupSock, 5000, CNAME, NULL, _audioRTPSource);
>     _audioRTCPSource->setByeHandler(subsessionByeHandler, _speakerSink);
>     _speakerSink->startPlaying(*_audioRTPSource, afterAudioListening,
> NULL);
>     //_taskScheduler->doEventLoop(&_volatile);
>     return true;
>
> If you'll notice I have _taskScheduler->doEventLoop(&_volatile);
> commented out because, if I uncomment this line, then I no longer get the
> crash.
>
> it's almost as if the _taskScheduler->doEventLoop() needs to be called on
> the same stack as the rest of the setup code. Again, I've encountered
> situations like this before when trying to separate the
> _taskScheduler->doEventLoop() call in another method.
>
> One final bit of information: if I comment out InitializeRoomClientAudioChannel
> and just let the RTSP stuff go (the webcam), with the
> _taskScheduler->doEventLoop() in its own method, then everything works
> fine.
>
> I recognize this might be a bit tough to follow over email, so if you need
> source that's fine. However, I also recognize people on this group are busy
> and so fundamentally I'd like to at least understand what might be the
> cause of the
>
> "BasicTaskScheduler::SingleStep(): select() fails: No error
> socket numbers used in the select() call: 812(r) 816(r)"
>
> error if nothing else.
>
> If I uncover what I'm doing before I get a response here then I'll send
> out another email to the group for posterity's sake.
>
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