From Daniel at smartcast.org Fri Sep 1 06:39:07 2006 From: Daniel at smartcast.org (Daniel Graupner) Date: Fri, 1 Sep 2006 15:39:07 +0200 Subject: [Live-devel] Basic Understanding Message-ID: <200609011539.07864.Daniel@smartcast.org> Hello, since there is no real documentation for the library, I want to describe my use case and my understanding of liveMedia. Please correct me if I am wrong. The program I want to use liveMedia for is a Multicast streaming server that allows to define an arbitrary number of streams that are started and stopped according to a specific time definition. The streaming part is done by liveMedia, we want to support "all" formats supported by liveMedia, for the first version we will focus on mp3. Some statements for using live libs in this project (please correct if wrong): - we have to derive our own UsageEnvironment to handle messages from the lib - the UsageEnvironment seems to be responsible for logging and has a unique instance of the TaskScheduler (the only one needed) - we need one Instance of our UsageEnvironment together with one TaskScheduler Instance - Input-Output-Chains connected to this environment can be startet and stopped in seperate threads, ?so I can create them at program start and start/stop them within their thread based on our time definition Some thoughts about the UsageEnvironment: Why are there different setResultMsg functions? Do they have different semantic meanings? What kind of messages are used together with this functions (INFO, WARNINg, CONFIG)? My understanding of error reporting is as follows: 1. special return values of function/method calls of the libs 2. setResultErrMsg(MsgString msg) and then reportBackgroundError() is called What is the result of an error? I need to know if the stream stops after this error, why the error occured, etc. How can I accomplish this? Aditionally to the setResultMsg methods there are the << operators, when are they used,what kind of output do they produce? OK, for our first version ?can find help in testMP3Streamer.cpp... - create a groupsock - create source and sink - start playing on sink Thanks for answers, Daniel. From Kirankumar.Csk at analog.com Fri Sep 1 06:40:33 2006 From: Kirankumar.Csk at analog.com (Csk, Kirankumar) Date: Fri, 1 Sep 2006 09:40:33 -0400 Subject: [Live-devel] RTP payload support for speex In-Reply-To: <7.0.1.0.1.20060830085156.020380e8@live555.com> Message-ID: <7D453D0504B6A2429F98F4D72CBEDE490B62AAAB@nwd2exm5.ad.analog.com> Hi, Does Live555 have support for RTP demuxing of Speex codec encoded streams? The draft of the RTP packetization can be found at: http://www.croczilla.com/zap/rfcs/draft-herlein-speex-rtp-profile-02.txt Thanks in advance! -Kiran From bigboy323 at 163.com Fri Sep 1 20:14:46 2006 From: bigboy323 at 163.com (yxx) Date: Sat, 2 Sep 2006 11:14:46 +0800 Subject: [Live-devel] How to stream h264 raw file by livemedia Message-ID: <44F8F72B.0540AC.13630> Hi: How to stream h264 raw file by livemedia. Thanks. ????????yxx ????????bigboy323 at 163.com ??????????2006-09-02 From Kirankumar.Csk at analog.com Sat Sep 2 01:02:44 2006 From: Kirankumar.Csk at analog.com (Csk, Kirankumar) Date: Sat, 2 Sep 2006 04:02:44 -0400 Subject: [Live-devel] RTP payload support for speex In-Reply-To: <7D453D0504B6A2429F98F4D72CBEDE490B62AAAB@nwd2exm5.ad.analog.com> Message-ID: <7D453D0504B6A2429F98F4D72CBEDE490B6CF093@nwd2exm5.ad.analog.com> Hi, I also have the same question for G.729 codec. Can Live555 support demuxing of G.729 RTP payload? Thanks! Kiran -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Csk, Kirankumar Sent: Friday, September 01, 2006 7:11 PM To: live-devel at ns.live555.com Subject: [Live-devel] RTP payload support for speex Hi, Does Live555 have support for RTP demuxing of Speex codec encoded streams? The draft of the RTP packetization can be found at: http://www.croczilla.com/zap/rfcs/draft-herlein-speex-rtp-profile-02.txt Thanks in advance! -Kiran _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel From barounis at ceid.upatras.gr Sat Sep 2 08:56:45 2006 From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr) Date: Sat, 2 Sep 2006 18:56:45 +0300 Subject: [Live-devel] Jitter calculation Message-ID: <1157212605.44f9a9bd542b8@my.ceid.upatras.gr> Hello to Ross and all members, just a small question... I'm receiving RTCP RR packets and I need to make some plots using the information included in these packets. Concerning the jitter, what kind of calculation should I make in order to convert the current value (unsigned integer) to time units ? I'm using RTSP Server making a wireless streaming over UMTS, with MPEG-2 Elementary Stream video files. Thank you very much Best regards ---------------------------------------------------- This mail was sent through http://my.ceid.upatras.gr From finlayson at live555.com Sun Sep 3 18:28:04 2006 From: finlayson at live555.com (Ross Finlayson) Date: Sun, 3 Sep 2006 18:28:04 -0700 Subject: [Live-devel] Basic Understanding In-Reply-To: <200609011539.07864.Daniel@smartcast.org> References: <200609011539.07864.Daniel@smartcast.org> Message-ID: >Some statements for using live libs in this project (please correct if wrong): >- we have to derive our own UsageEnvironment to handle messages from the lib Not necessarly. For many (most?) developers, the supplied subclass "BasicUsageEnvironment" (which outputs diagnostic/error messages to stderr) is sufficient. >- the UsageEnvironment seems to be responsible for logging and has a unique >instance of the TaskScheduler (the only one needed) Yes. >- we need one Instance of our UsageEnvironment together with one TaskScheduler >Instance Yes. >- Input-Output-Chains connected to this environment can be startet and stopped >in seperate threads No! Please read the FAQ! >Some thoughts about the UsageEnvironment: >Why are there different setResultMsg functions? Do they have different >semantic meanings? No, but they have different parameter signatures (the number of strings). If you're confused about how these are used, then I suggest looking at various places within the code where they are used. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From rajuec at rediffmail.com Mon Sep 4 00:08:05 2006 From: rajuec at rediffmail.com (rajesh tiwary) Date: 4 Sep 2006 07:08:05 -0000 Subject: [Live-devel] How to use rtsp url in openrtsp exe Message-ID: <20060904070805.1123.qmail@webmail62.rediffmail.com> ?HI ALL, I got list of RTSP URL after running testonDemandRTSPServer. Now I want to play this URL through the openRTSP. when I am trying ,I am getting this Screen dump---> //###SCREEN LOGS after runnign mpeg4ESVideoTest### openrtsp rtsp://10.0.1.248:8554/mpeg4ESVideoTest Failed to get a SDP description from URL "rtsp://10.0.1.248:8554/mpeg4ESVideoTes t": connect() failed: Unknown error with regards rajesh rajesh kumar software Engineer(Telecom) (91)9949902510 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060904/ee30f761/attachment.html From dbikash at gmail.com Mon Sep 4 02:18:15 2006 From: dbikash at gmail.com (Deeptendu Bikash) Date: Mon, 4 Sep 2006 14:48:15 +0530 Subject: [Live-devel] How to use rtsp url in openrtsp exe In-Reply-To: <20060904070805.1123.qmail@webmail62.rediffmail.com> References: <20060904070805.1123.qmail@webmail62.rediffmail.com> Message-ID: <29758ec70609040218y5b51f3d3udb1d191f45b46114@mail.gmail.com> The error says 'connect failed'. So its probably a network error. Check your network connections and try again. On 4 Sep 2006 07:08:05 -0000, rajesh tiwary wrote: > > HI ALL, > > I got list of RTSP URL after running testonDemandRTSPServer. > Now I want to play this URL through the openRTSP. > when I am trying ,I am getting this Screen dump---> > > //###SCREEN LOGS after runnign mpeg4ESVideoTest### > openrtsp rtsp://10.0.1.248:8554/mpeg4ESVideoTest > Failed to get a SDP description from URL > "rtsp://10.0.1.248:8554/mpeg4ESVideoTes > t": connect() failed: Unknown error > > with regards > rajesh > > rajesh kumar > software Engineer(Telecom) > (91)9949902510 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060904/6908035f/attachment-0001.html From lorenooliveira at gmail.com Mon Sep 4 06:08:09 2006 From: lorenooliveira at gmail.com (Loreno Oliveira) Date: Mon, 4 Sep 2006 10:08:09 -0300 Subject: [Live-devel] How to use rtsp url in openrtsp exe In-Reply-To: <29758ec70609040218y5b51f3d3udb1d191f45b46114@mail.gmail.com> References: <20060904070805.1123.qmail@webmail62.rediffmail.com> <29758ec70609040218y5b51f3d3udb1d191f45b46114@mail.gmail.com> Message-ID: Did you put the media files in the same directory as testOnDemandRTSPServer? For the stream you are trying to connect to you should have a mpeg4 file called test.m4v. Loreno On 9/4/06, Deeptendu Bikash wrote: > > The error says 'connect failed'. So its probably a network error. Check > your network connections and try again. > > > On 4 Sep 2006 07:08:05 -0000, rajesh tiwary wrote: > > > > HI ALL, > > > > I got list of RTSP URL after running testonDemandRTSPServer. > > Now I want to play this URL through the openRTSP. > > when I am trying ,I am getting this Screen dump---> > > > > //###SCREEN LOGS after runnign mpeg4ESVideoTest### > > openrtsp rtsp://10.0.1.248:8554/mpeg4ESVideoTest > > Failed to get a SDP description from URL > > "rtsp://10.0.1.248:8554/mpeg4ESVideoTes > > t": connect() failed: Unknown error > > > > with regards > > rajesh > > > > rajesh kumar > > software Engineer(Telecom) > > (91)9949902510 > > > > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060904/95ff57e2/attachment.html From ishan73 at yahoo.com Mon Sep 4 06:42:10 2006 From: ishan73 at yahoo.com (Ishan Vaishnavi) Date: Mon, 4 Sep 2006 14:42:10 +0100 (BST) Subject: [Live-devel] real servers In-Reply-To: Message-ID: <20060904134210.39977.qmail@web35601.mail.mud.yahoo.com> Hello all, I can see from the live page that live doesnot support real servers, beacause real servers use RDT for transport . However unintentionally I did stream a mp3 file from an open source real server and it worked. I just wanted to know what is the clear stand on this. Is it that live may work with real servers if I am lucky or is it that it works for certain file types but has issues with others? In the latter case do we a table which shows clearly with what formats it works and what formats don't ? Cheers Ishan ___________________________________________________________ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html From finlayson at live555.com Mon Sep 4 08:11:17 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 4 Sep 2006 08:11:17 -0700 Subject: [Live-devel] Jitter calculation In-Reply-To: <1157212605.44f9a9bd542b8@my.ceid.upatras.gr> References: <1157212605.44f9a9bd542b8@my.ceid.upatras.gr> Message-ID: >Hello to Ross and all members, > >just a small question... > >I'm receiving RTCP RR packets and I need to make some plots using the >information included in these packets. > >Concerning the jitter, what kind of calculation should I make in order to >convert the current value (unsigned integer) to time units ? The "jitter" value is as defined in RFC 3550, section 6.4.1. So, divide it by the RTP timestamp frequency to get a time value. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Mon Sep 4 08:24:17 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 4 Sep 2006 08:24:17 -0700 Subject: [Live-devel] RTP payload support for speex In-Reply-To: <7D453D0504B6A2429F98F4D72CBEDE490B62AAAB@nwd2exm5.ad.analog.com> References: <7D453D0504B6A2429F98F4D72CBEDE490B62AAAB@nwd2exm5.ad.analog.com> Message-ID: >Hi, >Does Live555 have support for RTP demuxing of Speex codec encoded >streams? Not specifically. However, because the Speex RTP payload format is so simple, it can probably be implemented using "SimpleRTPSink" (for sending) and "SimpleRTPSource" (for receiving). For guidance, see where those classes are used in the code. (In fact, if you look at "liveMedia/MediaSession.cpp", you'll see that receiving Speex is alreadyh supported.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Mon Sep 4 08:25:25 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 4 Sep 2006 08:25:25 -0700 Subject: [Live-devel] RTP payload support for speex In-Reply-To: <7D453D0504B6A2429F98F4D72CBEDE490B6CF093@nwd2exm5.ad.analog.com> References: <7D453D0504B6A2429F98F4D72CBEDE490B6CF093@nwd2exm5.ad.analog.com> Message-ID: >I also have the same question for G.729 codec. Can Live555 support >demuxing of G.729 RTP payload? Yes, in the same way as for Speex - using "SimpleRTPSink" (for sending) and "SimpleRTPSource" (for receiving). Note that we already support receiving G.729 RTP streams (see "liveMedia/MediaSession.cpp"). See RFC 3551 for more information about the RTP payload format for G.729. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Mon Sep 4 08:29:30 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 4 Sep 2006 08:29:30 -0700 Subject: [Live-devel] real servers In-Reply-To: <20060904134210.39977.qmail@web35601.mail.mud.yahoo.com> References: <20060904134210.39977.qmail@web35601.mail.mud.yahoo.com> Message-ID: >I can see from the live page that live doesnot support >real servers, beacause real servers use RDT for >transport . However unintentionally I did stream a mp3 >file from an open source real server and it worked. That's because this stream must have used standard RTP, not RDT. Receiving from RealNetworks servers will usually work provided that the stream is not using RealNetworks' proprietary codecs. Those are usually the only streams that get sent using RDT. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From poli_moraes at yahoo.com.br Mon Sep 4 12:32:26 2006 From: poli_moraes at yahoo.com.br (Poliana Moraes) Date: Mon, 4 Sep 2006 16:32:26 -0300 (ART) Subject: [Live-devel] port Live 555 to Blackfin uClinux Message-ID: <20060904193226.11031.qmail@web60624.mail.yahoo.com> Hello! My name is Poliana. I ported live 555 to Blackfin uCLinux. How can I submit it to the project? I create a file called config.bfin_uclinux in order to configure the library to blackfin: CROSS_COMPILER= bfin-uclinux- COMPILE_OPTS = $(INCLUDES) -I. -DSOCKLEN_T=socklen_t -D_LARGEFILE_SOURCE=1 -DUCLINUX C = c C_COMPILER = $(CROSS_COMPILER)gcc C_FLAGS = $(COMPILE_OPTS) -Wall CPP = cpp CPLUSPLUS_COMPILER = $(CROSS_COMPILER)g++ CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -Wall OBJ = o LINK = $(CROSS_COMPILER)g++ -Wl,-elf2flt -o LINK_OPTS = -L. CONSOLE_LINK_OPTS = $(LINK_OPTS) LIBRARY_LINK = $(CROSS_COMPILER)ld -o LIBRARY_LINK_OPTS = $(LINK_OPTS) -r -Bstatic LIB_SUFFIX = a LIBS_FOR_CONSOLE_APPLICATION = LIBS_FOR_GUI_APPLICATION = EXE = I edited the file GroupsockHelper.cpp $ diff ./groupsock/GroupsockHelper.cpp ../live_original/groupsock/GroupsockHelper.cpp 33,34d32 < #include < #include 553,573d550 < #ifdef UCLINUX < netAddressBits ourSourceAddressForMulticastUclinux(UsageEnvironment& env) { < int i=2; < int s = socket (PF_INET, SOCK_STREAM, 0); < < < struct ifreq ifr; < struct sockaddr_in *sin = (struct sockaddr_in *) &ifr.ifr_addr; < char *ip; < < ifr.ifr_ifindex = i; < ioctl (s, SIOCGIFNAME, &ifr); < < < /* now ifr.ifr_name is set */ < ioctl (s, SIOCGIFADDR, &ifr); < < return (sin->sin_addr.s_addr); < } < #endif < 575,577d551 < #ifdef UCLINUX < return ourSourceAddressForMulticastUclinux(env); < #endif Thanks in advanced! ------------------------------------------------------------- Poliana de Moraes cel: (12)9155-9981 4? ano Engenharia El?trica UNESP - campus de Guaratinguet? --------------------------------- Novidade no Yahoo! Mail: receba alertas de novas mensagens no seu celular. Registre seu aparelho agora! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060904/b2138b31/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: config.bfin_uclinux Type: application/octet-stream Size: 560 bytes Desc: pat73094434 Url : http://lists.live555.com/pipermail/live-devel/attachments/20060904/b2138b31/attachment-0001.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: GroupsockHelper.cpp Type: text/x-c++src Size: 22704 bytes Desc: pat307279412 Url : http://lists.live555.com/pipermail/live-devel/attachments/20060904/b2138b31/attachment-0001.bin From barounis at ceid.upatras.gr Mon Sep 4 13:10:14 2006 From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr) Date: Mon, 4 Sep 2006 23:10:14 +0300 Subject: [Live-devel] Jitter calculation In-Reply-To: References: <1157212605.44f9a9bd542b8@my.ceid.upatras.gr> Message-ID: <1157400614.44fc8826f0b04@my.ceid.upatras.gr> Hello and thanks for the response.. > The "jitter" value is as defined in RFC 3550, section 6.4.1. So, > divide it by the RTP timestamp frequency to get a time value. > -- > I have found that the frequency for MPEG video is 90KHz. So I divide the jitter value with 90 and the result is the jitter expressed in microseconds [10^(-6) seconds] ? Could you confirm, if what I have found is correct ? Thank you very much ---------------------------------------------------- This mail was sent through http://my.ceid.upatras.gr From finlayson at live555.com Mon Sep 4 17:04:11 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 4 Sep 2006 17:04:11 -0700 Subject: [Live-devel] Jitter calculation In-Reply-To: <1157400614.44fc8826f0b04@my.ceid.upatras.gr> References: <1157212605.44f9a9bd542b8@my.ceid.upatras.gr> <1157400614.44fc8826f0b04@my.ceid.upatras.gr> Message-ID: >I have found that the frequency for MPEG video is 90KHz. So I divide >the jitter >value with 90 and the result is the jitter expressed in microseconds >[10^(-6) seconds] ? > >Could you confirm, if what I have found is correct ? No, if you divide the jitter number by 90, you'll get the jitter in *milli*seconds. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From chenw at blrcsv.china.bell-labs.com Mon Sep 4 19:16:34 2006 From: chenw at blrcsv.china.bell-labs.com (chenwei) Date: Tue, 5 Sep 2006 10:16:34 +0800 Subject: [Live-devel] Why no video after reuseFirstSource = True Message-ID: <20060905020322.BEFDB8719@blrcsv.china.bell-labs.com> Dear All; I want use testOnDemandRTSPServer to stream a live source and according to FAQ,I set the variable "reuseFirstSource" to "True", but only audio was streamed. After I set the variable back to "False" ,video and audio were both streamed. I have analysed rtsp packets,nothing found:( ????????? ?? ????????chenwei ????????chenw at blrcsv.china.bell-labs.com ??????????2006-09-05 From rajuec at rediffmail.com Mon Sep 4 21:46:28 2006 From: rajuec at rediffmail.com (rajesh tiwary) Date: 5 Sep 2006 04:46:28 -0000 Subject: [Live-devel] How to use rtsp url in openrtsp exe Message-ID: <20060905044628.12857.qmail@webmail69.rediffmail.com> ? HI Loreno nice to see ur reply. test.m4v is present in the directory where testOnDemandRTSPServer.exe is present.test.m4v is of type MPEG-4 Movie. with regards rajesh On Mon, 04 Sep 2006 Loreno Oliveira wrote : >Did you put the media files in the same directory as testOnDemandRTSPServer? >For the stream you are trying to connect to you should have a mpeg4 file >called test.m4v. > >Loreno > >On 9/4/06, Deeptendu Bikash wrote: >> >>The error says 'connect failed'. So its probably a network error. Check >>your network connections and try again. >> >> >>On 4 Sep 2006 07:08:05 -0000, rajesh tiwary wrote: >> > >> > HI ALL, >> > >> > I got list of RTSP URL after running testonDemandRTSPServer. >> > Now I want to play this URL through the openRTSP. >> > when I am trying ,I am getting this Screen dump---> >> > >> > //###SCREEN LOGS after runnign mpeg4ESVideoTest### >> > openrtsp rtsp://10.0.1.248:8554/mpeg4ESVideoTest >> > Failed to get a SDP description from URL >> > "rtsp://10.0.1.248:8554/mpeg4ESVideoTes >> > t": connect() failed: Unknown error >> > >> > with regards >> > rajesh >> > >> > rajesh kumar >> > software Engineer(Telecom) >> > (91)9949902510 >> > >> >> >>_______________________________________________ >>live-devel mailing list >>live-devel at lists.live555.com >>http://lists.live555.com/mailman/listinfo/live-devel >> >> >> >_______________________________________________ >live-devel mailing list >live-devel at lists.live555.com >http://lists.live555.com/mailman/listinfo/live-devel rajesh kumar software Engineer(Telecom) (91)9949902510 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060904/172b3dc4/attachment.html From rajuec at rediffmail.com Mon Sep 4 21:57:05 2006 From: rajuec at rediffmail.com (rajesh tiwary) Date: 5 Sep 2006 04:57:05 -0000 Subject: [Live-devel] How to use rtsp url in openrtsp exe Message-ID: <20060905045705.27457.qmail@webmail69.rediffmail.com> ? Hi I am connected with the Net also.but as RTSP server I am using the rtsp url given by testOnDemandRTSPServer.exe with regards rajesh On Mon, 04 Sep 2006 Deeptendu Bikash wrote : >The error says 'connect failed'. So its probably a network error. Check your >network connections and try again. > >On 4 Sep 2006 07:08:05 -0000, rajesh tiwary wrote: >> >> HI ALL, >> >>I got list of RTSP URL after running testonDemandRTSPServer. >>Now I want to play this URL through the openRTSP. >>when I am trying ,I am getting this Screen dump---> >> >>//###SCREEN LOGS after runnign mpeg4ESVideoTest### >>openrtsp rtsp://10.0.1.248:8554/mpeg4ESVideoTest >>Failed to get a SDP description from URL >>"rtsp://10.0.1.248:8554/mpeg4ESVideoTes >>t": connect() failed: Unknown error >> >>with regards >>rajesh >> >> rajesh kumar >>software Engineer(Telecom) >>(91)9949902510 >> >_______________________________________________ >live-devel mailing list >live-devel at lists.live555.com >http://lists.live555.com/mailman/listinfo/live-devel rajesh kumar software Engineer(Telecom) (91)9949902510 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060904/0f40c4f3/attachment.html From finlayson at live555.com Tue Sep 5 01:21:54 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 5 Sep 2006 01:21:54 -0700 Subject: [Live-devel] Why no video after reuseFirstSource = True In-Reply-To: <20060905020322.BEFDB8719@blrcsv.china.bell-labs.com> References: <20060905020322.BEFDB8719@blrcsv.china.bell-labs.com> Message-ID: >Dear All; > I want use testOnDemandRTSPServer to stream a live >source and according to FAQ,I set the variable "reuseFirstSource" > >to "True", but only audio was streamed. After I set the variable >back to "False" ,video and audio were both streamed. I don't know what would cause this. What audio and video codecs were you using, and what sort of input were you using? E.g., a MPEG-1 or 2 Program Stream? Or something else? -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From cs1030161 at gmail.com Tue Sep 5 02:11:35 2006 From: cs1030161 at gmail.com (Dipankar Sarkar) Date: Tue, 5 Sep 2006 14:41:35 +0530 Subject: [Live-devel] Streaming ondemand client Message-ID: <521f77f10609050211i5b66e0a0j788b044ce12d1a7a@mail.gmail.com> We have to build this application which grabs frames from an MPEG4 stream "on demand" (that is it should return the most current frame when queried for) ... We have checked out the openRTSP sample code, but cannot figure out how exaclty to grab a single frame. Thanks in advance for any help Dipankar From lorenooliveira at gmail.com Tue Sep 5 05:06:07 2006 From: lorenooliveira at gmail.com (Loreno Oliveira) Date: Tue, 5 Sep 2006 04:06:07 -0800 Subject: [Live-devel] How to use rtsp url in openrtsp exe In-Reply-To: <20060905044628.12857.qmail@webmail69.rediffmail.com> References: <20060905044628.12857.qmail@webmail69.rediffmail.com> Message-ID: In this case, try to verify if you can reach the IP/port at the server host from the client host. I would use telnet client for testing this. I`m not sure about the telnet syntax on windows, but it should be something like this: telnet 10.0.1.248 8554 You type this in the client host and check if telnet can connects to the server host. If it can not, so there is no process listening on the port 8554 in the host 10.0.1.248, or you are experiencig some sort of firewall or routing trouble. Loreno On 5 Sep 2006 04:46:28 -0000, rajesh tiwary wrote: > > > HI Loreno > nice to see ur reply. > test.m4v is present in the directory where testOnDemandRTSPServer.exe is > present.test.m4v is of type MPEG-4 Movie. > > with regards > rajesh > > > > On Mon, 04 Sep 2006 Loreno Oliveira wrote : > >Did you put the media files in the same directory as > testOnDemandRTSPServer? > >For the stream you are trying to connect to you should have a mpeg4 file > >called test.m4v. > > > >Loreno > > > >On 9/4/06, Deeptendu Bikash wrote: > >> > >>The error says 'connect failed'. So its probably a network error. Check > >>your network connections and try again. > >> > >> > >>On 4 Sep 2006 07:08:05 -0000, rajesh tiwary > wrote: > >> > > >> > HI ALL, > >> > > >> > I got list of RTSP URL after running testonDemandRTSPServer. > >> > Now I want to play this URL through the openRTSP. > >> > when I am trying ,I am getting this Screen dump---> > >> > > >> > //###SCREEN LOGS after runnign mpeg4ESVideoTest### > >> > openrtsp rtsp://10.0.1.248:8554/mpeg4ESVideoTest > >> > Failed to get a SDP description from URL > >> > "rtsp://10.0.1.248:8554/mpeg4ESVideoTes > >> > t": connect() failed: Unknown error > >> > > >> > with regards > >> > rajesh > >> > > >> > rajesh kumar > >> > software Engineer(Telecom) > >> > (91)9949902510 > >> > > >> > >> > >>_______________________________________________ > >>live-devel mailing list > >>live-devel at lists.live555.com > >>http://lists.live555.com/mailman/listinfo/live-devel > >> > >> > >> > >_______________________________________________ > >live-devel mailing list > >live-devel at lists.live555.com > >http://lists.live555.com/mailman/listinfo/live-devel > rajesh kumar > software Engineer(Telecom) > (91)9949902510 > > > > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060905/ebe20ede/attachment-0001.html From rajeshkumar.r at imimobile.com Tue Sep 5 06:00:52 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Tue, 5 Sep 2006 18:30:52 +0530 Subject: [Live-devel] testOnDemandRTSPServer is not starting second time when in first attemp trying to play video Message-ID: <004801c6d0eb$515b0790$f801000a@imidomain.com> Hi All I am using "TestOnDemandRTSPServer.exe" in that there is rtsp url for the play rtsp://10.0.1.248:8554/mpeg1or2AudioVideoTest. This Url I am using in VLC in rtsp Url. test.mpg is present in testProg directory. when I tried to play the streaming thru VLC the TestOnDemandRTSPServer.exe stopped . Next time when i tried to start TestOnDemandRTSPServer.exe,Its Not starting and giving error belo // "D:\Projects\VIDEO_PORTAL\VideoStreaming\live\testProgs>testOnDemandRTSPServer "mpeg4ESVideoTest" stream, from the file "test.m4v" Play this stream using the URL "rtsp://10.0.1.248:8554/mpeg4ESVideoTest" BasicTaskScheduler::SingleStep(): select() fails: No error // please help me in this regard with regards rajesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060905/ad619ef6/attachment.html From finlayson at live555.com Tue Sep 5 19:22:13 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 5 Sep 2006 19:22:13 -0700 Subject: [Live-devel] port Live 555 to Blackfin uClinux In-Reply-To: <20060904193226.11031.qmail@web60624.mail.yahoo.com> References: <20060904193226.11031.qmail@web60624.mail.yahoo.com> Message-ID: >Hello! > >My name is Poliana. I ported live 555 to Blackfin uCLinux. How can I >submit it to the project? > >I create a file called config.bfin_uclinux in order to configure >the library to blackfin: Thanks. I will include your new "config.bfin_uclinux" file in the next release of the software. However, right now I don't think that I will include your change to "GroupsockHelper.cpp", because it is specific to one particular OS. (I'm also not sure how it is supposed to work. In particular, what is the significance of "i=2"?) I suggest that you instead try to get the original code to work on your system. Often, this requires only that you add a route for or 224.0.0/4 to your routing tables - so that multicast will work. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From shrinivas_balulad at yahoo.co.in Tue Sep 5 22:15:02 2006 From: shrinivas_balulad at yahoo.co.in (shrinivas balulad) Date: Wed, 6 Sep 2006 06:15:02 +0100 (BST) Subject: [Live-devel] Is GSM codec supported in OpenRTSP? Message-ID: <20060906051505.94209.qmail@web7608.mail.in.yahoo.com> Hi, I am using OpenRTSP project to wirte video to a file in quick time format. This works fine for ulaw and alaw ( video is encoded using H263+). But for GSM codec, i am not able to here anything when i play the file. I am playing the video file using VLC media player. I am able to view the video but not able here any audio. I think GSM codec is supported in OpenRTSP. Please suggest me what is the problem. Thanks in advance Shrinivas --------------------------------- Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060905/d8870a2d/attachment.html From finlayson at live555.com Tue Sep 5 22:27:14 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 5 Sep 2006 22:27:14 -0700 Subject: [Live-devel] Is GSM codec supported in OpenRTSP? In-Reply-To: <20060906051505.94209.qmail@web7608.mail.in.yahoo.com> References: <20060906051505.94209.qmail@web7608.mail.in.yahoo.com> Message-ID: >I am using OpenRTSP project to wirte video to a file in quick time >format. This works fine for ulaw and alaw ( video is encoded using >H263+). But for GSM codec, i am not able to here anything when i >play the file. I am playing the video file using VLC media player. I >am able to view the video but not able here any audio. > >I think GSM codec is supported in OpenRTSP. >Please suggest me what is the problem. I don't know. GSM is supported in "QuickTimeFileSink", and so should work. Perhaps, in this case, the media player is the problem? You could try using QuickTime Player instead of VLC. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From chenw at blrcsv.china.bell-labs.com Wed Sep 6 02:22:15 2006 From: chenw at blrcsv.china.bell-labs.com (chenwei) Date: Wed, 6 Sep 2006 17:22:15 +0800 Subject: [Live-devel] why can't testOnDemandRTSPServer stream the video file gernerated by openRTSP? Message-ID: <20060906090859.738A68723@blrcsv.china.bell-labs.com> Dear All: I run openRTSP and received a mpeg1 ps stream from a Darwin server. As we know, one video and one audio file were generated. I want to test testOnDemandRTSPServer, so I change the name of test.mpg to video-MPV-1, But it cannot be streamed! I found after the client sent the "PLAY" RTSP packet, the server sent a "Goodbye" RTP packet. Why? ????????? ?? ????????chenwei ????????chenw at blrcsv.china.bell-labs.com ??????????2006-09-06 From finlayson at live555.com Wed Sep 6 02:32:28 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 6 Sep 2006 02:32:28 -0700 Subject: [Live-devel] why can't testOnDemandRTSPServer stream the video file gernerated by openRTSP? In-Reply-To: <20060906090859.738A68723@blrcsv.china.bell-labs.com> References: <20060906090859.738A68723@blrcsv.china.bell-labs.com> Message-ID: >Dear All: > I run openRTSP and received a mpeg1 ps stream from a >Darwin server. As we know, one video and one audio file were > >generated. > > I want to test testOnDemandRTSPServer, so I change >the name of test.mpg to video-MPV-1 I think (hope) you meant to say that you changed the name of "video-MPV-1" to "test.mpg". But even that won't work (see below). >But it cannot be streamed! The problem is that the file "video-MPV-1" that openRTSP recorded is a MPEG Video *Elementary* Stream - i.e., a video-only stream - *not* a MPEG Program Stream. To stream this using "testOnDemandRTSPServer", you will need to rename it as "testv.mpg", and use the "rtsp:// ... /mpeg1or2ESVideoTest" URL. Then, it should work. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From mandelfi at free.fr Wed Sep 6 08:48:15 2006 From: mandelfi at free.fr (Mandelfi) Date: Wed, 06 Sep 2006 17:48:15 +0200 Subject: [Live-devel] File size with OpenRTSP Message-ID: <44FEEDBF.1020407@free.fr> Hi all, I am using OpenRTSP like this on my Linux box: openRTSP -F my_file rtsp://my_url to capture a streaming video program. This works very good but the file cannot be bigger than 2Go. Why ? I tried to use different filesystems like ext3 and reiserfs, it is the same... Thanks in advance Adrien From finlayson at live555.com Wed Sep 6 09:03:13 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 6 Sep 2006 09:03:13 -0700 Subject: [Live-devel] File size with OpenRTSP In-Reply-To: <44FEEDBF.1020407@free.fr> References: <44FEEDBF.1020407@free.fr> Message-ID: >Hi all, > >I am using OpenRTSP like this on my Linux box: > >openRTSP -F my_file rtsp://my_url > >to capture a streaming video program. > >This works very good but the file cannot be bigger than 2Go. Why ? I don't know. The file output code (to implement the "FileSink" class) uses the standard routines "fopen()" and "fwrite()" (and doesn't try to seek at all within the file), so I don't know why the output file would get restricted to 2**31 bytes in size, if your OS supports larger sizes. What version of Linux are you using? Ross. From ishan73 at yahoo.com Wed Sep 6 11:33:45 2006 From: ishan73 at yahoo.com (Ishan Vaishnavi) Date: Wed, 6 Sep 2006 19:33:45 +0100 (BST) Subject: [Live-devel] File size with OpenRTSP In-Reply-To: Message-ID: <20060906183345.70073.qmail@web35603.mail.mud.yahoo.com> You are hitting standard unix limits for 32 bit machines I think, since 2^31 = 2 G. A lot of programs have 2 G limitation for example zip - unzip. HTH Ishan --- Ross Finlayson wrote: > >Hi all, > > > >I am using OpenRTSP like this on my Linux box: > > > >openRTSP -F my_file rtsp://my_url > > > >to capture a streaming video program. > > > >This works very good but the file cannot be bigger > than 2Go. Why ? > > I don't know. The file output code (to implement > the "FileSink" > class) uses the standard routines "fopen()" and > "fwrite()" (and > doesn't try to seek at all within the file), so I > don't know why the > output file would get restricted to 2**31 bytes in > size, if your OS > supports larger sizes. What version of Linux are > you using? > > Ross. > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > ___________________________________________________________ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html From finlayson at live555.com Wed Sep 6 17:12:08 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 6 Sep 2006 17:12:08 -0700 Subject: [Live-devel] File size with OpenRTSP In-Reply-To: <20060906183345.70073.qmail@web35603.mail.mud.yahoo.com> References: <20060906183345.70073.qmail@web35603.mail.mud.yahoo.com> Message-ID: >You are hitting standard unix limits for 32 bit >machines I think, since 2^31 = 2 G. However, modern operating systems can support files greater than 2^31 (or even 2^32) in size, even on 32-bit computers. That's why I asked what version of Linux the original questioner was using. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From zhangzx at RCS-9000.COM Wed Sep 6 20:34:04 2006 From: zhangzx at RCS-9000.COM (=?ISO-8859-1?Q?=D5=C5=D6=BE=D1=A7?=) Date: Thu, 7 Sep 2006 11:34:04 +0800 Subject: [Live-devel] RTP over TCP Message-ID: <200609071134.AA50266344@RCS-9000.COM> I test "-t" option of openrtsp. It seems that after the client has setup and played session, "RTSPClientSession::incomingRequestHandler1()" never be called in server. So "noteLiveness()" will not be called. And serversession will be stoped and deleted after fReclamationTestSeconds seconds. thanks for your reply From rajeshkumar.r at imimobile.com Wed Sep 6 21:38:36 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 10:08:36 +0530 Subject: [Live-devel] unable to do Video Streaming References: <20060906051505.94209.qmail@web7608.mail.in.yahoo.com> Message-ID: <006001c6d237$7bd83900$f801000a@imidomain.com> HI I am unable to do video streaming thru "testOnDemandRTSPServer".I am using client as VLC.In VLC ,I am mentioning RTSP url mentioned in the "testOnDemandRTSPServer".when VLC is sending the Request for the RTSP url, the error is below mentioned and "testOnDemandRTSP server is not stopping and next time also when I am running the same error is coming. The Error is -> "mpeg4ESVideoTest" stream, from the file "test.m4v" Play this stream using the URL "rtsp://10.0.1.248:8554/mpeg4ESVideoTest" BasicTaskScheduler::SingleStep(): select() fails: No error ----- Original Message ----- From: shrinivas balulad To: live-devel at ns.live555.com Sent: Wednesday, September 06, 2006 10:45 AM Subject: [Live-devel] Is GSM codec supported in OpenRTSP? Hi, I am using OpenRTSP project to wirte video to a file in quick time format. This works fine for ulaw and alaw ( video is encoded using H263+). But for GSM codec, i am not able to here anything when i play the file. I am playing the video file using VLC media player. I am able to view the video but not able here any audio. I think GSM codec is supported in OpenRTSP. Please suggest me what is the problem. Thanks in advance Shrinivas ------------------------------------------------------------------------------ Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060906/8369a0b7/attachment-0001.html From rajeshkumar.r at imimobile.com Wed Sep 6 22:02:00 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 10:32:00 +0530 Subject: [Live-devel] plz help me to understand the cause of the error Message-ID: <009201c6d23a$c02d7ae0$f801000a@imidomain.com> HI All I am receiving one error BasicTaskScheduler::SingleStep() function The error is --> perror("BasicTaskScheduler::SingleStep(): select() fails"); ///////////////////////////////////////////////////////////////////////////////////////// void BasicTaskScheduler::SingleStep(unsigned maxDelayTime) { fd_set readSet = fReadSet; // make a copy for this select() call DelayInterval const& timeToDelay = fDelayQueue.timeToNextAlarm(); struct timeval tv_timeToDelay; tv_timeToDelay.tv_sec = timeToDelay.seconds(); tv_timeToDelay.tv_usec = timeToDelay.useconds(); // Very large "tv_sec" values cause select() to fail. // Don't make it any larger than 1 million seconds (11.5 days) const long MAX_TV_SEC = MILLION; if (tv_timeToDelay.tv_sec > MAX_TV_SEC) { tv_timeToDelay.tv_sec = MAX_TV_SEC; } // Also check our "maxDelayTime" parameter (if it's > 0): if (maxDelayTime > 0 && (tv_timeToDelay.tv_sec > (long)maxDelayTime/MILLION || (tv_timeToDelay.tv_sec == (long)maxDelayTime/MILLION && tv_timeToDelay.tv_usec > (long)maxDelayTime%MILLION))) { tv_timeToDelay.tv_sec = maxDelayTime/MILLION; tv_timeToDelay.tv_usec = maxDelayTime%MILLION; } int selectResult = select(fMaxNumSockets, &readSet, NULL, NULL, &tv_timeToDelay); if (selectResult < 0) { #if defined(__WIN32__) || defined(_WIN32) int err = WSAGetLastError(); // For some unknown reason, select() in Windoze sometimes fails with WSAEINVAL if // it was called with no entries set in "readSet". If this happens, ignore it: if (err == WSAEINVAL && readSet.fd_count == 0) { err = 0; // To stop this from happening again, create a dummy readable socket: int dummySocketNum = socket(AF_INET, SOCK_DGRAM, 0); FD_SET((unsigned)dummySocketNum, &fReadSet); } if (err != 0) { #else if (errno != EINTR && errno != EAGAIN) { #endif // Unexpected error - treat this as fatal: #if !defined(_WIN32_WCE) perror("BasicTaskScheduler::SingleStep(): select() fails"); #endif exit(0); } } // Handle any delayed event that may have come due: fDelayQueue.handleAlarm(); // Call the handler function for one readable socket: HandlerIterator iter(*fReadHandlers); HandlerDescriptor* handler; // To ensure forward progress through the handlers, begin past the last // socket number that we handled: if (fLastHandledSocketNum >= 0) { while ((handler = iter.next()) != NULL) { if (handler->socketNum == fLastHandledSocketNum) break; } if (handler == NULL) { fLastHandledSocketNum = -1; iter.reset(); // start from the beginning instead } } while ((handler = iter.next()) != NULL) { if (FD_ISSET(handler->socketNum, &readSet) && FD_ISSET(handler->socketNum, &fReadSet) /* sanity check */ && handler->handlerProc != NULL) { fLastHandledSocketNum = handler->socketNum; // Note: we set "fLastHandledSocketNum" before calling the handler, // in case the handler calls "doEventLoop()" reentrantly. (*handler->handlerProc)(handler->clientData, SOCKET_READABLE); break; } } if (handler == NULL && fLastHandledSocketNum >= 0) { // We didn't call a handler, but we didn't get to check all of them, // so try again from the beginning: iter.reset(); while ((handler = iter.next()) != NULL) { if (FD_ISSET(handler->socketNum, &readSet) && FD_ISSET(handler->socketNum, &fReadSet) /* sanity check */ && handler->handlerProc != NULL) { fLastHandledSocketNum = handler->socketNum; // Note: we set "fLastHandledSocketNum" before calling the handler, // in case the handler calls "doEventLoop()" reentrantly. (*handler->handlerProc)(handler->clientData, SOCKET_READABLE); break; } } if (handler == NULL) fLastHandledSocketNum = -1;//because we didn't call a handler } } ////////////////////////////////////////////////////////////////////////// with regards rajesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060906/5edc1e23/attachment.html From zhangzx at rcs-9000.com Wed Sep 6 22:03:17 2006 From: zhangzx at rcs-9000.com (Zhixue Zhang) Date: Thu, 7 Sep 2006 13:03:17 +0800 Subject: [Live-devel] =?gb2312?b?tPC4tDogIHVuYWJsZSB0byBkbyBWaWRlbyBTdHJl?= =?gb2312?b?YW1pbmc=?= In-Reply-To: <006001c6d237$7bd83900$f801000a@imidomain.com> Message-ID: <000501c6d23a$f1350e00$347457c6@rcs9000.com> Please #define READ_FROM_FILES_SYNCHRONOUSLY in ByteStreamFileSource. and try again. Maybe, it can be solved. Zhixue Zhang Nari-relays Electric Corporation, Ltd. Tel:025-52100626 -----????----- ???: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns. live555.com] ?? Rajesh ????: 2006?9?7? 12:39 ???: LIVE555 Streaming Media - development & use ??: [Live-devel] unable to do Video Streaming HI I am unable to do video streaming thru "testOnDemandRTSPServer".I am using client as VLC.In VLC ,I am mentioning RTSP url mentioned in the "testOnDemandRTSPServer".when VLC is sending the Request for the RTSP url, the error is below mentioned and "testOnDemandRTSP server is not stopping and next time also when I am running the same error is coming. The Error is -> "mpeg4ESVideoTest" stream, from the file "test.m4v" Play this stream using the URL "rtsp://10.0.1.248:8554/mpeg4ESVideoTest" BasicTaskScheduler::SingleStep(): select() fails: No error ----- Original Message ----- From: shrinivas balulad To: live-devel at ns.live555.com Sent: Wednesday, September 06, 2006 10:45 AM Subject: [Live-devel] Is GSM codec supported in OpenRTSP? Hi, I am using OpenRTSP project to wirte video to a file in quick time format. This works fine for ulaw and alaw ( video is encoded using H263+). But for GSM codec, i am not able to here anything when i play the file. I am playing the video file using VLC media player. I am able to view the video but not able here any audio. I think GSM codec is supported in OpenRTSP. Please suggest me what is the problem. Thanks in advance Shrinivas _____ Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW _____ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060906/551ccb3e/attachment.html From rajeshkumar.r at imimobile.com Wed Sep 6 22:55:13 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 11:25:13 +0530 Subject: [Live-devel] #define READ_FROM_FILES_SYNCHRONOUSLY solved my problem References: <000501c6d23a$f1350e00$347457c6@rcs9000.com> Message-ID: <00c601c6d242$2f83f750$f801000a@imidomain.com> ??HI Thanks for helping me. I want to stream 3gp files.How to do this with regards rajesh ----- Original Message ----- From: Zhixue Zhang To: 'LIVE555 Streaming Media - development & use' Sent: Thursday, September 07, 2006 10:33 AM Subject: [Live-devel] ??: unable to do Video Streaming Please #define READ_FROM_FILES_SYNCHRONOUSLY in ByteStreamFileSource. and try again. Maybe, it can be solved. Zhixue Zhang Nari-relays Electric Corporation, Ltd. Tel:025-52100626 -----????----- ???: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] ?? Rajesh ????: 2006?9?7? 12:39 ???: LIVE555 Streaming Media - development & use ??: [Live-devel] unable to do Video Streaming HI I am unable to do video streaming thru "testOnDemandRTSPServer".I am using client as VLC.In VLC ,I am mentioning RTSP url mentioned in the "testOnDemandRTSPServer".when VLC is sending the Request for the RTSP url, the error is below mentioned and "testOnDemandRTSP server is not stopping and next time also when I am running the same error is coming. The Error is -> "mpeg4ESVideoTest" stream, from the file "test.m4v" Play this stream using the URL "rtsp://10.0.1.248:8554/mpeg4ESVideoTest" BasicTaskScheduler::SingleStep(): select() fails: No error ----- Original Message ----- From: shrinivas balulad To: live-devel at ns.live555.com Sent: Wednesday, September 06, 2006 10:45 AM Subject: [Live-devel] Is GSM codec supported in OpenRTSP? Hi, I am using OpenRTSP project to wirte video to a file in quick time format. This works fine for ulaw and alaw ( video is encoded using H263+). But for GSM codec, i am not able to here anything when i play the file. I am playing the video file using VLC media player. I am able to view the video but not able here any audio. I think GSM codec is supported in OpenRTSP. Please suggest me what is the problem. Thanks in advance Shrinivas -------------------------------------------------------------------------- Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060906/c3a74731/attachment-0001.html From rajeshkumar.r at imimobile.com Wed Sep 6 23:18:34 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 11:48:34 +0530 Subject: [Live-devel] getting parsing error while streaming mpeg4 codec Message-ID: <011501c6d245$7272f270$f801000a@imidomain.com> Hi All when I am starting testOnDemandRTSPServer I am getting below error . D:\Projects\VIDEO_PORTAL\VideoStreaming\live\testProgs>testOnDemandRTSPServer "mpeg4ESVideoTest" stream, from the file "test.m4v" Play this stream using the URL "rtsp://10.0.1.248:8554/mpeg4ESVideoTest" MPEGProgramStreamParser::parsePESPacket() error: PES_packet_length (26630) excee ds max frame size asked for (10000) please help me . with regards rajesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060906/42367e6f/attachment.html From rajeshkumar.r at imimobile.com Wed Sep 6 23:22:24 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 11:52:24 +0530 Subject: [Live-devel] Live media supports 3gp file format Message-ID: <012601c6d245$fbbaaf50$f801000a@imidomain.com> Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060906/af93ba4c/attachment.html From mandelfi at free.fr Thu Sep 7 00:34:38 2006 From: mandelfi at free.fr (Mandelfi) Date: Thu, 07 Sep 2006 09:34:38 +0200 Subject: [Live-devel] File size with OpenRTSP In-Reply-To: References: <44FEEDBF.1020407@free.fr> Message-ID: <44FFCB8E.2000002@free.fr> >> Hi all, >> >> I am using OpenRTSP like this on my Linux box: >> openRTSP -F my_file rtsp://my_url >> to capture a streaming video program. >> This works very good but the file cannot be bigger than 2Go. Why ? > > I don't know. The file output code (to implement the "FileSink" > class) uses the standard routines "fopen()" and "fwrite()" (and > doesn't try to seek at all within the file), so I don't know why the > output file would get restricted to 2**31 bytes in size, if your OS > supports larger sizes. What version of Linux are you using? I use a standard kernel on a debian Sarge (2.6.8). I made this test on the box, look : The file "size_test_100Mo" size is 98Mo ; at begin, "size_test" size is 0. # while /bin/true ; do cat size_test_100Mo >> size_test ; ls -lh size_test ; sleep 1 ; done -rw-r--r-- 1 root root 98M 2006-09-07 09:25 size_test -rw-r--r-- 1 root root 195M 2006-09-07 09:25 size_test -rw-r--r-- 1 root root 292M 2006-09-07 09:25 size_test -rw-r--r-- 1 root root 389M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 486M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 583M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 681M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 778M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 875M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 972M 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 1,1G 2006-09-07 09:26 size_test -rw-r--r-- 1 root root 1,2G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,3G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,4G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,5G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,6G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,7G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,8G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,9G 2006-09-07 09:27 size_test -rw-r--r-- 1 root root 1,9G 2006-09-07 09:28 size_test -rw-r--r-- 1 root root 2,0G 2006-09-07 09:28 size_test -rw-r--r-- 1 root root 2,1G 2006-09-07 09:28 size_test -rw-r--r-- 1 root root 2,2G 2006-09-07 09:28 size_test -rw-r--r-- 1 root root 2,3G 2006-09-07 09:28 size_test -rw-r--r-- 1 root root 2,4G 2006-09-07 09:28 size_test -rw-r--r-- 1 root root 2,5G 2006-09-07 09:28 size_test So there is no problem to generate files bigger than 2Go on the box. I am quite lost... From rajeshkumar.r at imimobile.com Thu Sep 7 00:46:12 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 13:16:12 +0530 Subject: [Live-devel] File size with OpenRTSP References: <44FEEDBF.1020407@free.fr> <44FFCB8E.2000002@free.fr> Message-ID: <03de01c6d251$b08814d0$f801000a@imidomain.com> which video format(codec) u are using. with regards rajesh ----- Original Message ----- From: "Mandelfi" To: "LIVE555 Streaming Media - development & use" Sent: Thursday, September 07, 2006 1:04 PM Subject: Re: [Live-devel] File size with OpenRTSP >>> Hi all, >>> >>> I am using OpenRTSP like this on my Linux box: >>> openRTSP -F my_file rtsp://my_url >>> to capture a streaming video program. >>> This works very good but the file cannot be bigger than 2Go. Why ? >> >> I don't know. The file output code (to implement the "FileSink" >> class) uses the standard routines "fopen()" and "fwrite()" (and >> doesn't try to seek at all within the file), so I don't know why the >> output file would get restricted to 2**31 bytes in size, if your OS >> supports larger sizes. What version of Linux are you using? > > I use a standard kernel on a debian Sarge (2.6.8). > I made this test on the box, look : The file "size_test_100Mo" size is > 98Mo ; at begin, "size_test" size is 0. > > # while /bin/true ; do cat size_test_100Mo >> size_test ; ls -lh > size_test ; sleep 1 ; done > -rw-r--r-- 1 root root 98M 2006-09-07 09:25 size_test > -rw-r--r-- 1 root root 195M 2006-09-07 09:25 size_test > -rw-r--r-- 1 root root 292M 2006-09-07 09:25 size_test > -rw-r--r-- 1 root root 389M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 486M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 583M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 681M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 778M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 875M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 972M 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 1,1G 2006-09-07 09:26 size_test > -rw-r--r-- 1 root root 1,2G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,3G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,4G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,5G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,6G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,7G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,8G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,9G 2006-09-07 09:27 size_test > -rw-r--r-- 1 root root 1,9G 2006-09-07 09:28 size_test > -rw-r--r-- 1 root root 2,0G 2006-09-07 09:28 size_test > -rw-r--r-- 1 root root 2,1G 2006-09-07 09:28 size_test > -rw-r--r-- 1 root root 2,2G 2006-09-07 09:28 size_test > -rw-r--r-- 1 root root 2,3G 2006-09-07 09:28 size_test > -rw-r--r-- 1 root root 2,4G 2006-09-07 09:28 size_test > -rw-r--r-- 1 root root 2,5G 2006-09-07 09:28 size_test > > So there is no problem to generate files bigger than 2Go on the box. > > I am quite lost... > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From shaswata at alumnux.com Thu Sep 7 01:19:21 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Thu, 7 Sep 2006 13:49:21 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com> Message-ID: <001601c6d256$52650390$2e0aa8c0@NITU> Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/ae1becf7/attachment.html From rajeshkumar.r at imimobile.com Thu Sep 7 01:36:02 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 14:06:02 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com> <001601c6d256$52650390$2e0aa8c0@NITU> Message-ID: <043b01c6d258$a6a16d70$f801000a@imidomain.com> Hi Shaswata when I tried with mpeg4 also it's not reading the file. i used testOnDemandRTSPServer and used mpeg4 file as test.m4v. The file is not opening. can u plz guide me for mpeg4 format. with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 1:49 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/43d5a9cc/attachment.html From mandelfi at free.fr Thu Sep 7 01:51:02 2006 From: mandelfi at free.fr (Mandelfi) Date: Thu, 07 Sep 2006 10:51:02 +0200 Subject: [Live-devel] File size with OpenRTSP In-Reply-To: <03de01c6d251$b08814d0$f801000a@imidomain.com> References: <44FEEDBF.1020407@free.fr> <44FFCB8E.2000002@free.fr> <03de01c6d251$b08814d0$f801000a@imidomain.com> Message-ID: <44FFDD76.8040802@free.fr> > which video format(codec) u are using. > > with regards > rajesh Hi Rajesh, I think that it is MPEG-2 TS. I am not sure because it is a broadcast from a french Internet/TV provider named "Free" (http://www.free.fr). I receive RTSP MPEG2 from it and I save broadcast to disk with OpenRTSP. If you want, I can send you a small file (few seconds of video < 1MB). Regards, Mandelfi From rajeshkumar.r at imimobile.com Thu Sep 7 02:08:16 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 14:38:16 +0530 Subject: [Live-devel] File size with OpenRTSP References: <44FEEDBF.1020407@free.fr> <44FFCB8E.2000002@free.fr><03de01c6d251$b08814d0$f801000a@imidomain.com> <44FFDD76.8040802@free.fr> Message-ID: <04c101c6d25d$27a20980$f801000a@imidomain.com> HI The same I want to do with mpeg4. How did u save it to the file(which command). Actually I want to save the same thing in file at every 10 second with regards rajesh ----- Original Message ----- From: "Mandelfi" To: "LIVE555 Streaming Media - development & use" Sent: Thursday, September 07, 2006 2:21 PM Subject: Re: [Live-devel] File size with OpenRTSP >> which video format(codec) u are using. >> >> with regards >> rajesh > > Hi Rajesh, > > I think that it is MPEG-2 TS. I am not sure because it is a broadcast > from a french Internet/TV provider named "Free" (http://www.free.fr). I > receive RTSP MPEG2 from it and I save broadcast to disk with OpenRTSP. > If you want, I can send you a small file (few seconds of video < 1MB). > > Regards, > > Mandelfi > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From shaswata at alumnux.com Thu Sep 7 02:23:34 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Thu, 7 Sep 2006 14:53:34 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com><001601c6d256$52650390$2e0aa8c0@NITU> <043b01c6d258$a6a16d70$f801000a@imidomain.com> Message-ID: <003501c6d25f$4a98ece0$2e0aa8c0@NITU> Hi Rajesh, Most probably it is the problem with the test.m4v file itself. The test.m4v file must conform to MPEG4 video elementary stream format. You can verify it easily by playing it through MPlayer. Even you can create a .m4v file by yourself by using 'mp4creator' tool of MPEG4IP project (just extract the video track from any .mp4 file). For further verification, just check with the etherial capture whether testOnDemandRTSPServer is sending the RTP packets. If it is not sending then you can be almost certain that 'test.m4v' is corrupt. Waiting for your test result... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:06 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata when I tried with mpeg4 also it's not reading the file. i used testOnDemandRTSPServer and used mpeg4 file as test.m4v. The file is not opening. can u plz guide me for mpeg4 format. with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 1:49 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/cc804646/attachment.html From finlayson at live555.com Thu Sep 7 02:40:11 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Sep 2006 02:40:11 -0700 Subject: [Live-devel] File size with OpenRTSP In-Reply-To: <44FFCB8E.2000002@free.fr> References: <44FEEDBF.1020407@free.fr> <44FFCB8E.2000002@free.fr> Message-ID: >So there is no problem to generate files bigger than 2Go on the box. > >I am quite lost... I am also puzzled by this. Which RTSP *server* are you using? (Is it ours - e.g., "testOnDemandRTSPServer"?) Are you sure that the server is *sending* more than 2^31 bytes of data? -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From rajeshkumar.r at imimobile.com Thu Sep 7 02:49:28 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 15:19:28 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com><001601c6d256$52650390$2e0aa8c0@NITU><043b01c6d258$a6a16d70$f801000a@imidomain.com> <003501c6d25f$4a98ece0$2e0aa8c0@NITU> Message-ID: <053f01c6d262$e9234ab0$f801000a@imidomain.com> Hi Shaswata using testOnDemandRTSpServer and sending req from VLc I am able to play the file. But my limitaions is 3gp format.I can play file only in 3gp format or at max mpg fomat. plz suggest in this way with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:53 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, Most probably it is the problem with the test.m4v file itself. The test.m4v file must conform to MPEG4 video elementary stream format. You can verify it easily by playing it through MPlayer. Even you can create a .m4v file by yourself by using 'mp4creator' tool of MPEG4IP project (just extract the video track from any .mp4 file). For further verification, just check with the etherial capture whether testOnDemandRTSPServer is sending the RTP packets. If it is not sending then you can be almost certain that 'test.m4v' is corrupt. Waiting for your test result... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:06 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata when I tried with mpeg4 also it's not reading the file. i used testOnDemandRTSPServer and used mpeg4 file as test.m4v. The file is not opening. can u plz guide me for mpeg4 format. with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 1:49 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh ------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/c859371d/attachment-0001.html From finlayson at live555.com Thu Sep 7 02:45:50 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Sep 2006 02:45:50 -0700 Subject: [Live-devel] RTP over TCP In-Reply-To: <200609071134.AA50266344@RCS-9000.COM> References: <200609071134.AA50266344@RCS-9000.COM> Message-ID: >I test "-t" option of openrtsp. >It seems that after the client has setup and played session, >"RTSPClientSession::incomingRequestHandler1()" never be called in >server. So "noteLiveness()" will not be called. Thanks for reporting this. This was a bug! I have now released a new version (2006.09.07) of the "LIVE555 Streaming Media" code that fixes this bug. Ross. From shaswata at alumnux.com Thu Sep 7 04:11:48 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Thu, 7 Sep 2006 16:41:48 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com><001601c6d256$52650390$2e0aa8c0@NITU><043b01c6d258$a6a16d70$f801000a@imidomain.com><003501c6d25f$4a98ece0$2e0aa8c0@NITU> <053f01c6d262$e9234ab0$f801000a@imidomain.com> Message-ID: <002701c6d26e$694ac0a0$2e0aa8c0@NITU> Hi Rajesh, I think live supports .mpg (both program and transport stream) file for streaming. But i'm pretty confused on looking your requirement: Mpeg Program Stream and transport Stream are completely different from .3gp format. It wil be easier to suggest if you mention little bit of the use case (including what kind of video codec is expected)... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 3:19 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata using testOnDemandRTSpServer and sending req from VLc I am able to play the file. But my limitaions is 3gp format.I can play file only in 3gp format or at max mpg fomat. plz suggest in this way with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:53 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, Most probably it is the problem with the test.m4v file itself. The test.m4v file must conform to MPEG4 video elementary stream format. You can verify it easily by playing it through MPlayer. Even you can create a .m4v file by yourself by using 'mp4creator' tool of MPEG4IP project (just extract the video track from any .mp4 file). For further verification, just check with the etherial capture whether testOnDemandRTSPServer is sending the RTP packets. If it is not sending then you can be almost certain that 'test.m4v' is corrupt. Waiting for your test result... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:06 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata when I tried with mpeg4 also it's not reading the file. i used testOnDemandRTSPServer and used mpeg4 file as test.m4v. The file is not opening. can u plz guide me for mpeg4 format. with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 1:49 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh ---------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/4faccd16/attachment.html From rajeshkumar.r at imimobile.com Thu Sep 7 04:21:57 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 7 Sep 2006 16:51:57 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com><001601c6d256$52650390$2e0aa8c0@NITU><043b01c6d258$a6a16d70$f801000a@imidomain.com><003501c6d25f$4a98ece0$2e0aa8c0@NITU><053f01c6d262$e9234ab0$f801000a@imidomain.com> <002701c6d26e$694ac0a0$2e0aa8c0@NITU> Message-ID: <05d601c6d26f$d4cceb40$f801000a@imidomain.com> Hi Shaswata 3gp file format (H263(Video) + AMR(Audio)).If I am receiving stream in this format then only I can play the file.can u tell me that live Media supports .mpg file format with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 4:41 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, I think live supports .mpg (both program and transport stream) file for streaming. But i'm pretty confused on looking your requirement: Mpeg Program Stream and transport Stream are completely different from .3gp format. It wil be easier to suggest if you mention little bit of the use case (including what kind of video codec is expected)... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 3:19 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata using testOnDemandRTSpServer and sending req from VLc I am able to play the file. But my limitaions is 3gp format.I can play file only in 3gp format or at max mpg fomat. plz suggest in this way with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:53 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, Most probably it is the problem with the test.m4v file itself. The test.m4v file must conform to MPEG4 video elementary stream format. You can verify it easily by playing it through MPlayer. Even you can create a .m4v file by yourself by using 'mp4creator' tool of MPEG4IP project (just extract the video track from any .mp4 file). For further verification, just check with the etherial capture whether testOnDemandRTSPServer is sending the RTP packets. If it is not sending then you can be almost certain that 'test.m4v' is corrupt. Waiting for your test result... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:06 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata when I tried with mpeg4 also it's not reading the file. i used testOnDemandRTSPServer and used mpeg4 file as test.m4v. The file is not opening. can u plz guide me for mpeg4 format. with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 1:49 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh -------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/926cfa4c/attachment-0001.html From shaswata at alumnux.com Thu Sep 7 06:36:56 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Thu, 7 Sep 2006 19:06:56 +0530 Subject: [Live-devel] Live media supports 3gp file format References: <012601c6d245$fbbaaf50$f801000a@imidomain.com><001601c6d256$52650390$2e0aa8c0@NITU><043b01c6d258$a6a16d70$f801000a@imidomain.com><003501c6d25f$4a98ece0$2e0aa8c0@NITU><053f01c6d262$e9234ab0$f801000a@imidomain.com><002701c6d26e$694ac0a0$2e0aa8c0@NITU> <05d601c6d26f$d4cceb40$f801000a@imidomain.com> Message-ID: <003601c6d282$afed8c40$2e0aa8c0@NITU> Hi Rajesh, I can think of two possibilities: 1. Offline (i.e. before starting the streaming of .3gp video file) extract the video(H263) and audio track(AMR) in two separate file. Now you have to stream this file simultaneously by creating two separate subsession. H263 Video : H263plusVideoFileServerMediaSubsession (H263plusVideoRTPSink + H263plusVideoStreamFramer) AMR audio : AMRAudioFileServerMediaSubsession ( AMRAudioRTPSink + AMRAudioFileSource) The tracks can be easily be extracted by using 'mp4creator' tool of MPEG4IP project. Audio and video synchronization will be taken care by the player (client) in most cases (if at all). 2. Develop a completely new class derived from a FramedSource which can read individual frame from a .3gp file directly (ISO Base media compliant). Now write a new FileServerMediaSubsession which calls those new FramedSource. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 4:51 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata 3gp file format (H263(Video) + AMR(Audio)).If I am receiving stream in this format then only I can play the file.can u tell me that live Media supports .mpg file format with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 4:41 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, I think live supports .mpg (both program and transport stream) file for streaming. But i'm pretty confused on looking your requirement: Mpeg Program Stream and transport Stream are completely different from .3gp format. It wil be easier to suggest if you mention little bit of the use case (including what kind of video codec is expected)... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 3:19 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata using testOnDemandRTSpServer and sending req from VLc I am able to play the file. But my limitaions is 3gp format.I can play file only in 3gp format or at max mpg fomat. plz suggest in this way with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:53 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, Most probably it is the problem with the test.m4v file itself. The test.m4v file must conform to MPEG4 video elementary stream format. You can verify it easily by playing it through MPlayer. Even you can create a .m4v file by yourself by using 'mp4creator' tool of MPEG4IP project (just extract the video track from any .mp4 file). For further verification, just check with the etherial capture whether testOnDemandRTSPServer is sending the RTP packets. If it is not sending then you can be almost certain that 'test.m4v' is corrupt. Waiting for your test result... With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 2:06 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Shaswata when I tried with mpeg4 also it's not reading the file. i used testOnDemandRTSPServer and used mpeg4 file as test.m4v. The file is not opening. can u plz guide me for mpeg4 format. with regards rajesh ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 1:49 PM Subject: Re: [Live-devel] Live media supports 3gp file format Hi Rajesh, To my best knowledge, Live Media does not support 3gp file format. A 3gp file format (most commonly with .3gp extension) can be of two type either hinted or without hint. In both of the case, the file format conforms to ISO Base Media file format. In live library, there is no FramedSource to read from a ISO BaseMedia file format. With regards, Shaswata ----- Original Message ----- From: Rajesh To: LIVE555 Streaming Media - development & use Sent: Thursday, September 07, 2006 11:52 AM Subject: [Live-devel] Live media supports 3gp file format Hi All Live Media supports 3gp file format.if it supports,how to use this. with regards rajesh ------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/8075529c/attachment.html From millallo at gmail.com Thu Sep 7 08:23:48 2006 From: millallo at gmail.com (Emiliano Parasassi) Date: Thu, 07 Sep 2006 17:23:48 +0200 Subject: [Live-devel] FW packets from RTSP Client Message-ID: <45003984.5090109@gmail.com> Hi Ross, As described in 6.1.2 of this draft of IETF: http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-03 [...] To allow UDP packets to arrive from the server to a client behind a restricted NAT, the client must send the very first UDP packet to pinch a hole in the NAT. The client, before sending a RTSP PLAY request, [...] I wrote this patch that allows RTSPClient to receive RTP/RTCP packets even if they are behind a restricted NAT. I've tested in this architecture; RTSPServer <--> Public Router <--> Internet <--> Public Router <--> RTSPClient ...and seems to work fine. I hope this help Emiliano Parasassi -------------- next part -------------- A non-text attachment was scrubbed... Name: live555_fw_packet.patch Type: text/x-patch Size: 887 bytes Desc: not available Url : http://lists.live555.com/pipermail/live-devel/attachments/20060907/b9d03e4f/attachment.bin From bidibulle at operamail.com Thu Sep 7 09:10:54 2006 From: bidibulle at operamail.com (David BERTRAND) Date: Thu, 07 Sep 2006 17:10:54 +0100 Subject: [Live-devel] exit(1) in AMRDeinterleavingBuffer Message-ID: <20060907161055.196E643D35@ws5-1.us4.outblaze.com> Hi Ross, When receiving buggy AMR packets, my app enter deliverIncomingFrame() and then exits with return code 1. The code in AMRDeinterleavingBuffer is the following : void AMRDeinterleavingBuffer ::deliverIncomingFrame(unsigned frameSize, RawAMRRTPSource* source, struct timeval presentationTime) { unsigned char const ILL = source->ILL(); unsigned char const ILP = source->ILP(); unsigned frameIndex = source->frameIndex(); unsigned short packetSeqNum = source->curPacketRTPSeqNum(); // First perform a sanity check on the parameters: // (This is overkill, as the source should have already done this.) if (ILP > ILL || frameIndex == 0) { #ifdef DEBUG fprintf(stderr, "AMRDeinterleavingBuffer::deliverIncomingFrame() param sanity check failed (%d,%d,%d,%d)\n", frameSize, ILL, ILP, frameIndex); #endif exit(1); } Actually, it exits because frameIndex=0. The origin of the problem is some invalid FT (frame type) fields in the AMR packets received, resulting in a mismatch between the expected frame sizes and the real frame sizes. At some time, this leads to (fHead-fTail) equalling zero in getNextEnclosedFrameParameters() call, which itself leads to dataSize=0 in nextEnclosedFrameSize() call, which therefore returns immediately without incrementing the frameIndex. How can avoid an exit call in AMRAudioRTPSource ? RFC 3267 indicates that any packet containing forbidden FT should be discarded. I guess it is what should be done. Here is my stderr output when running in DEBUG mode : RTCPInstance[0x827c500]::RTCPInstance() schedule(1.925937->1157559526.310532) RTCPInstance[0x827ded0]::RTCPInstance() schedule(2.093193->1157559526.478279) schedule(0.797310->1157559527.115028) schedule(0.849278->1157559527.336935) schedule(1.135812->1157559528.253455) schedule(2.708205->1157559530.045905) schedule(3.942898->1157559532.200478) schedule(1.475081->1157559531.522707) schedule(4.692819->1157559536.220446) schedule(2.080536->1157559534.288186) Unpacking 'bandwidth-efficient' payload (33 bytes): f9:3c:91:17:16:be:66:79:e1:e0:01:e7:af:f0:00:00:00:80:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00: Warning: Unpacking frame 4 of 4: want 39 bits, but only 0 are available! packetSize: 40, ILL: 0, ILP: 0 TOC entry: F 1, FT 2, Q 0 TOC entry: F 1, FT 14, Q 0 TOC entry: F 1, FT 2, Q 0 TOC entry: F 0, FT 8, Q 1 TOC contains 4 entries (3 non-empty) Total special header size: 5 AMRBufferedPacket::nextEnclosedFrameSize(): frame #: 0, FT: 2, isWideband: 0 => frameSize: 15 (dataSize: 35) AMRDeinterleavingBuffer::deliverIncomingFrame(): new interleave group AMRDeinterleavingBuffer::deliverIncomingFrame(): frameIndex 0 (0,0) put in bank 1, bin 0 (1): size 15, header 0x10, presentationTime 1157559532.532693 AMRBufferedPacket::nextEnclosedFrameSize(): frame #: 1, FT: 14, isWideband: 0 => frameSize: 0 (dataSize: 20) AMRDeinterleavingBuffer::deliverIncomingFrame(): new interleave group AMRDeinterleavingBuffer::deliverIncomingFrame(): frameIndex 1 (1,0) put in bank 0, bin 0 (1): size 0, header 0x70, presentationTime 1157559532.552693 AMRDeinterleavingBuffer::retrieveFrame(): from bank 1, bin 0: size 15, header 0x10, presentationTime 1157559532.532693 AMRBufferedPacket::nextEnclosedFrameSize(): frame #: 2, FT: 2, isWideband: 0 => frameSize: 15 (dataSize: 20) AMRDeinterleavingBuffer::deliverIncomingFrame(): new interleave group AMRDeinterleavingBuffer::deliverIncomingFrame(): frameIndex 2 (2,0) put in bank 1, bin 0 (1): size 15, header 0x10, presentationTime 1157559532.572693 AMRDeinterleavingBuffer::retrieveFrame(): from bank 0, bin 0: size 0, header 0x78, presentationTime 1157559532.552693 AMRBufferedPacket::nextEnclosedFrameSize(): frame #: 3, FT: 8, isWideband: 0 => frameSize: 5 (dataSize: 5) AMRDeinterleavingBuffer::deliverIncomingFrame(): new interleave group AMRDeinterleavingBuffer::deliverIncomingFrame(): frameIndex 3 (3,0) put in bank 0, bin 0 (1): size 5, header 0x44, presentationTime 1157559532.592693 AMRDeinterleavingBuffer::retrieveFrame(): from bank 1, bin 0: size 15, header 0x10, presentationTime 1157559532.572693 Unpacking 'bandwidth-efficient' payload (33 bytes): f7:3c:10:29:83:28:b4:51:c7:ec:8a:ee:00:e4:57:40:31:40:00:6e:dd:c0:1d:29:48:00:02:18:8b:f1:2b:28:20: Warning: 254 bits remain unused! packetSize: 2, ILL: 0, ILP: 0 TOC entry: F 0, FT 14, Q 0 TOC contains 1 entries (0 non-empty) Total special header size: 2 AMRDeinterleavingBuffer::deliverIncomingFrame() param sanity check failed (0,0,0,0) --->exit(1) Thanks, David -- _______________________________________________ Surf the Web in a faster, safer and easier way: Download Opera 9 at http://www.opera.com Powered by Outblaze From finlayson at live555.com Thu Sep 7 10:14:52 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Sep 2006 10:14:52 -0700 Subject: [Live-devel] exit(1) in AMRDeinterleavingBuffer In-Reply-To: <20060907161055.196E643D35@ws5-1.us4.outblaze.com> References: <20060907161055.196E643D35@ws5-1.us4.outblaze.com> Message-ID: >How can avoid an exit call in AMRAudioRTPSource ? RFC 3267 indicates >that any packet containing forbidden FT should be discarded. I guess >it is what should be done. Yes, I'll take another look at that code, and see if I can make it more robust when invalid AMR data. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From darnold at futurec.net Thu Sep 7 10:18:32 2006 From: darnold at futurec.net (David Arnold) Date: Thu, 7 Sep 2006 10:18:32 -0700 Subject: [Live-devel] Capturing H.264 content with openRTSP In-Reply-To: <20060907161055.196E643D35@ws5-1.us4.outblaze.com> Message-ID: We need to capture streaming H.264 content with openRTSP. In this case, we are using openRTSP as a recorder. The problem is that there is no way to distinguish NAL units in the saved file (video-H264-1) created by openRTSP. VLC cannot play this file. The file is basically unusable. The byte stream format prefix (0 0 0 1) as described in annex B of ISO 14496-10 has been removed by our H.264 DeviceSource and openRTSP simply captures raw NAL units and saves them. Do I need to write a H264FileSink class to reinsert the byte stream format prefix as described in annex B of ISO 14496-10 and use this class in playCommon.cpp or is our DeviceSource class doing something incorrectly? Ideally, we would like to capture the media in a QuickTime container file, but the QuickTimeFileSink does not currently support H.264 content. What would I need to do to QuickTimeFileSink so that QuickTime would be able to play the container file correctly? Your suggestions are greatly appreciated. Thank you, Dave Arnold Future Concepts, La Verne The information contained in this electronic mail transmission is intended only for the use of the individual or entity named above and is privileged and confidential. If you are not the intended recipient, please do not read, copy, use or disclose this communication to others. Any dissemination, distribution or copying of this communication other than to the person or entity named above is strictly prohibited. If you have received this communication in error, please immediately delete it from your system. From finlayson at live555.com Thu Sep 7 10:38:49 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Sep 2006 10:38:49 -0700 Subject: [Live-devel] FW packets from RTSP Client In-Reply-To: <45003984.5090109@gmail.com> References: <45003984.5090109@gmail.com> Message-ID: >As described in 6.1.2 of this draft of IETF: >http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-03 > >[...] > To allow UDP packets to arrive from the server to a client behind a > restricted NAT, the client must send the very first UDP packet to > pinch a hole in the NAT. The client, before sending a RTSP PLAY > request, >[...] > >I wrote this patch that allows RTSPClient to receive RTP/RTCP packets >even if they are behind a restricted NAT. I've tested in this architecture; > >RTSPServer <--> Public Router <--> Internet <--> Public Router <--> RTSPClient > >...and seems to work fine. Thanks for the note. At some point I hope to add more complete STUN/ICE functionality to the RTSP implementation (client and server). In the meantime, if people want to try using your patch, then they they may find it useful; thanks for making it available. A couple of things to note about your patch, however: - Your packet data is not valid RTP or RTCP, so has the potential to confuse the receiving server, or at least to cause it to log invalid incoming data (especially for RTCP). - You are sending your packet with TTL 0, so you're lucky it leaves the local machine, let alone traverses NATs. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Sep 7 10:45:46 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Sep 2006 10:45:46 -0700 Subject: [Live-devel] Capturing H.264 content with openRTSP In-Reply-To: References: Message-ID: >Do I need to write a H264FileSink class to reinsert the byte stream format >prefix as described in annex B of ISO 14496-10 and use this class in >playCommon.cpp Yes, that would work. (Note how we already do something similar for incoming AMR audio frames; search for "AMRAudioFileSink" in "playCommon.cpp". > or is our DeviceSource class doing something incorrectly? No, your "DeviceSource" is correct by not including the "0 0 0 1" prefix. That is not supposed to be part of the NAL unit data that gets sent in outgoing RTP packets. >Ideally, we would like to capture the media in a QuickTime container file, >but the QuickTimeFileSink does not currently support H.264 content. What >would I need to do to QuickTimeFileSink so that QuickTime would be able to >play the container file correctly? Update the implementation of "QuickTimeFileSink" to support H.264 content :-) (If you want to try doing this, look for "h263" and "mp4v" for examples of how it already supports H.263 and (regular) MPEG-4 video.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From millallo at gmail.com Thu Sep 7 12:02:57 2006 From: millallo at gmail.com (Emiliano Parasassi) Date: Thu, 7 Sep 2006 21:02:57 +0200 Subject: [Live-devel] FW packets from RTSP Client In-Reply-To: References: <45003984.5090109@gmail.com> Message-ID: <9585da7b0609071202u7932ad97p849b625585d79ec1@mail.gmail.com> 2006/9/7, Ross Finlayson : > > >As described in 6.1.2 of this draft of IETF: > >http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-03 > > > >[...] > > To allow UDP packets to arrive from the server to a client behind a > > restricted NAT, the client must send the very first UDP packet to > > pinch a hole in the NAT. The client, before sending a RTSP PLAY > > request, > >[...] > > > >I wrote this patch that allows RTSPClient to receive RTP/RTCP packets > >even if they are behind a restricted NAT. I've tested in this > architecture; > > > >RTSPServer <--> Public Router <--> Internet <--> Public Router <--> > RTSPClient > > > >...and seems to work fine. > > Thanks for the note. At some point I hope to add more complete > STUN/ICE functionality to the RTSP implementation (client and > server). In the meantime, if people want to try using your patch, > then they they may find it useful; thanks for making it available. > > A couple of things to note about your patch, however: > - Your packet data is not valid RTP or RTCP, so has the potential to > confuse the receiving server, or at least to cause it to log invalid > incoming data (especially for RTCP). - You are sending your packet with TTL 0, so you're lucky it leaves > the local machine, let alone traverses NATs. It's true, but in the architecture described above, the scope is only to open an hole in the first node (the FW), and not to arrive to RTSP Server. However increment TTL is right at all. I hope you'll implement STUN/ICE directives soon, before the ufficial release of the new version of Videolan, so it will be possible to use RTSPClient in more circumstances. Regards Emiliano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/2d32597b/attachment.html From paul_li20 at yahoo.com.cn Thu Sep 7 12:19:05 2006 From: paul_li20 at yahoo.com.cn (Li bz) Date: Fri, 8 Sep 2006 03:19:05 +0800 (CST) Subject: [Live-devel] time duration in H264 streaming Message-ID: <20060907191905.57975.qmail@web15106.mail.cnb.yahoo.com> Hi Ross, I am trying to use the H264VideoSink to do H.264 streaming. I know I have to analyze the H.264 video stream to get the NAL unit, then stream it according to RFC3984. After I get the NAL unit from the H.264 video file, I have one question about how to calculate the time duration between two frame of the video, which relates to the timestamps of RTP packets when I stream them. For those packets belonging to the first frame, I choose the random time for the RTP packets which belongs to the first frame, however, for those packets belonging to second frame or later frames, I can not get any information from the H.264 video elementary stream. I did the experientment by using the time duration according to frame rate(25 or 30), I can stream some H.264 video stream file, but the playing speed is obviously not right. I should get the accurate time duration from the video file, but I don't know where to find these information. Can you give me some idea how to get the time duration between consecutive access unit of H.264 video? Is there any place that I missed in understanding RFC3984? Thanks in advance. Paul ___________________________________________________________ Mp3???-??????? http://music.yahoo.com.cn/?source=mail_mailbox_footer From shawn.baker at librestream.com Thu Sep 7 12:43:18 2006 From: shawn.baker at librestream.com (Shawn Baker) Date: Thu, 7 Sep 2006 14:43:18 -0500 Subject: [Live-devel] secure RTP (SRTP) Message-ID: <8230E1CC35AF9F43839F3049E930169A3888F8@yang.LibreStream.local> Has anybody looked into adding support for SRTP (secure RTP)? If so, what approach were you going to take? We were thinking that we'd try to integrate the libsrtp code into live, but any advice you could give us would be appreciated. Just so you know, we are currently using the live code in a custom video camera that streams its video over a wireless connection to a desktop application. Everything is still being developed, but we do have prototype cameras and we are successfully streaming. We would now like to secure that stream using SRTP. --- Shawn Baker Senior Software Developer LibreStream Technologies Inc. 204-487-0612 Ext. 211 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060907/10bbdace/attachment.html From finlayson at live555.com Thu Sep 7 13:58:50 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Sep 2006 13:58:50 -0700 Subject: [Live-devel] FW packets from RTSP Client In-Reply-To: <9585da7b0609071202u7932ad97p849b625585d79ec1@mail.gmail.com> References: <45003984.5090109@gmail.com> <9585da7b0609071202u7932ad97p849b625585d79ec1@mail.gmail.com> Message-ID: >I hope you'll implement STUN/ICE directives soon, before the >ufficial release of the new >version of Videolan, so it will be possible to use RTSPClient in >more circumstances. Note that VLC is just one of the many applications that uses the "LIVE555 Streaming Media" code. Also, there is not one 'official release' of VLC; instead, new versions get released often. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From chenw at blrcsv.china.bell-labs.com Thu Sep 7 20:03:13 2006 From: chenw at blrcsv.china.bell-labs.com (chenwei) Date: Fri, 8 Sep 2006 11:03:13 +0800 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs Message-ID: <20060908024945.299148722@blrcsv.china.bell-labs.com> live-devel???? I have changed the variable Boolean reuseFirstSource = True. the first opened player can play the stream quite well, but after I opened a player on another machine to play the same video(mpeg1 es video), at start it can play and then a segmentation fault occurs. Finally the server was killed. Is it a bug? ????????? ?? ????????chenwei ????????chenw at blrcsv.china.bell-labs.com ??????????2006-09-08 From finlayson at live555.com Fri Sep 8 00:08:13 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Sep 2006 00:08:13 -0700 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs In-Reply-To: <20060908024945.299148722@blrcsv.china.bell-labs.com> References: <20060908024945.299148722@blrcsv.china.bell-labs.com> Message-ID: > I have changed the variable Boolean reuseFirstSource = True. >the first opened player > >can play the stream quite well, but after I opened a player on >another machine to play the > >same video(mpeg1 es video), at start it can play and then a >segmentation fault occurs. > >Finally the server was killed. > > Is it a bug? Possibly, but unfortunately I have not been able to reproduce this problem myself. Does the problem happen only when you set "reuseFirstSource" to True, or does it also occur when "reuseFirstSource" is False? Also, does the problem happen with several different MPEG-1 ES video files (named "testv.mpg"), or only with one such file? -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From ymreddy at ssdi.sharp.co.in Fri Sep 8 01:30:48 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Fri, 8 Sep 2006 14:00:48 +0530 Subject: [Live-devel] temporal reference and Picture types for TS streams Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CF79@ssdimailsrvnt01.ssdi.sharp.co.in> Hello All, We are trying to capture the temporal reference and picture types for TS streams from MPEG Video-specific header after parsing the RTP fixed headers on the client side. We could able to do it for ES and PS streams in networkReadHandler() function. But the same functionality is not working for TS streams. Also we observed that on the server side for TS streams, MPEG Video-specific header is not attached to each RTP packet after the RTP fixed header. We are using testMPEG2TransportStreamer.cpp for streaming TS files. Any clues on this. Thanks and Regards Y. Mallikharjuna Reddy From finlayson at live555.com Fri Sep 8 02:01:35 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Sep 2006 02:01:35 -0700 Subject: [Live-devel] temporal reference and Picture types for TS streams In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520337CF79@ssdimailsrvnt01.ssdi.sharp.co.in > References: <7FB4685EA93D014C8E30AA087B66E7520337CF79@ssdimailsrvnt01.ssdi.sharp.co.in > Message-ID: >We are trying to capture the temporal reference and picture types for TS >streams from MPEG Video-specific header after parsing the RTP fixed headers >on the client side. We could able to do it for ES and PS streams in >networkReadHandler() function. But the same functionality is not working for >TS streams. Also we observed that on the server side for TS streams, MPEG >Video-specific header is not attached to each RTP packet after the RTP fixed >header That's correct. See RFC 2550, section 2. Unlike the RTP payload format for MPEG-1 or 2 Elementary Streams, the RTP payload format for MPEG Transport Streams does not include any extra RTP header; instead, Transport Stream data is packed, 'as is', into RTP packets. Therefore you will need to parse the Transport Stream data in order to get MPEG-specific information. Sorry. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From shaswata at alumnux.com Fri Sep 8 02:33:03 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Fri, 8 Sep 2006 15:03:03 +0530 Subject: [Live-devel] temporal reference and Picture types for TS streams References: <7FB4685EA93D014C8E30AA087B66E7520337CF79@ssdimailsrvnt01.ssdi.sharp.co.in> Message-ID: <000f01c6d329$c84eb8c0$2e0aa8c0@NITU> Hi Mallikharjuna, According to RFC-2250, encapsulation (i.e. adding MPEG1/2 specific fixed headers) is done only on elementary streams. Even if MPEG -PS is used, live first extracts the ES from it and then uses the encapsulation rule over it. In fact according to the RFC, if a server would have directly streamed the MPEG-PS, there is no need of any encapsulation (Ref: Section 2 of the RFC). Similarly for MPEG-TS, the RFC states only that each RTP packet must contain integral number of 188 bytes of MTS packets (Ref: Section 2 of the RFC) and there is no need of MPEG specific headers until and unless the RTP packets contains TS packets as it is. So if the server is intelligent enough to extract the elementary stream from a TS stream and then stream out following the encapsulating rule of RFC-2250, then only in the client side those information can be found. Another solution can be - let the server stream the TS packets, but the client parses those packets to find the information of temporal reference and picture type (because these informations can be found from the MPEG-bit streams too). Also, can we not use the virtual-function called processSpecialHeader() (which can be accessed in classes derived from RTPSource) to get the each RTP packet in the client side - in that case, will it not be cleaner design? With regards, Shaswata ----- Original Message ----- From: "Mallikharjuna Reddy (NAVT)" To: "'LIVE555 Streaming Media - development & use'" Sent: Friday, September 08, 2006 2:00 PM Subject: [Live-devel] temporal reference and Picture types for TS streams > > Hello All, > > We are trying to capture the temporal reference and picture types for TS > streams from MPEG Video-specific header after parsing the RTP fixed headers > on the client side. We could able to do it for ES and PS streams in > networkReadHandler() function. But the same functionality is not working for > TS streams. Also we observed that on the server side for TS streams, MPEG > Video-specific header is not attached to each RTP packet after the RTP fixed > header. We are using testMPEG2TransportStreamer.cpp for streaming TS files. > > Any clues on this. > > Thanks and Regards > Y. Mallikharjuna Reddy > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From rajeshkumar.r at imimobile.com Fri Sep 8 02:35:05 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Fri, 8 Sep 2006 15:05:05 +0530 Subject: [Live-devel] temporal reference and Picture types for TS streams References: <7FB4685EA93D014C8E30AA087B66E7520337CF79@ssdimailsrvnt01.ssdi.sharp.co.in> <000f01c6d329$c84eb8c0$2e0aa8c0@NITU> Message-ID: <074901c6d32a$10d7c0a0$f801000a@imidomain.com> Hi All what is the functionality of testMPEG2TransportStreamer.exe. with regards rajesh ----- Original Message ----- From: "Shaswata Jash" To: "LIVE555 Streaming Media - development & use" Sent: Friday, September 08, 2006 3:03 PM Subject: Re: [Live-devel] temporal reference and Picture types for TS streams > Hi Mallikharjuna, > > According to RFC-2250, encapsulation (i.e. adding MPEG1/2 specific fixed > headers) is done only on elementary streams. Even if MPEG -PS is used, > live > first extracts the ES from it and then uses the encapsulation rule over > it. > In fact according to the RFC, if a server would have directly streamed the > MPEG-PS, there is no need of any encapsulation (Ref: Section 2 of the > RFC). > Similarly for MPEG-TS, the RFC states only that each RTP packet must > contain > integral number of 188 bytes of MTS packets (Ref: Section 2 of the RFC) > and > there is no need of MPEG specific headers until and unless the RTP packets > contains TS packets as it is. > So if the server is intelligent enough to extract the elementary stream > from > a TS stream and then stream out following the encapsulating rule of > RFC-2250, then only in the client side those information can be found. > Another solution can be - let the server stream the TS packets, but the > client parses those packets to find the information of temporal reference > and picture type (because these informations can be found from the > MPEG-bit > streams too). > Also, can we not use the virtual-function called processSpecialHeader() > (which can be accessed in classes derived from RTPSource) to get the each > RTP packet in the client side - in that case, will it not be cleaner > design? > > With regards, > Shaswata > > ----- Original Message ----- > From: "Mallikharjuna Reddy (NAVT)" > To: "'LIVE555 Streaming Media - development & use'" > > Sent: Friday, September 08, 2006 2:00 PM > Subject: [Live-devel] temporal reference and Picture types for TS streams > > >> >> Hello All, >> >> We are trying to capture the temporal reference and picture types for TS >> streams from MPEG Video-specific header after parsing the RTP fixed > headers >> on the client side. We could able to do it for ES and PS streams in >> networkReadHandler() function. But the same functionality is not working > for >> TS streams. Also we observed that on the server side for TS streams, MPEG >> Video-specific header is not attached to each RTP packet after the RTP > fixed >> header. We are using testMPEG2TransportStreamer.cpp for streaming TS > files. >> >> Any clues on this. >> >> Thanks and Regards >> Y. Mallikharjuna Reddy >> _______________________________________________ >> live-devel mailing list >> live-devel at lists.live555.com >> http://lists.live555.com/mailman/listinfo/live-devel >> > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From ymreddy at ssdi.sharp.co.in Fri Sep 8 03:07:19 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Fri, 8 Sep 2006 15:37:19 +0530 Subject: [Live-devel] temporal reference and Picture types for TS streams Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CF7A@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Shaswata, Thanks for the information. The solution is that the client can parse the packets to get these values. We can do this in the processSpecialHeader() function. Thanks and Regards Y. Mallikharjuna Reddy -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com]On Behalf Of Shaswata Jash Sent: Friday, September 08, 2006 3:03 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] temporal reference and Picture types for TS streams Hi Mallikharjuna, According to RFC-2250, encapsulation (i.e. adding MPEG1/2 specific fixed headers) is done only on elementary streams. Even if MPEG -PS is used, live first extracts the ES from it and then uses the encapsulation rule over it. In fact according to the RFC, if a server would have directly streamed the MPEG-PS, there is no need of any encapsulation (Ref: Section 2 of the RFC). Similarly for MPEG-TS, the RFC states only that each RTP packet must contain integral number of 188 bytes of MTS packets (Ref: Section 2 of the RFC) and there is no need of MPEG specific headers until and unless the RTP packets contains TS packets as it is. So if the server is intelligent enough to extract the elementary stream from a TS stream and then stream out following the encapsulating rule of RFC-2250, then only in the client side those information can be found. Another solution can be - let the server stream the TS packets, but the client parses those packets to find the information of temporal reference and picture type (because these informations can be found from the MPEG-bit streams too). Also, can we not use the virtual-function called processSpecialHeader() (which can be accessed in classes derived from RTPSource) to get the each RTP packet in the client side - in that case, will it not be cleaner design? With regards, Shaswata ----- Original Message ----- From: "Mallikharjuna Reddy (NAVT)" To: "'LIVE555 Streaming Media - development & use'" Sent: Friday, September 08, 2006 2:00 PM Subject: [Live-devel] temporal reference and Picture types for TS streams > > Hello All, > > We are trying to capture the temporal reference and picture types for TS > streams from MPEG Video-specific header after parsing the RTP fixed headers > on the client side. We could able to do it for ES and PS streams in > networkReadHandler() function. But the same functionality is not working for > TS streams. Also we observed that on the server side for TS streams, MPEG > Video-specific header is not attached to each RTP packet after the RTP fixed > header. We are using testMPEG2TransportStreamer.cpp for streaming TS files. > > Any clues on this. > > Thanks and Regards > Y. Mallikharjuna Reddy > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel From jeremy at electrosilk.net Fri Sep 8 03:24:30 2006 From: jeremy at electrosilk.net (Jeremy) Date: Fri, 08 Sep 2006 18:24:30 +0800 Subject: [Live-devel] PES Packet length problem Message-ID: <450144DE.2080808@electrosilk.net> Hello In function InputESSourceRecord::deliverBufferToClient() there is the following code unsigned PES_packet_length = fInputBufferBytesAvailable - 6; fInputBuffer[4] = PES_packet_length>>8; fInputBuffer[5] = PES_packet_length; In the case that PES_packet_length exceeds 65536 the packet length will be recorded incorrectly. It is quite possible within the existing framework to exceed a value of 65536 (as I found out). What is the correct fix for this situation? Should an error be issued? Or should the data be split into smaller components? If the latter, what presentation time stamp should be used. Thanks Jeremy From jmbaio at gmail.com Fri Sep 8 04:58:43 2006 From: jmbaio at gmail.com (Juan Manuel Lopez Baio) Date: Fri, 8 Sep 2006 08:58:43 -0300 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs In-Reply-To: References: <20060908024945.299148722@blrcsv.china.bell-labs.com> Message-ID: On 9/8/06, Ross Finlayson wrote: > > I have changed the variable Boolean reuseFirstSource = True. > >the first opened player > > > >can play the stream quite well, but after I opened a player on > >another machine to play the > > > >same video(mpeg1 es video), at start it can play and then a > >segmentation fault occurs. > > > >Finally the server was killed. > > > > Is it a bug? > > Possibly, but unfortunately I have not been able to reproduce this > problem myself. > > Does the problem happen only when you set "reuseFirstSource" to True, > or does it also occur when "reuseFirstSource" is False? Also, does > the problem happen with several different MPEG-1 ES video files > (named "testv.mpg"), or only with one such file? > -- I had the same problem. I was testing two .mpg videos, and with one of them, if reuseFirstSource was true, it happened than when I connected to the stream with VLC, and then I connected from another VLC, the video "synchronized" (it didn't start from the begining) with the one already playing, and when they both got to the end, the server exploded with a seg fault. The problem went away when I set reuseFirstSource to false, and the video now also plays independantly. From finlayson at live555.com Fri Sep 8 05:44:26 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Sep 2006 05:44:26 -0700 Subject: [Live-devel] temporal reference and Picture types for TS streams In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520337CF7A@ssdimailsrvnt01.ssdi.sharp.co.in > References: <7FB4685EA93D014C8E30AA087B66E7520337CF7A@ssdimailsrvnt01.ssdi.sharp.co.in > Message-ID: >he solution is that the client can parse the packets to get these values. >We can do this in the processSpecialHeader() function. No, you should process the received Transport Stream data in the 'sink' object that receives it. "processSpecialHeader" is used - by "RTPSource" subclasses - to process any special RTP-payload-format-specific header that's present in the RTP packet. But, for MPEG Transport Streams, there's *no* special header. I.e., you shouldn't modify any of the RTP-specific code. Instead, write a new sink object (or a filter) to process the received Transport Stream data. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Fri Sep 8 05:52:33 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Sep 2006 05:52:33 -0700 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs In-Reply-To: References: <20060908024945.299148722@blrcsv.china.bell-labs.com> Message-ID: >I had the same problem. I was testing two .mpg videos, and with one of >them, if reuseFirstSource was true, it happened than when I connected >to the stream with VLC, and then I connected from another VLC, the >video "synchronized" (it didn't start from the begining) with the one >already playing Yes - that is exactly what setting "reuseFirstSource" to True is supposed to do. ("reuseFirstSource" means that the input source is opened and read just once, regardless of how many clients are currently connected to play the stream. That's why you are supposed to set "reuseFirstSource" to True when you're streaming from a live source.) >, and when they both got to the end, the server >exploded with a seg fault. However, *that* is not supposed to happen :-) Thanks for the report. I'll take another look at this, to see if I can reproduce the problem. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Fri Sep 8 06:09:31 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Sep 2006 06:09:31 -0700 Subject: [Live-devel] PES Packet length problem In-Reply-To: <450144DE.2080808@electrosilk.net> References: <450144DE.2080808@electrosilk.net> Message-ID: >Hello > >In function InputESSourceRecord::deliverBufferToClient() there is the >following code > >unsigned PES_packet_length = fInputBufferBytesAvailable - 6; >fInputBuffer[4] = PES_packet_length>>8; >fInputBuffer[5] = PES_packet_length; > >In the case that PES_packet_length exceeds 65536 the packet length will >be recorded incorrectly. >It is quite possible within the existing framework to exceed a value of >65536 Not if your input data is correct. According to the MPEG specifications, the maximum size of a PES packet is 6+65535. If the input to your "MPEG2TransportStreamFromPESSource" object delivers data in chunks greater than this, then you'll need to fix this. (However, I agree that the current code should print out a warning message if "PES_packet_length" gets to big.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Fri Sep 8 12:47:53 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Sep 2006 12:47:53 -0700 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs Message-ID: I have now reproduced this problem, and it seems to be related to the recent change to "ByteStreamFileSource" to use asynchronous file reading. (The problem does not, in fact, seem to be related to the setting of "reuseFirstSource".) Until this bug is fixed, you should be able to overcome it by adding the line #define READ_FROM_FILES_SYNCHRONOUSLY 1 To the start of "liveMedia/ByteStreamFileSource.cpp" -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From jeremy at electrosilk.net Fri Sep 8 17:55:33 2006 From: jeremy at electrosilk.net (Jeremy) Date: Sat, 09 Sep 2006 08:55:33 +0800 Subject: [Live-devel] PES Packet length problem In-Reply-To: References: <450144DE.2080808@electrosilk.net> Message-ID: <45021105.7050403@electrosilk.net> Ross Finlayson wrote: > According to the MPEG specifications, the maximum size of a PES > packet is 6+65535. If the input to your > "MPEG2TransportStreamFromPESSource" object delivers data in chunks > greater than this, then you'll need to fix this. > > (However, I agree that the current code should print out a warning > message if "PES_packet_length" gets to big.) PES packets can be longer than 6+65535. Quoting from ISO 13818-1, 2.4.3.7 PES_packet_length ? A 16-bit field specifying the number of bytes in the PES packet following the last byte of the field. A value of 0 indicates that the PES packet length is neither specified nor bounded and is allowed only in PES packets whose payload consists of bytes from a video elementary stream contained in Transport Stream packets. From jmbaio at gmail.com Fri Sep 8 21:02:18 2006 From: jmbaio at gmail.com (Juan Manuel Lopez Baio) Date: Sat, 9 Sep 2006 01:02:18 -0300 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs In-Reply-To: References: Message-ID: On 9/8/06, Ross Finlayson wrote: > I have now reproduced this problem, and it seems to be related to the > recent change to "ByteStreamFileSource" to use asynchronous file > reading. (The problem does not, in fact, seem to be related to the > setting of "reuseFirstSource".) Nonetheless, as I described before, the problem ocurred when "reuseFirstSource" was true, and stopped as soon as I set it to false (which, as you said before, also changed the behaviour, more to my liking). I don't know how recent the ByteStreamFileSource change is, but I should add that I'm using the library as downloaded almost three weeks ago. I could check on Monday the exact version, if necessary. From finlayson at live555.com Sat Sep 9 03:18:22 2006 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 9 Sep 2006 03:18:22 -0700 Subject: [Live-devel] PES Packet length problem In-Reply-To: <45021105.7050403@electrosilk.net> References: <450144DE.2080808@electrosilk.net> <45021105.7050403@electrosilk.net> Message-ID: >PES packets can be longer than 6+65535. Quoting from ISO 13818-1, 2.4.3.7 > >PES_packet_length - A 16-bit field specifying the number of bytes in >the PES packet following the last byte of the >field. A value of 0 indicates that the PES packet length is neither >specified nor bounded and is allowed only in >PES packets whose payload consists of bytes from a video elementary >stream contained in Transport Stream packets. OK, I stand corrected - thanks. I have now updated the "MPEG2TransportStreamFromPESSource" implementation so that the "PES_packet_length" field is set to 0 if the input data is too large to fit in 16 bits. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Sat Sep 9 03:22:06 2006 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 9 Sep 2006 03:22:06 -0700 Subject: [Live-devel] When using testOnDemandRTSPServer to stream the same stream concurrently, a segmentation fault occurs In-Reply-To: References: Message-ID: >I don't know how recent the ByteStreamFileSource change is, but I >should add that I'm using the library as downloaded almost three weeks >ago. In general, no support is given for anything other than the latest version of the software. If you're still seeing a reproduceable problem with the latest version of the software, please let us know. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From vkozhuhov at gmx.net Sun Sep 10 18:37:37 2006 From: vkozhuhov at gmx.net (Victor Kozhuhov) Date: Mon, 11 Sep 2006 04:37:37 +0300 Subject: [Live-devel] RTCP endless loop Message-ID: <004901c6d543$039bdff0$0100a8c0@sleepy> Hello, I found that RTCP interval calculation returns incorrect values, when average RTCP bandwidth is zero. It is not very common case, but sometimes it happens, and OnExpire(...) function receives rtcp_bw parameter equal to zero ( 0.0(0) ). Actually, less RTCP bandwidth result in bigger timeout, but not when bandwidth is 0. rtcp_from_spec.c file, rtcp_interval(...) function contains the following lines: t = avg_rtcp_size * n / rtcp_bw; if (t < rtcp_min_time) t = rtcp_min_time; After first line t is equal to 1.#INF000000000000, and second line fails to fix it - condition (t < rtcp_min_time) --> (1.#INF000000000000 < 5.0000000000000000) is false, so t remains equal to 1.#INF000000000000, that converted to microseconds (integer value) stands for 0. Is it a bug? Scheduler falls into endless loop performing background RTCP read with zero timeout... With best regards, Victor. From finlayson at live555.com Mon Sep 11 00:50:41 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 11 Sep 2006 00:50:41 -0700 Subject: [Live-devel] RTCP endless loop In-Reply-To: <004901c6d543$039bdff0$0100a8c0@sleepy> References: <004901c6d543$039bdff0$0100a8c0@sleepy> Message-ID: >I found that RTCP interval calculation returns incorrect values, when >average RTCP bandwidth is zero. >It is not very common case, but sometimes it happens It really shouldn't. Passing a value of zero as the "totSessionBW" parameter to "RTCPInstance::RTCPInstance()" should not be allowed. In the next released version of the code, if this happens, an error message will be printed, and a value of 1 (kbps) will be used instead. Thanks for noting this problem. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From millallo at gmail.com Mon Sep 11 07:12:11 2006 From: millallo at gmail.com (Emiliano Parasassi) Date: Mon, 11 Sep 2006 16:12:11 +0200 Subject: [Live-devel] ~H264VideoRTPSink bug Message-ID: <45056EBB.6000407@gmail.com> Hi Ross, I'm playing with H264 and seems to be a SEGV during the deletion of H264VideoRTPSink during the RTSP DESCRIBE handler. The ~H264VideoRTPSink destrucor misses the fSource = NULL. This fSource was replaced by H264FUAFragmenter at H264VideoRTPSink.cpp:76. So in the frame #1 the stopGettingFrames() function is called to an already freed object. #0 0x654d3131 in ?? () #1 0x0835d522 in FramedSource::stopGettingFrames (this=0x868ab58) at FramedSource.cpp:134 #2 0x08377ea6 in MediaSink::stopPlaying (this=0x868a900) at MediaSink.cpp:83 #3 0x08378026 in ~MediaSink (this=0x868a900) at MediaSink.cpp:32 #4 0x083867de in ~RTPSink (this=0x868a900) at RTPSink.cpp:72 #5 0x08387c86 in ~MultiFramedRTPSink (this=0x868a900) at MultiFramedRTPSink.cpp:53 #6 0x08388da4 in ~VideoRTPSink (this=0x868a900) at VideoRTPSink.cpp:32 #7 0x0837c497 in ~H264VideoRTPSink (this=0x868a900) at H264VideoRTPSink.cpp:54 #8 0x0835cac6 in MediaLookupTable::remove (this=0x8573588, name=0x868a908 "liveMedia14") at Media.cpp:175 #9 0x0835cc7a in Medium::close (env=@0x8573de8, name=0x868a908 "liveMedia14") at Media.cpp:77 #10 0x0835ccb5 in Medium::close (medium=0x868a900) at Media.cpp:83 #11 0x083a160b in OnDemandServerMediaSubsession::sdpLines (this=0x861d0b0) at OnDemandServerMediaSubsession.cpp:92 #12 0x0839f89c in ServerMediaSession::generateSDPDescription (this=0x861cff0) at ServerMediaSession.cpp:219 Regards Emiliano Parasassi From finlayson at live555.com Mon Sep 11 07:54:59 2006 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 11 Sep 2006 07:54:59 -0700 Subject: [Live-devel] ~H264VideoRTPSink bug In-Reply-To: <45056EBB.6000407@gmail.com> References: <45056EBB.6000407@gmail.com> Message-ID: Thanks. This will be fixed in the next release of the software. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From rajeshkumar.r at imimobile.com Mon Sep 11 22:01:11 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Tue, 12 Sep 2006 10:31:11 +0530 Subject: [Live-devel] ~H264VideoRTPSink bug References: <45056EBB.6000407@gmail.com> Message-ID: <001501c6d628$77ade9c0$f801000a@imidomain.com> Hi Ross is there any way through that server will stream video in H263 and Audio in AMR .The client will receive the same format and will write into the file and then we can read the file and stream it to the 3g mobile with regards rajesh ----- Original Message ----- From: "Ross Finlayson" To: "LIVE555 Streaming Media - development & use" Sent: Monday, September 11, 2006 8:24 PM Subject: Re: [Live-devel] ~H264VideoRTPSink bug > Thanks. This will be fixed in the next release of the software. > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From shaswata at alumnux.com Tue Sep 12 04:44:44 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Tue, 12 Sep 2006 17:14:44 +0530 Subject: [Live-devel] ~H264VideoRTPSink bug References: <45056EBB.6000407@gmail.com> <001501c6d628$77ade9c0$f801000a@imidomain.com> Message-ID: <000901c6d660$d72d9110$2e0aa8c0@NITU> Hi Rajesh, I am not able to understand your requirement. If you want to stream to a 3G Mobile, then you are supposed to use the existing streaming client which follows RTP. In that case, the client will obviously expect the RTP payload according to RFC specified packetization rule. So the server is not supposed to stream H263 video and AMR audio as it is. But if you are developping a customised client application in mobile, then obviously you can make your streaming server as required by your client. But i think that should not be your requirement. Any way, if you further clarify your requirements then it will be easier for the Live community to answer your query. With regards, Shaswata ----- Original Message ----- From: "Rajesh" To: "LIVE555 Streaming Media - development & use" Sent: Tuesday, September 12, 2006 10:31 AM Subject: Re: [Live-devel] ~H264VideoRTPSink bug > Hi Ross > is there any way through that server will stream video in H263 and Audio in > AMR .The client will receive the same format and will write into the file > and > then we can read the file and stream it to the 3g mobile > with regards > rajesh > > > > ----- Original Message ----- > From: "Ross Finlayson" > To: "LIVE555 Streaming Media - development & use" > > Sent: Monday, September 11, 2006 8:24 PM > Subject: Re: [Live-devel] ~H264VideoRTPSink bug > > > > Thanks. This will be fixed in the next release of the software. > > -- > > > > Ross Finlayson > > Live Networks, Inc. > > http://www.live555.com/ > > _______________________________________________ > > live-devel mailing list > > live-devel at lists.live555.com > > http://lists.live555.com/mailman/listinfo/live-devel > > > > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From nitin.e at gmail.com Tue Sep 12 05:27:06 2006 From: nitin.e at gmail.com (nitin jain) Date: Tue, 12 Sep 2006 17:57:06 +0530 Subject: [Live-devel] sending RTCP Report continously Message-ID: Hello everyone I am trying to send RTCP RR Report continously from testMPEG1or2VideoReceiver demo application. I am doing this OnExpire() function in the file rtcp_from_spec.c file. I added SendRTCPReport(e) statement 15 times in the loop "if (tn <= tc) {". I observed that the SSRC is getting removed the hash table in the function RTCPMemberDatabase::reapOldMembers() and a new entry is for the SSRC is created. Any clues why its happening. In this function, RTCPMemberDatabase::reapOldMembers(), the timecount value is updated with the return value from the function iter->next(key). Any clues where this value is being updated in the library. Regards Nitin Jain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060912/d7790e2b/attachment-0001.html From clem.taylor at gmail.com Tue Sep 12 19:41:37 2006 From: clem.taylor at gmail.com (Clem Taylor) Date: Tue, 12 Sep 2006 22:41:37 -0400 Subject: [Live-devel] '/' in streamName for rtsp server url? Message-ID: I'm trying to create a stream with a '/' in the url with live555 rtsp server. I'm trying to emulate the rtsp url used in an Axis camera to convince a third party software package to archive the video from the live555 code. Axis uses the url 'rtsp://host/mpeg4/1/media.amp'. I can connect to this URL on an Axis camera with vlc without problems. But when I try to do the same with my live555 based rtsp server, vlc gets a 'RTSP/1.0 404 Stream Not Found' from the DESCRIBE. I'm calling ServerMediaSession::createNew() with "mpeg4/1/media.amp" for the streamName. If I change the stream to "mpeg4=1=media.amp" then everything is happy. A '/' seems to be a valid character rtsp url according to rfc2326 and rfc2396. Is there some issue with having a '/' in the streamName? Do I need to so something special (like create ServerMediaSessions each of the sub-paths) to get this to work? I was using 2005.06.16 and updated to 2006.09.12 with similar results. Thanks, Clem From alexr at vigilanttechnology.com Wed Sep 13 01:36:51 2006 From: alexr at vigilanttechnology.com (Alex Rier) Date: Wed, 13 Sep 2006 10:36:51 +0200 Subject: [Live-devel] H.264 RTP Streaming Message-ID: <683BC86C0162454BAC43B789A7FB196434380B@herlios.adyoron.com> Hi, Following the discussion about "H.264 RTP Streaming", I've found That if you encapsulate RAW H.264 in MPEG TS container (test.ts) Using Mpeg4ip mp4creator or similar utilities, then testOnDemandRTSPServer and similar demos can stream that file. Can you explain how does that happen? Thanks, Alex This mail passed through VIGILANT TECHNOLOGY Mail-SeCure. ************************************************************************************ This footnote confirms that this email message has been scanned by PineApp Mail-SeCure for the presence of malicious code, vandals & computer viruses. ************************************************************************************ From finlayson at live555.com Wed Sep 13 00:38:50 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Sep 2006 00:38:50 -0700 Subject: [Live-devel] '/' in streamName for rtsp server url? In-Reply-To: References: Message-ID: >Is there some issue with having a '/' in the streamName? Yes. Our RTSP server implementation does not allow this (because it uses '/' for selecting individual substreams (of an audio+video session). > Do I >need to so something special (like create ServerMediaSessions each of >the sub-paths) to get this to work? No, you can't do this with the current code. You'll have to omit '/' characters from the stream name. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Wed Sep 13 00:52:44 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Sep 2006 00:52:44 -0700 Subject: [Live-devel] H.264 RTP Streaming In-Reply-To: <683BC86C0162454BAC43B789A7FB196434380B@herlios.adyoron.com> References: <683BC86C0162454BAC43B789A7FB196434380B@herlios.adyoron.com> Message-ID: >Following the discussion about "H.264 RTP Streaming", I've found >That if you encapsulate RAW H.264 in MPEG TS container (test.ts) >Using Mpeg4ip mp4creator or similar utilities, then >testOnDemandRTSPServer >and similar demos can stream that file. > >Can you explain how does that happen? This shouldn't surprise you. Our code for streaming MPEG Transport Streams doesn't care what kind of data is contained within the stream. As long as the Transport Stream is well-formed, our code will stream it. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From cnraghavendra at gmail.com Wed Sep 13 01:47:45 2006 From: cnraghavendra at gmail.com (Raghu CN) Date: Wed, 13 Sep 2006 14:17:45 +0530 Subject: [Live-devel] regarding video stream... Message-ID: <4084bc0b0609130147h628499a6u80a5d33e6eede1db@mail.gmail.com> HI, As I am running testMPEG1or2AudioVideoStreamer.exe with test.mpg, but I am getting error message as Beginning streaming... Beginning to read from file... BasicTaskScheduler::SingleStep(): select() fails: No error Kindly can any one help me in solving this problem. Regards, Raghavendra C N -- Thanks & Regards Raghavendra.C.N. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060913/c22be69d/attachment.html From rajeshkumar.r at imimobile.com Wed Sep 13 02:20:20 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Wed, 13 Sep 2006 14:50:20 +0530 Subject: [Live-devel] regarding video stream... References: <4084bc0b0609130147h628499a6u80a5d33e6eede1db@mail.gmail.com> Message-ID: <00af01c6d715$d5805400$f801000a@imidomain.com> hi Raghu Please #define READ_FROM_FILES_SYNCHRONOUSLY in ByteStreamFileSource. and try again with regards rajesh ----- Original Message ----- From: Raghu CN To: live-devel at ns.live555.com Sent: Wednesday, September 13, 2006 2:17 PM Subject: [Live-devel] regarding video stream... HI, As I am running testMPEG1or2AudioVideoStreamer.exe with test.mpg, but I am getting error message as Beginning streaming... Beginning to read from file... BasicTaskScheduler::SingleStep(): select() fails: No error Kindly can any one help me in solving this problem. Regards, Raghavendra C N -- Thanks & Regards Raghavendra.C.N. ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060913/228d57ee/attachment.html From cnraghavendra at gmail.com Wed Sep 13 03:58:14 2006 From: cnraghavendra at gmail.com (Raghu CN) Date: Wed, 13 Sep 2006 16:28:14 +0530 Subject: [Live-devel] regarding video stream... In-Reply-To: <00af01c6d715$d5805400$f801000a@imidomain.com> References: <4084bc0b0609130147h628499a6u80a5d33e6eede1db@mail.gmail.com> <00af01c6d715$d5805400$f801000a@imidomain.com> Message-ID: <4084bc0b0609130358n465213d5nf0cb0a5bfb0586ad@mail.gmail.com> hey thanks a lot rajesh.... now there is no error for transmitting the video stream. but at the receive end i am playing the received stream in winamp by giving the command as "winamp rtp://239.255.42.42:8888 " i am getting "timeout" in winamp. can u suggest me whats wrong with this. On 9/13/06, Rajesh wrote: > > hi Raghu > > Please #define READ_FROM_FILES_SYNCHRONOUSLY > in ByteStreamFileSource. and try again > > with regards > rajesh > > ----- Original Message ----- > *From:* Raghu CN > *To:* live-devel at ns.live555.com > *Sent:* Wednesday, September 13, 2006 2:17 PM > *Subject:* [Live-devel] regarding video stream... > > > > HI, > > > > As I am running testMPEG1or2AudioVideoStreamer.exe with test.mpg, but I am > getting error message as > > > > Beginning streaming... > > Beginning to read from file... > > BasicTaskScheduler::SingleStep(): select() fails: No error > > > > Kindly can any one help me in solving this problem. > > > > Regards, > > Raghavendra C N > > > > > -- > Thanks & Regards > Raghavendra.C.N. > > ------------------------------ > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > -- Thanks & Regards Raghavendra.C.N. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060913/60030260/attachment.html From xcsmith at rockwellcollins.com Wed Sep 13 13:54:44 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Wed, 13 Sep 2006 15:54:44 -0500 Subject: [Live-devel] please help with understanding trick mode Message-ID: Hello LIVE! I have some questions about understanding trick mode. When I look through the WAV and MP3 trick mode implementations, I notice that the file pointers are skipped ahead by using the scale factor, like in WAVAudioFileSource::doGetNextFrame --> // We read every 'fScaleFactor'th sample: 00222 fFrameSize = 0; 00223 while (bytesToRead > 0) { 00224 size_t bytesRead = fread(fTo, 1, bytesPerSample, fFid); 00225 if (bytesRead <= 0) break; 00226 fTo += bytesRead; 00227 fFrameSize += bytesRead; 00228 bytesToRead -= bytesRead; 00229 00230 // Seek to the appropriate place for the next sample: 00231 fseek(fFid, (fScaleFactor-1)*bytesPerSample, SEEK_CUR); So, what I think this does is: skip a little ahead, read a short sample, repeat. So is LIVE actually just playing little "chunks" of the audio stream, instead of simulating what an analog tape would sound like(you hear everything, but at a higher pitch)if you played it faster? I am looking at this code because I want to have trick mode available to a client that can only read Transport Streams. I don't think it will work to just skip the file pointer a little ahead in a TS file? Also, my TS files will contain 1 MPEG2 video ES and 1 MPEG2 audio ES, both of them synchronized with TS PCR set. Doesn't a decoder expect that only video key frames will be sent during trick mode? I don't see any way to set a constant so that if I seek my file pointer ahead in my file by that amount, I will get a key frame on my next call to doGetNextFrame(). So, now I have to rip out my ES streams, parse them out so I can set DSM-CC trick mode flags, find only the key frames, and put just the key frames back in a transport stream to send? Can you please tell me if I am thinking the correct way about this, and/or provide advice on how to approach this problem? I've tried to understand it by reading the FAQ, but I don't understand how it can work to jump the file pointer ahead in my TS file by a set amount, or how doing so can set the DSM-CC trick mode flag so that the decoder will ignore ES presentation times. I also notice this in the FAQ: Support for 'trick play' (fast forward and reverse play) on MPEG Transport Stream files has been implemented, but as a custom project - not part of the public "LIVE555 Streaming Media" source code release. (It is possible, however, that sometime in late 2006, similar functionality might also become part of the public "LIVE555 Streaming Media" code.) Any updates to this information? Thanks! ~Medra From finlayson at live555.com Wed Sep 13 15:37:02 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Sep 2006 15:37:02 -0700 Subject: [Live-devel] please help with understanding trick mode In-Reply-To: References: Message-ID: >I have some questions about understanding trick mode. When I look through >the WAV and MP3 trick mode implementations, I notice that the file pointers >are skipped ahead by using the scale factor, like in >WAVAudioFileSource::doGetNextFrame --> > > > // We read every 'fScaleFactor'th sample: > 00222 fFrameSize = 0; > 00223 while (bytesToRead > 0) { > 00224 size_t bytesRead = fread(fTo, 1, bytesPerSample, fFid); > 00225 if (bytesRead <= 0) break; > 00226 fTo += bytesRead; > 00227 fFrameSize += bytesRead; > 00228 bytesToRead -= bytesRead; > 00229 > 00230 // Seek to the appropriate place for the next sample: > 00231 fseek(fFid, (fScaleFactor-1)*bytesPerSample, SEEK_CUR); > > So, what I think this does is: skip a little ahead, read a short sample, > repeat. So is LIVE actually just playing little "chunks" of the audio > stream Yes. This works because samples in WAV audio files are self contained, and independent - so it's possible to get n-times speedup simply by selecting every nth sample in the file. Unfortunately many other media types (especially MPEG Transport Streams) are not like this, so this simple approach won't work for them. >I also notice this in the FAQ: >Support for 'trick play' (fast forward and reverse play) on MPEG Transport >Stream files has been implemented, but as a custom project - not part of >the public "LIVE555 Streaming Media" source code release. (It is possible, >however, that sometime in late 2006, similar functionality might also >become part of the public "LIVE555 Streaming Media" code.) >Any updates to this information? Yes. Replace "possible" with "likely". -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From freegnu at 163.com Wed Sep 13 19:53:32 2006 From: freegnu at 163.com (freegnu) Date: Thu, 14 Sep 2006 10:53:32 +0800 Subject: [Live-devel] how to deliver mp4 file Message-ID: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang> hi all, im a newbie to livemdia and mediaprogramming i want to deliver mp4 file from a server, and the client use the realplayer to play it. i find the testOnDemandRTSPServer.cpp just supports m4v format. i want to know , how to deliver the mp4 file to the client. please give me an advice thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060913/5e99eacb/attachment.html From rajeshkumar.r at imimobile.com Wed Sep 13 22:52:56 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 14 Sep 2006 11:22:56 +0530 Subject: [Live-devel] unable to get the Response from Server when issuing the command DESCRIBE References: Message-ID: <014f01c6d7c2$06b7bd90$f801000a@imidomain.com> Hi All I am connecting with the Darwin Server.defaukt Port is 554. and sending this request DESCRIBE rtsp://10.0.1.248/8.3gp RTSP/1.0 CSeq: 1 Accept: application/x-rtsp-mh, application/sdp I am not getting any response. with regards rajesh From shaswata at alumnux.com Thu Sep 14 00:23:20 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Thu, 14 Sep 2006 12:53:20 +0530 Subject: [Live-devel] how to deliver mp4 file References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang> Message-ID: <002f01c6d7ce$a7a98f60$2e0aa8c0@NITU> Live doesn't yet support for ISO Base media file format and .mp4 file format is one such type. Instead you can use Darwin Streaming Server (DSS) to stream .mp4 file. With regards, Shaswata ----- Original Message ----- From: freegnu To: live-devel at ns.live555.com Sent: Thursday, September 14, 2006 8:23 AM Subject: [Live-devel] how to deliver mp4 file hi all, im a newbie to livemdia and mediaprogramming i want to deliver mp4 file from a server, and the client use the realplayer to play it. i find the testOnDemandRTSPServer.cpp just supports m4v format. i want to know , how to deliver the mp4 file to the client. please give me an advice thanks ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060914/30660d51/attachment.html From freegnu at 163.com Thu Sep 14 00:53:11 2006 From: freegnu at 163.com (freegnu) Date: Thu, 14 Sep 2006 15:53:11 +0800 Subject: [Live-devel] how to deliver mp4 file References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang> <002f01c6d7ce$a7a98f60$2e0aa8c0@NITU> Message-ID: <002501c6d7d2$d4543430$0501a8c0@zwenqiang> thanks for your answer btw: if i have a PS stream from encode, how to deliver it to the client, and can be played by realplay use the livemedia? ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 3:23 PM Subject: Re: [Live-devel] how to deliver mp4 file Live doesn't yet support for ISO Base media file format and .mp4 file format is one such type. Instead you can use Darwin Streaming Server (DSS) to stream .mp4 file. With regards, Shaswata ----- Original Message ----- From: freegnu To: live-devel at ns.live555.com Sent: Thursday, September 14, 2006 8:23 AM Subject: [Live-devel] how to deliver mp4 file hi all, im a newbie to livemdia and mediaprogramming i want to deliver mp4 file from a server, and the client use the realplayer to play it. i find the testOnDemandRTSPServer.cpp just supports m4v format. i want to know , how to deliver the mp4 file to the client. please give me an advice thanks ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060914/e621fc01/attachment.html From shaswata at alumnux.com Thu Sep 14 01:49:41 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Thu, 14 Sep 2006 14:19:41 +0530 Subject: [Live-devel] how to deliver mp4 file References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU> <002501c6d7d2$d4543430$0501a8c0@zwenqiang> Message-ID: <002201c6d7da$b7f3ac00$2e0aa8c0@NITU> Are you saying about MPEG Program Stream? If so, live can stream that - take a look at MPEG1or2VideoFileServerMediaSubsession (MPEG1or2VideoStreamFramer as FramedSource and MPEG1or2VideoRTPSink as RTPSink). There is no issue on the RealPlayer side as long as it gets MPEG1 video and MPEG Layer 2/3 audio as RTP streams. With regards, Shaswata ----- Original Message ----- From: freegnu To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 1:23 PM Subject: Re: [Live-devel] how to deliver mp4 file thanks for your answer btw: if i have a PS stream from encode, how to deliver it to the client, and can be played by realplay use the livemedia? ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 3:23 PM Subject: Re: [Live-devel] how to deliver mp4 file Live doesn't yet support for ISO Base media file format and .mp4 file format is one such type. Instead you can use Darwin Streaming Server (DSS) to stream .mp4 file. With regards, Shaswata ----- Original Message ----- From: freegnu To: live-devel at ns.live555.com Sent: Thursday, September 14, 2006 8:23 AM Subject: [Live-devel] how to deliver mp4 file hi all, im a newbie to livemdia and mediaprogramming i want to deliver mp4 file from a server, and the client use the realplayer to play it. i find the testOnDemandRTSPServer.cpp just supports m4v format. i want to know , how to deliver the mp4 file to the client. please give me an advice thanks -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060914/83cd27a6/attachment-0001.html From freegnu at 163.com Thu Sep 14 02:00:48 2006 From: freegnu at 163.com (freegnu) Date: Thu, 14 Sep 2006 17:00:48 +0800 Subject: [Live-devel] how to deliver mp4 file References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang> <002201c6d7da$b7f3ac00$2e0aa8c0@NITU> Message-ID: <004701c6d7dc$4635b660$0501a8c0@zwenqiang> thanks , i will read it first ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 4:49 PM Subject: Re: [Live-devel] how to deliver mp4 file Are you saying about MPEG Program Stream? If so, live can stream that - take a look at MPEG1or2VideoFileServerMediaSubsession (MPEG1or2VideoStreamFramer as FramedSource and MPEG1or2VideoRTPSink as RTPSink). There is no issue on the RealPlayer side as long as it gets MPEG1 video and MPEG Layer 2/3 audio as RTP streams. With regards, Shaswata ----- Original Message ----- From: freegnu To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 1:23 PM Subject: Re: [Live-devel] how to deliver mp4 file thanks for your answer btw: if i have a PS stream from encode, how to deliver it to the client, and can be played by realplay use the livemedia? ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 3:23 PM Subject: Re: [Live-devel] how to deliver mp4 file Live doesn't yet support for ISO Base media file format and .mp4 file format is one such type. Instead you can use Darwin Streaming Server (DSS) to stream .mp4 file. With regards, Shaswata ----- Original Message ----- From: freegnu To: live-devel at ns.live555.com Sent: Thursday, September 14, 2006 8:23 AM Subject: [Live-devel] how to deliver mp4 file hi all, im a newbie to livemdia and mediaprogramming i want to deliver mp4 file from a server, and the client use the realplayer to play it. i find the testOnDemandRTSPServer.cpp just supports m4v format. i want to know , how to deliver the mp4 file to the client. please give me an advice thanks ------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060914/394c2f74/attachment.html From rajeshkumar.r at imimobile.com Thu Sep 14 04:00:56 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 14 Sep 2006 16:30:56 +0530 Subject: [Live-devel] HI References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang> <002201c6d7da$b7f3ac00$2e0aa8c0@NITU> Message-ID: <02f101c6d7ed$0e102d40$f801000a@imidomain.com> Hi All when sending Describe and Options I am getting the Proper response. but when I am sending SETUP command I am getting the Bad Request. can any one help me. SETUP rtsp://10.0.1.248/8.3gp RTSP/1.0 CSeq: 3 Transport: RTP/AVP //######RESPONSE ****RTSP/1.0 400 Bad Request Server: DSS/5.5.1 (Build/489.8; Platform/Win32; Release/Darwin; ) Cseq: 3 Connection: Close ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 2:19 PM Subject: Re: [Live-devel] how to deliver mp4 file Are you saying about MPEG Program Stream? If so, live can stream that - take a look at MPEG1or2VideoFileServerMediaSubsession (MPEG1or2VideoStreamFramer as FramedSource and MPEG1or2VideoRTPSink as RTPSink). There is no issue on the RealPlayer side as long as it gets MPEG1 video and MPEG Layer 2/3 audio as RTP streams. With regards, Shaswata ----- Original Message ----- From: freegnu To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 1:23 PM Subject: Re: [Live-devel] how to deliver mp4 file thanks for your answer btw: if i have a PS stream from encode, how to deliver it to the client, and can be played by realplay use the livemedia? ----- Original Message ----- From: Shaswata Jash To: LIVE555 Streaming Media - development & use Sent: Thursday, September 14, 2006 3:23 PM Subject: Re: [Live-devel] how to deliver mp4 file Live doesn't yet support for ISO Base media file format and .mp4 file format is one such type. Instead you can use Darwin Streaming Server (DSS) to stream .mp4 file. With regards, Shaswata ----- Original Message ----- From: freegnu To: live-devel at ns.live555.com Sent: Thursday, September 14, 2006 8:23 AM Subject: [Live-devel] how to deliver mp4 file hi all, im a newbie to livemdia and mediaprogramming i want to deliver mp4 file from a server, and the client use the realplayer to play it. i find the testOnDemandRTSPServer.cpp just supports m4v format. i want to know , how to deliver the mp4 file to the client. please give me an advice thanks ------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060914/446b6666/attachment.html From rajeshkumar.r at imimobile.com Thu Sep 14 04:04:03 2006 From: rajeshkumar.r at imimobile.com (Rajesh) Date: Thu, 14 Sep 2006 16:34:03 +0530 Subject: [Live-devel] Not Getting Proper Response of SETUP References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang> <002201c6d7da$b7f3ac00$2e0aa8c0@NITU> Message-ID: <032901c6d7ed$7d487780$f801000a@imidomain.com> Hi All when sending Describe and Options I am getting the Proper response. but when I am sending SETUP command I am getting the Bad Request. can any one help me. SETUP rtsp://10.0.1.248/8.3gp RTSP/1.0 CSeq: 3 Transport: RTP/AVP //######RESPONSE ****RTSP/1.0 400 Bad Request Server: DSS/5.5.1 (Build/489.8; Platform/Win32; Release/Darwin; ) Cseq: 3 Connection: Close with regards rajesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060914/fc52568a/attachment-0001.html From awolf at rhythmnewmedia.com Thu Sep 14 07:54:12 2006 From: awolf at rhythmnewmedia.com (Amir Wolf) Date: Thu, 14 Sep 2006 07:54:12 -0700 Subject: [Live-devel] Not Getting Proper Response of SETUP References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang><002201c6d7da$b7f3ac00$2e0aa8c0@NITU> <032901c6d7ed$7d487780$f801000a@imidomain.com> Message-ID: <9FA71E73BF462E4C96C3A9C074D50F707257BF@DHOST001-39.DEX001.intermedia.net> Setup request should include the track identifier in the url. This is extracted from the SDP, it will be helpfull if you can send the response you got for the describe request. Amir -----Original Message----- From: live-devel-bounces at ns.live555.com on behalf of Rajesh Sent: Thu 14/09/2006 04:04 To: LIVE555 Streaming Media - development & use Subject: [Live-devel] Not Getting Proper Response of SETUP Hi All when sending Describe and Options I am getting the Proper response. but when I am sending SETUP command I am getting the Bad Request. can any one help me. SETUP rtsp://10.0.1.248/8.3gp RTSP/1.0 CSeq: 3 Transport: RTP/AVP //######RESPONSE ****RTSP/1.0 400 Bad Request Server: DSS/5.5.1 (Build/489.8; Platform/Win32; Release/Darwin; ) Cseq: 3 Connection: Close with regards rajesh -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3226 bytes Desc: not available Url : http://lists.live555.com/pipermail/live-devel/attachments/20060914/ddb12307/attachment.bin From ymreddy at ssdi.sharp.co.in Fri Sep 15 00:04:57 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Fri, 15 Sep 2006 12:34:57 +0530 Subject: [Live-devel] Out of order packets Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CF90@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Everybody, We are using LIVE server (testMPEG1or2VideoStreamer.cpp) and client (testMPEG1or2VideoReceiver.cpp) to stream MPEG data. We found that when the sequence number is wrapped around, and when the server sends out of order packets, the calculation of newSeqNum is found to be wrong in the function noteIncomingPacket() in file RTPSource.cpp. For example, when the server starts sending the stream with sequence number starting from 65000 and the sequence number is wrapped wround, and when the server try to send the out of order packet, for example, with the sequence number 65500, then the calculation of newSeqNum seems to be wrong. We need to store this out of order packet with the sequence number 65500 in the correct order. Since this calculation gives some incorrect value, this out of order packet is getting stored in a wrong place. Any clues on how to correct the above problem. Thanks and Regards Y. Mallikharjuna Reddy From finlayson at live555.com Fri Sep 15 00:58:23 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 15 Sep 2006 00:58:23 -0700 Subject: [Live-devel] Out of order packets In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520337CF90@ssdimailsrvnt01.ssdi.sharp.co.in > References: <7FB4685EA93D014C8E30AA087B66E7520337CF90@ssdimailsrvnt01.ssdi.sharp.co.in > Message-ID: >We are using LIVE server (testMPEG1or2VideoStreamer.cpp) and client >(testMPEG1or2VideoReceiver.cpp) to stream MPEG data. We found that when the >sequence number is wrapped around, and when the server sends out of order >packets Our server code should not be sending out-of-order packets! It's always possible that the network (between the server and client) may reorder some packets, but the server should not be sending them out of order. >, the calculation of newSeqNum is found to be wrong in the function >noteIncomingPacket() in file RTPSource.cpp. For example, when the server >starts sending the stream with sequence number starting from 65000 and the >sequence number is wrapped wround, and when the server try to send the out >of order packet, for example, with the sequence number 65500, then the >calculation of newSeqNum seems to be wrong. We need to store this out of >order packet with the sequence number 65500 in the correct order. Since this >calculation gives some incorrect value, this out of order packet is getting >stored in a wrong place. I don't understand the problem that you're referring to. I suspect that you are misunderstanding the purpose of the "noteIncomingPacket()" function (note that it just records *statistics*; it doesn't actually store the packet). If you have a specific example (illustrated with a specific series of incoming sequence numbers) that demonstrates the 'problem', then let us know (with a detailed explanation of what you think the problem is). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From ymreddy at ssdi.sharp.co.in Fri Sep 15 02:42:41 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Fri, 15 Sep 2006 15:12:41 +0530 Subject: [Live-devel] Out of order packets Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CF91@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Ross, Thanks for the information. We are simulating in the server program to send the out of order packets. In the client side, we are storing the received packets numbers into a buffer. This buffer is a linked list, which is sorted based on the sequence numbers. When the sequence numbers are wrapped, since the sequence numbers start with 0, we are using this variable "newSeqNum", in our linked list. This variable "newSeqNum" gives the extended sequence numbers, beyond the magic number 65535. In this scenario, when the out of order packet comes, then this variable "newSeqNum" gives wrong value. Following example illustrates the scenario: Assume the server is sending the stream starting with sequence number 65000. Assume there is a out of order packet for the sequence number 65500. When the sequence number is wrapped around, the sequence number starts with zero. In the client side, in the noteIncomingPacket() function, the variable "newSeqNum" becomes 65536. At this point of time, if the client receives out of order packet with sequence number 65500, then the variable "newSeqNum" becomes 131036, which we feel is a wrong number. Please provide your suggesions. Thanks and Regards Y. Mallikharjuna Reddy -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com]On Behalf Of Ross Finlayson Sent: Friday, September 15, 2006 1:28 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Out of order packets >We are using LIVE server (testMPEG1or2VideoStreamer.cpp) and client >(testMPEG1or2VideoReceiver.cpp) to stream MPEG data. We found that when the >sequence number is wrapped around, and when the server sends out of order >packets Our server code should not be sending out-of-order packets! It's always possible that the network (between the server and client) may reorder some packets, but the server should not be sending them out of order. >, the calculation of newSeqNum is found to be wrong in the function >noteIncomingPacket() in file RTPSource.cpp. For example, when the server >starts sending the stream with sequence number starting from 65000 and the >sequence number is wrapped wround, and when the server try to send the out >of order packet, for example, with the sequence number 65500, then the >calculation of newSeqNum seems to be wrong. We need to store this out of >order packet with the sequence number 65500 in the correct order. Since this >calculation gives some incorrect value, this out of order packet is getting >stored in a wrong place. I don't understand the problem that you're referring to. I suspect that you are misunderstanding the purpose of the "noteIncomingPacket()" function (note that it just records *statistics*; it doesn't actually store the packet). If you have a specific example (illustrated with a specific series of incoming sequence numbers) that demonstrates the 'problem', then let us know (with a detailed explanation of what you think the problem is). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel From finlayson at live555.com Fri Sep 15 16:29:39 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 15 Sep 2006 16:29:39 -0700 Subject: [Live-devel] Out of order packets In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520337CF91@ssdimailsrvnt01.ssdi.sharp.co.in > References: <7FB4685EA93D014C8E30AA087B66E7520337CF91@ssdimailsrvnt01.ssdi.sharp.co.in > Message-ID: >Assume the server is sending the stream starting with sequence number 65000. >Assume there is a out of order packet for the sequence number 65500. When >the sequence number is wrapped around, the sequence number starts with zero. >In the client side, in the noteIncomingPacket() function, the variable >"newSeqNum" becomes 65536. At this point of time, if the client receives out >of order packet with sequence number 65500, then the variable "newSeqNum" >becomes 131036, which we feel is a wrong number. OK, thanks for the explanation. You can fix this problem by replacing the following code (in "RTPSource.cpp"): // Check whether the sequence number has wrapped around: unsigned seqNumCycle = (fHighestExtSeqNumReceived&0xFFFF0000); unsigned oldSeqNum = (fHighestExtSeqNumReceived&0xFFFF); unsigned seqNumDifference = (unsigned)((int)seqNum-(int)oldSeqNum); if (seqNumDifference >= 0x8000 && seqNumLT((u_int16_t)oldSeqNum, seqNum)) { // sequence number wrapped around => start a new cycle: seqNumCycle += 0x10000; } unsigned newSeqNum = seqNumCycle|seqNum; if (newSeqNum > fHighestExtSeqNumReceived) { fHighestExtSeqNumReceived = newSeqNum; } with // Check whether the new sequence number is the highest yet seen: unsigned oldSeqNum = (fHighestExtSeqNumReceived&0xFFFF); if (seqNumLT((u_int16_t)oldSeqNum, seqNum)) { // This packet was not an old packet received out of order, so check it: unsigned seqNumCycle = (fHighestExtSeqNumReceived&0xFFFF0000); unsigned seqNumDifference = (unsigned)((int)seqNum-(int)oldSeqNum); if (seqNumDifference >= 0x8000) { // The sequence number wrapped around, so start a new cycle: seqNumCycle += 0x10000; } unsigned newSeqNum = seqNumCycle|seqNum; if (newSeqNum > fHighestExtSeqNumReceived) { fHighestExtSeqNumReceived = newSeqNum; } } This change will be included in the next release of the software. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From ivanjia at hotmail.com Sat Sep 16 02:14:56 2006 From: ivanjia at hotmail.com (jia junfeng) Date: Sat, 16 Sep 2006 17:14:56 +0800 Subject: [Live-devel] To support multicast loopback for win32 Message-ID: Hi, All, Winsock support mutlicast loopback, but in GroupsockHelper.cpp, setupDatagramSocket() function will not set loopback for win32. To use multicast loopback in win32, you need to move the code out of the #define. Thanks, Ivan From finlayson at live555.com Sun Sep 17 19:56:00 2006 From: finlayson at live555.com (Ross Finlayson) Date: Sun, 17 Sep 2006 19:56:00 -0700 Subject: [Live-devel] To support multicast loopback for win32 In-Reply-To: References: Message-ID: >Winsock support mutlicast loopback, but in GroupsockHelper.cpp, >setupDatagramSocket() function will not set loopback for win32. >To use multicast loopback in win32, you need to move the code >out of the #define. An earlier version of Windows (prior to XP) had problems with "setsockopt(..., IP_MULTICAST_LOOP)". Perhaps this has been fixed in XP?? Anyway, I've now moved this code out of the #define. We'll see if anyone complains about this... -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From mailinglists at erwinrol.com Mon Sep 18 10:28:19 2006 From: mailinglists at erwinrol.com (Erwin Rol) Date: Mon, 18 Sep 2006 19:28:19 +0200 Subject: [Live-devel] liveMedia cmake file Message-ID: <1158600499.2892.113.camel@xpc.home.erwinrol.com> Hey all, Attached is a patch that adds a number of CMakeLists.txt files to the liveMedia library that can be used to build the library with CMake. I only tested it on Linux but it should work on most Unix systems. Mac and Windows are also supported but they might need some tuning. Since the files don't remove or change anything i hope they can be added and with time, when every supported platform works correct, replace the current Makefile hacks. - Erwin -------------- next part -------------- A non-text attachment was scrubbed... Name: live_cmake.diff Type: text/x-patch Size: 12264 bytes Desc: not available Url : http://lists.live555.com/pipermail/live-devel/attachments/20060918/4987492e/attachment-0001.bin From freegnu at 163.com Mon Sep 18 22:39:03 2006 From: freegnu at 163.com (freegnu) Date: Tue, 19 Sep 2006 13:39:03 +0800 Subject: [Live-devel] how to transfers mpeg4 PS stream? References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang><002201c6d7da$b7f3ac00$2e0aa8c0@NITU> <004701c6d7dc$4635b660$0501a8c0@zwenqiang> Message-ID: <005501c6dbad$eb351eb0$0501a8c0@zwenqiang> if i have a mpeg4 PS stream from encode, how to transfers it to the client? the testOnDemandRTSPServer seens not to support it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060918/8d4d90d6/attachment.html From shaswata at alumnux.com Tue Sep 19 00:09:18 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Tue, 19 Sep 2006 12:39:18 +0530 Subject: [Live-devel] how to transfers mpeg4 PS stream? References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang><002201c6d7da$b7f3ac00$2e0aa8c0@NITU><004701c6d7dc$4635b660$0501a8c0@zwenqiang> <005501c6dbad$eb351eb0$0501a8c0@zwenqiang> Message-ID: <002101c6dbba$85efe7d0$2e0aa8c0@NITU> What do you mean by MPEG4 PS ? There is obviously MPEG1/2 PS or MPEG4 video ES may even reside in MPEG1/2 TS. But probably there is no familiar concept of MPEG4 PS (Program Stream). Would you clarify it? With regards, Shaswata ----- Original Message ----- From: freegnu To: LIVE555 Streaming Media - development & use Sent: Tuesday, September 19, 2006 11:09 AM Subject: [Live-devel] how to transfers mpeg4 PS stream? if i have a mpeg4 PS stream from encode, how to transfers it to the client? the testOnDemandRTSPServer seens not to support it ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060918/c3d4405b/attachment.html From sdhays.neon.com.tw at gmail.com Tue Sep 19 00:47:07 2006 From: sdhays.neon.com.tw at gmail.com (Scott Hays) Date: Tue, 19 Sep 2006 15:47:07 +0800 Subject: [Live-devel] how to transfers mpeg4 PS stream? In-Reply-To: <002101c6dbba$85efe7d0$2e0aa8c0@NITU> References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang> <002f01c6d7ce$a7a98f60$2e0aa8c0@NITU> <002501c6d7d2$d4543430$0501a8c0@zwenqiang> <002201c6d7da$b7f3ac00$2e0aa8c0@NITU> <004701c6d7dc$4635b660$0501a8c0@zwenqiang> <005501c6dbad$eb351eb0$0501a8c0@zwenqiang> <002101c6dbba$85efe7d0$2e0aa8c0@NITU> Message-ID: <9866ce4f0609190047h346ca764p43eee4fc3bc3a356@mail.gmail.com> MPEG PS can contain MPEG4 content. But program stream support for MPEG4 is not currently available in LIVE555. It is possible, though. I had a hack sort of working last year, but it wasn't good enough to submit as a patch (and we decided to move to a different container format before I perfected it). You'll need to edit MPEG1or2Demux.cpp, I believe, and then make sure that the demuxed MPEG4-ES stream gets parsed by the appropriate MPEG4Video*Framer (I can't remember if you can use the discrete framer or not, although I think you can't use the discrete framer). Scott On 9/19/06, Shaswata Jash wrote: > > What do you mean by MPEG4 PS ? There is obviously MPEG1/2 PS or MPEG4 > video ES may even reside in MPEG1/2 TS. But probably there is no familiar > concept of MPEG4 PS (Program Stream). Would you clarify it? > > With regards, > Shaswata > ----- Original Message ----- > *From:* freegnu > *To:* LIVE555 Streaming Media - development & use > *Sent:* Tuesday, September 19, 2006 11:09 AM > *Subject:* [Live-devel] how to transfers mpeg4 PS stream? > > if i have a mpeg4 PS stream from encode, how to transfers it to the > client? > the testOnDemandRTSPServer seens not to support it > > ------------------------------ > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060919/5dee245a/attachment.html From freegnu at 163.com Tue Sep 19 01:12:43 2006 From: freegnu at 163.com (freegnu) Date: Tue, 19 Sep 2006 16:12:43 +0800 Subject: [Live-devel] how to transfers mpeg4 PS stream? References: <005a01c6d7a8$f74d3e70$0501a8c0@zwenqiang><002f01c6d7ce$a7a98f60$2e0aa8c0@NITU><002501c6d7d2$d4543430$0501a8c0@zwenqiang><002201c6d7da$b7f3ac00$2e0aa8c0@NITU><004701c6d7dc$4635b660$0501a8c0@zwenqiang><005501c6dbad$eb351eb0$0501a8c0@zwenqiang><002101c6dbba$85efe7d0$2e0aa8c0@NITU> <9866ce4f0609190047h346ca764p43eee4fc3bc3a356@mail.gmail.com> Message-ID: <002c01c6dbc3$6279e860$0501a8c0@zwenqiang> thanks im a newbie to this domain, and i am anxious to embed a library to montavista, so that it can transfers the MPEG4 program stream to the client, and the client uses VLC to play it. is there another library which open source for embeding montavista ----- Original Message ----- From: Scott Hays To: LIVE555 Streaming Media - development & use Sent: Tuesday, September 19, 2006 3:47 PM Subject: Re: [Live-devel] how to transfers mpeg4 PS stream? MPEG PS can contain MPEG4 content. But program stream support for MPEG4 is not currently available in LIVE555. It is possible, though. I had a hack sort of working last year, but it wasn't good enough to submit as a patch (and we decided to move to a different container format before I perfected it). You'll need to edit MPEG1or2Demux.cpp, I believe, and then make sure that the demuxed MPEG4-ES stream gets parsed by the appropriate MPEG4Video*Framer (I can't remember if you can use the discrete framer or not, although I think you can't use the discrete framer). Scott On 9/19/06, Shaswata Jash wrote: What do you mean by MPEG4 PS ? There is obviously MPEG1/2 PS or MPEG4 video ES may even reside in MPEG1/2 TS. But probably there is no familiar concept of MPEG4 PS (Program Stream). Would you clarify it? With regards, Shaswata ----- Original Message ----- From: freegnu To: LIVE555 Streaming Media - development & use Sent: Tuesday, September 19, 2006 11:09 AM Subject: [Live-devel] how to transfers mpeg4 PS stream? if i have a mpeg4 PS stream from encode, how to transfers it to the client? the testOnDemandRTSPServer seens not to support it ---------------------------------------------------------------------------- _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060919/caff465a/attachment.html From rahul.ray at gmail.com Tue Sep 19 15:54:05 2006 From: rahul.ray at gmail.com (Rahul Ray) Date: Tue, 19 Sep 2006 15:54:05 -0700 Subject: [Live-devel] mdatFloating point exception Message-ID: Hi Gurus, I am working on an academic project where I need to open an rtsp stream and save it intermediately. I am going to use openRTSP code by you. But, here it gives a problem "mdatFloating point exception". Any quick help would be greatly appreciated. I compiled live on linux ( using redhat). Is there any commandline arguments that can help save avi/mov/mpeg-4 format using openRTSP ? -best, Dr. Rahul Ray ps. The exception occurs with -i, -4 too. ps. I would like to store the recorded video preferably in avi format or mov or mpeg-4. [ipicsspm2 197] ~/live/testProgs > openRTSP -q -e 10 rtsp://64.210.18.18/mpeg4?cam=0/ Warning: The -q, -4 or -i option was used, but not -w. Assuming a video width of 240 pixels Warning: The -q, -4 or -i option was used, but not -h. Assuming a video height of 180 pixels Warning: The -q, -4 or -i option was used, but not -f. Assuming a video frame rate of 15 frames-per-second Opened URL "rtsp://64.210.18.18/mpeg4?cam=0/", returning a SDP description: v=0 o=MangoDSP 9487895444231339407 9487895444231339407 IN IP4 64.210.18.18 s=Mango DSP Audio/Video c=IN IP4 0.0.0.0 m=video 0 RTP/AVP 96 a=rtpmap:96 MP4V-ES/90000 a=fmtp:96 profile-level-id=1;config=000001B003000001B50900000101000001200086E00024E22003E80068824800F628B021E0A31F000001B3001007 a=control:__StreamID=2149782080 Created receiver for "video/MP4V-ES" subsession (client ports 32794-32795) Setup "video/MP4V-ES" subsession (client ports 32794-32795) Started playing session Receiving streamed data (for up to 10.000000 seconds)... mdatFloating point exception From finlayson at live555.com Tue Sep 19 16:06:51 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 19 Sep 2006 16:06:51 -0700 Subject: [Live-devel] mdatFloating point exception In-Reply-To: References: Message-ID: >Hi Gurus, > I am working on an academic project where I need to open an rtsp >stream and save it intermediately. I am going to use openRTSP code >by you. >But, here it gives a problem "mdatFloating point exception". I suggest recompiling the code with "-g" (to do this, add "-g" to the COMPILE_OPTS line in config.linux, rerun "genMakefiles linux", and "make"). Then, rerun the program under "gdb", to see where the error occurs. (BTW, note that "openRTSP -q" (or "-4" or "-i" outputs to 'stdout'.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From rahul.ray at gmail.com Tue Sep 19 16:55:32 2006 From: rahul.ray at gmail.com (Rahul Ray) Date: Tue, 19 Sep 2006 16:55:32 -0700 Subject: [Live-devel] black screen always Message-ID: Hi, I am doing: ./openRTSP -b 80000 -i rtsp://64.210.18.18/mpeg4?cam=0/ > see.avi The log is: Warning: The -q, -4 or -i option was used, but not -w. Assuming a video width of 240 pixels Warning: The -q, -4 or -i option was used, but not -h. Assuming a video height of 180 pixels Warning: The -q, -4 or -i option was used, but not -f. Assuming a video frame rate of 15 frames-per-second Opened URL "rtsp://64.210.18.18/mpeg4?cam=0/", returning a SDP description: v=0 o=MangoDSP 9487915909729030003 9487915909729030003 IN IP4 64.210.18.18 s=Mango DSP Audio/Video c=IN IP4 0.0.0.0 m=video 0 RTP/AVP 96 a=rtpmap:96 MP4V-ES/90000 a=fmtp:96 profile-level-id=1;config=000001B003000001B50900000101000001200086E00024E22003E80068824800F628B021E0A31F000001B3001007 a=control:__StreamID=2149779624 Created receiver for "video/MP4V-ES" subsession (client ports 32796-32797) Setup "video/MP4V-ES" subsession (client ports 32796-32797) Started playing session Receiving streamed data (signal with "kill -HUP 24035" or "kill -USR1 24035" to terminate)... ---END OF LOG -------------- However, when I try to see see.avi, on windows, using BSPlayer/WMP I get a black screen ( without any error message). Any idea ? -best, Rahul From jeremy at electrosilk.net Wed Sep 20 00:06:05 2006 From: jeremy at electrosilk.net (Jeremy) Date: Wed, 20 Sep 2006 15:06:05 +0800 (WST) Subject: [Live-devel] Time Resolution and Accuracy ? Message-ID: <1584.202.189.68.42.1158735965.squirrel@mail.ardley.org> Hello Ross, Can you please give an estimate as to what the timer resolution and accuracy is for times generated or used in the code - in particular for linux systems runing standard 2.6 kernel. In various places in the code I see gettimeofday() being called to set a presentation time for some data. Is that time accurate to the microsecond? or some much larger value? In other cases a kernel driver will set a presentation time - again how accurate is that time? I also see in the UsageEnvironment code that you use a select function with timeout - I think to set a precise delay for your scheduler. Is this accurate - to the microsecond to allow the programe to contend for execution - or is it some much larger value? Thanks Jeremy From finlayson at live555.com Wed Sep 20 00:19:39 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Sep 2006 00:19:39 -0700 Subject: [Live-devel] Time Resolution and Accuracy ? In-Reply-To: <1584.202.189.68.42.1158735965.squirrel@mail.ardley.org> References: <1584.202.189.68.42.1158735965.squirrel@mail.ardley.org> Message-ID: >Can you please give an estimate as to what the timer resolution and >accuracy is for times generated or used in the code - in particular for >linux systems runing standard 2.6 kernel. These are OS-related questions, that are probably best answered on appropriate OS-related mailing lists. However, I should note that the "LIVE555 Streaming Media" code does not depend upon "gettimeofday()" being especially precise. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From jeremy at electrosilk.net Wed Sep 20 00:32:12 2006 From: jeremy at electrosilk.net (Jeremy) Date: Wed, 20 Sep 2006 15:32:12 +0800 (WST) Subject: [Live-devel] Time Resolution and Accuracy ? In-Reply-To: <1584.202.189.68.42.1158735965.squirrel@mail.ardley.org> References: <1584.202.189.68.42.1158735965.squirrel@mail.ardley.org> Message-ID: <1728.202.189.68.42.1158737532.squirrel@mail.ardley.org> Hello Ross, Thanks for the previous answer. Further searching shows that is is believed that gettimeofday is pretty accurate for Linux systems - ie microsecond accuracy. There was one item in my question (below) that you didn't answer. As the scheduler is used amongst other things to determine when to actually deliver a packet I expect it should be relatively accurate - so that data is delivered at a constant rate rather than in bursts a couple of times per video frame? > I also see in the UsageEnvironment code that you use a select function > with timeout - I think to set a precise delay for your scheduler. Is this > accurate - to the microsecond to allow the programe to contend for > execution > - or is it some much larger value? From millallo at gmail.com Wed Sep 20 07:31:37 2006 From: millallo at gmail.com (Emiliano Parasassi) Date: Wed, 20 Sep 2006 16:31:37 +0200 Subject: [Live-devel] ~H264VideoRTPSink bug In-Reply-To: References: <45056EBB.6000407@gmail.com> Message-ID: <9585da7b0609200731y7a27cbe1x5e8a71da524da6ae@mail.gmail.com> 2006/9/11, Ross Finlayson : > > Thanks. This will be fixed in the next release of the software. > -- Hi Ross, Are you sure you have fixed this? ;) Bye Emiliano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060920/eef3520f/attachment.html From finlayson at live555.com Wed Sep 20 11:14:47 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Sep 2006 11:14:47 -0700 Subject: [Live-devel] ~H264VideoRTPSink bug In-Reply-To: <9585da7b0609200731y7a27cbe1x5e8a71da524da6ae@mail.gmail.com> References: <45056EBB.6000407@gmail.com> <9585da7b0609200731y7a27cbe1x5e8a71da524da6ae@mail.gmail.com> Message-ID: >2006/9/11, Ross Finlayson ><finlayson at live555.com>: > >Thanks. This will be fixed in the next release of the software. >-- > > >Hi Ross, >Are you sure you have fixed this? ;) Oops - possibly not. Try changing line 53 of H264VideoRTPSink.cpp from delete fOurFragmenter; to Medium::close(fOurFragmenter); -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060920/ac9b7dcd/attachment.html From rahul.ray at gmail.com Wed Sep 20 11:39:58 2006 From: rahul.ray at gmail.com (Rahul Ray) Date: Wed, 20 Sep 2006 11:39:58 -0700 Subject: [Live-devel] how the frames are saved ? Message-ID: Hi, I am calling openRTSP -m option to save the frames. But, I could never see them using any image viewer. > file video-MP4V-ES-1-1158777340.853709 > video-MP4V-ES-1-1158777433.112991: data However, this works fine with video-JPEG-1-1158777554.422911 and I can see it. From tranzmatt at gmail.com Wed Sep 20 12:43:16 2006 From: tranzmatt at gmail.com (Tranzmatt) Date: Wed, 20 Sep 2006 15:43:16 -0400 Subject: [Live-devel] Looking for known working m4v file Message-ID: <423100740609201243o5528323crf642a1ac9374113e@mail.gmail.com> Is there an mpeg4 video file that is known to work with testMPEG4VideoStreamer and testOnDemandRTSPServer? I've tried just about every m4v file I can lay my hands on and even converted some basic mpeg1 into mpeg4 with mp4creator and ffmpeg and none of it will stream from liveMedia to vlc. Thanks, Matt From finlayson at live555.com Wed Sep 20 13:33:59 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Sep 2006 13:33:59 -0700 Subject: [Live-devel] how the frames are saved ? In-Reply-To: References: Message-ID: >Hi, > I am calling openRTSP -m option to save the frames. > > But, I could never see them using any image viewer. That's not surprising. Read again the section in the "openRTSP" document that describes the "-m" option. Also, BTW, I'm not going to be able to help you with your problem playing the saved AVI-format file. You're going to have to figure out yourself (from the content of the file) why it's not playing properly. However, I strongly suspect that it's (at least in part) because you're not telling "openRTSP" the correct video width, height and frame rate (using the "-w", "-h" and "-f" options). If you give the correct video information to "openRTSP", there's a much better chance that your saved AVI (or MOV or MPEG-4) format file will be playable. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060920/0bc71b9f/attachment.html From finlayson at live555.com Wed Sep 20 13:44:21 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Sep 2006 13:44:21 -0700 Subject: [Live-devel] Looking for known working m4v file In-Reply-To: <423100740609201243o5528323crf642a1ac9374113e@mail.gmail.com> References: <423100740609201243o5528323crf642a1ac9374113e@mail.gmail.com> Message-ID: >Is there an mpeg4 video file that is known to work with >testMPEG4VideoStreamer and testOnDemandRTSPServer? See > I've tried just >about every m4v file I can lay my hands on and even converted some >basic mpeg1 into mpeg4 with mp4creator and ffmpeg and none of it will >stream from liveMedia to vlc. These are not MPEG-4 Video Elementary Stream files, so they won't work. Unfortunately, Apple coopted (misused) the '.m4v' filename extension to mean "MPEG-4 video contained within a MPEG-4 format file", rather than "a MPEG-4 Video Elementary Stream file", which is how everyone else had been using the extension before. Because of this confusion, I may end up changing the code to use the extension ".m4e" (instead of ".m4v") to mean "a MPEG-4 Video Elementary Stream file". -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060920/3d901533/attachment-0001.html From sachin.sn at lgsoftindia.com Wed Sep 20 21:49:43 2006 From: sachin.sn at lgsoftindia.com (Sachin S N) Date: Thu, 21 Sep 2006 10:19:43 +0530 Subject: [Live-devel] Reading from File Message-ID: <607194575BD8CE46BCC8D747996FA7F901E167F7@SI-RD10-MS01.LGE.NET> Hi all, Im new to the this fourm . Im trying to write a parser for H 264, so I need info on how to read from the file or buffer in Live555 stack. I came to know that we can use functions like test4Bytes(),get4Bytes(),getByte(), SkipBytes() defined in StreamParser.hh can be used for the file operations. Are there any other methods to read from file or buffer? How can we find that EOF is reached while reading from file or buffer? Thanks in advance. Regards Sachin ######################################################### THIS EMAIL MESSAGE IS FOR THE SOLE USE OF THE INTENDED RECIPIENT(S) AND MAY CONTAIN CONFIDENTIAL AND PRIVILEGED INFORMATION. ANY UNAUTHORIZED REVIEW, USE, DISCLOSURE OR DISTRIBUTION IS PROHIBITED.BEFORE OPENING ANY ATTACHMENTS PLEASE CHECK FOR VIRUSES AND DEFECTS.IF YOU ARE NOT THE INTENDED RECIPIENT, PLEASE NOTIFY US IMMEDIATELY BY REPLY E-MAIL AND DELETE THE ORIGINAL MESSAGE. ########################################################## -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060920/d714c09b/attachment.html From millallo at gmail.com Thu Sep 21 01:30:04 2006 From: millallo at gmail.com (Emiliano Parasassi) Date: Thu, 21 Sep 2006 10:30:04 +0200 Subject: [Live-devel] ~H264VideoRTPSink bug In-Reply-To: References: <45056EBB.6000407@gmail.com> <9585da7b0609200731y7a27cbe1x5e8a71da524da6ae@mail.gmail.com> Message-ID: <45124D8C.8090504@gmail.com> Ross Finlayson wrote: > Oops - possibly not. > > Try changing line 53 of H264VideoRTPSink.cpp from > delete fOurFragmenter; > to > Medium::close(fOurFragmenter); > I have mistaken, the code is right with fSource = NULL; Sorry Emiliano From neel.palrecha at gmail.com Thu Sep 21 07:23:00 2006 From: neel.palrecha at gmail.com (Neel Palrecha) Date: Thu, 21 Sep 2006 15:23:00 +0100 Subject: [Live-devel] Problem streaming AAC in Windows Message-ID: Hello, I have compiled the live555 libraries along with the testOnDemandRTSPServer test application in windows. When attempting to stream an AAC file, the applications crashes (upon the DESCRIBE command). I've been able to isolate it to sdpDescription = session->generateSDPDescription; in RTSPServer.cpp see: void RTSPServer::RTSPClientSession ::handleCmd_DESCRIBE(char const* cseq, char const* urlSuffix, char const* fullRequestStr). I have no problems streaming MP3 and WAV. Has anyone seen this problem, or have any fixes? Neel Palrecha neel.palrecha at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060921/e8e95cc8/attachment.html From alexr at vigilanttechnology.com Thu Sep 21 08:24:56 2006 From: alexr at vigilanttechnology.com (Alex Rier) Date: Thu, 21 Sep 2006 17:24:56 +0200 Subject: [Live-devel] ~H264VideoRTPSink bug - SOLUTION Message-ID: <683BC86C0162454BAC43B789A7FB1964393BC6@herlios.adyoron.com> Hi, Bug Description: RTP package causes Segmentation Fault at ~H264VideoRTPSink invokation. The fault is caused when trying to call doStopGettingFrames() from FramedSource::stopGettingFrames() . Reason: FramedSource object was destroyed before calling that method. Solution: // According to Ross recommendation, // replace "delete fOurFramgmenter;" in ~H264VideoRTPSink() with Medium::close (fOurFragmenter); fOurFragmenter = NULL; // Added to Ross recommendation and solved the bug: fSource = NULL; Regards, Alex This mail passed through VIGILANT TECHNOLOGY Mail-SeCure. ************************************************************************************ This footnote confirms that this email message has been scanned by PineApp Mail-SeCure for the presence of malicious code, vandals & computer viruses. ************************************************************************************ From tranzmatt at gmail.com Thu Sep 21 07:39:42 2006 From: tranzmatt at gmail.com (Tranzmatt) Date: Thu, 21 Sep 2006 10:39:42 -0400 Subject: [Live-devel] Looking for known working m4v file Message-ID: <423100740609210739h6a401ea1kf03041bf7090b350@mail.gmail.com> > Unfortunately, Apple coopted (misused) the '.m4v' filename extension > to mean "MPEG-4 video contained within a MPEG-4 format file", rather > than "a MPEG-4 Video Elementary Stream file", which is how everyone > else had been using the extension before. Well, that would explain it. So liveMedia can't handle the mpeg4 container/file format? The reason I ask is that I've successfully built liveMedia on a PPC board running VxWorks 5.5 and wanted to use it to stream some processed video data off to a media client like vlc. The mp3/mpeg1or2 tests work fine, but I wanted to use mpeg4 because of it's apparent ability to contain "tracks" of various types, plus I really want to stream using only an intra-frame lossless codec, so the whole DCT stuff in mpeg1/2 doesn't work for me. Am I out of luck in trying to use liveMedia? Matt From finlayson at live555.com Thu Sep 21 10:13:18 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Sep 2006 10:13:18 -0700 Subject: [Live-devel] Looking for known working m4v file In-Reply-To: <423100740609210739h6a401ea1kf03041bf7090b350@mail.gmail.com> References: <423100740609210739h6a401ea1kf03041bf7090b350@mail.gmail.com> Message-ID: > > Unfortunately, Apple coopted (misused) the '.m4v' filename extension >> to mean "MPEG-4 video contained within a MPEG-4 format file", rather >> than "a MPEG-4 Video Elementary Stream file", which is how everyone >> else had been using the extension before. > >Well, that would explain it. So liveMedia can't handle the mpeg4 >container/file format? There is currently no code in the "LIVE555 Streaming Media" libraries for demultiplexing and reading from MPEG-4 format files. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Sep 21 10:15:35 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Sep 2006 10:15:35 -0700 Subject: [Live-devel] ~H264VideoRTPSink bug - SOLUTION In-Reply-To: <683BC86C0162454BAC43B789A7FB1964393BC6@herlios.adyoron.com> References: <683BC86C0162454BAC43B789A7FB1964393BC6@herlios.adyoron.com> Message-ID: >Bug Description: > > RTP package causes Segmentation Fault at ~H264VideoRTPSink invokation. > The fault is caused when trying to call doStopGettingFrames() from > FramedSource::stopGettingFrames() . This is fixed in the latest version (2006.09.20) of the "LIVE555 Streaming Media" code. Please upgrade. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From xcsmith at rockwellcollins.com Thu Sep 21 11:43:47 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Thu, 21 Sep 2006 13:43:47 -0500 Subject: [Live-devel] Packet Size Message-ID: Hi LIVE! I am trying to use LIVE streaming server on a device that cannot send or receive network packets larger than 576 bytes. Any packets larger than this are dropped. I think this leaves 536 bytes for data after TCP header and 548 bytes after removing UDP header. Is there a way for me to limit the size of the packets LIVE makes? Thx! ~Medra From finlayson at live555.com Thu Sep 21 13:42:43 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Sep 2006 13:42:43 -0700 Subject: [Live-devel] Packet Size In-Reply-To: References: Message-ID: >Hi LIVE! > >I am trying to use LIVE streaming server on a device that cannot send or >receive network packets larger than 576 bytes. Any packets larger than >this are dropped. I think this leaves 536 bytes for data after TCP header >and 548 bytes after removing UDP header. Is there a way for me to limit >the size of the packets LIVE makes? Yes. After creating each "RTPSink" (actually, "MultiFramedRTPSink") object, call setPacketSizes(preferredPacketSize, maxPacketSize); on it. In your case, maxPacketSize would be 548. I suggest a value of 500 for "preferredPacketSize". (Both values should be multiples of 4, to avoid potential problems when streaming PCM audio.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From freegnu at 163.com Thu Sep 21 20:35:49 2006 From: freegnu at 163.com (freegnu) Date: Fri, 22 Sep 2006 11:35:49 +0800 Subject: [Live-devel] are there any reference about some members' initialization References: <45056EBB.6000407@gmail.com> <9585da7b0609200731y7a27cbe1x5e8a71da524da6ae@mail.gmail.com> <45124D8C.8090504@gmail.com> Message-ID: <000b01c6ddf8$3331fde0$0501a8c0@zwenqiang> hi,all, i want to implement a class called MPEG4ProgramSteamFileServerMediaSubsession to support mpeg4 stream, but when i read the code of MPEG1or2Demux.cpp i find there are many variables and functions are not open source and dont give the function prototype, and i dont how to use them. is any one give me and advice for how to use the functions or variables? for example: MPEG1or2Demux ::MPEG1or2Demux(UsageEnvironment& env, FramedSource* inputSource, Boolean reclaimWhenLastESDies) : Medium(env), fInputSource(inputSource), fMPEGversion(0), fNextAudioStreamNumber(0), fNextVideoStreamNumber(0), fReclaimWhenLastESDies(reclaimWhenLastESDies), fNumOutstandingESs(0), fNumPendingReads(0), fHaveUndeliveredData(False) { fParser = new MPEGProgramStreamParser(this, inputSource); for (unsigned i = 0; i < 256; ++i) { fOutput[i].savedDataHead = fOutput[i].savedDataTail = NULL; fOutput[i].isPotentiallyReadable = False; fOutput[i].isCurrentlyActive = False; fOutput[i].isCurrentlyAwaitingData = False; } } //how can i know how to use fInputSource,fMPEGversion...... From ambuj.sinha at iptvlabs.biz Thu Sep 21 22:45:41 2006 From: ambuj.sinha at iptvlabs.biz (Ambuj Sinha) Date: Fri, 22 Sep 2006 11:15:41 +0530 Subject: [Live-devel] Ref: testOnDemandRTSPServer Message-ID: <200609220545.k8M5jmKQ036966@ns.live555.com> Hi, I want to create a new RTSP Server using testOnDemandRTSPServer which can stream MPEG-TS and can communicate with Amino STBs how I should start building my new RTSP Server with Trick Play functionalities (FF, REW, Pause, etc.) added to it. Please do let me know the steps to follow. Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. Email: ambuj.sinha at iptvlabs.biz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060921/7b77ed79/attachment.html From ambuj.sinha at iptvlabs.biz Fri Sep 22 03:15:12 2006 From: ambuj.sinha at iptvlabs.biz (Ambuj Sinha) Date: Fri, 22 Sep 2006 15:45:12 +0530 Subject: [Live-devel] Ref: testOnDemandRTSPServer Message-ID: <200609221015.k8MAFI1N000242@ns.live555.com> Hi I have installed Latest Live555 library on the Suse Linux 10.0. However, I have a problem which goes like this. I have a MPEG-PS file (test.mpg). I convert it into a TS file (test.ts) using the testMPEG1or2ProgramToTransportStream, following which I use testOnDemandRTSPServer to stream the TS file (test.ts). In the code for testOnDemandRTSPServer.cpp, I have made changes to streamName and inputFileName (in the portion which deals with MPEG-TS streams). After compiling and running I get the RTSP URL and can see my file using VLC. However, I am not able to Pause the stream. It Plays the file for the full duration and then sends a Teardown Message after it is finished. Please suggest where I am wrong. Is Pause functionality available on MPEG-TS streams? Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060922/ff98d3f5/attachment.html From nitin.e at gmail.com Fri Sep 22 04:40:46 2006 From: nitin.e at gmail.com (nitin jain) Date: Fri, 22 Sep 2006 17:10:46 +0530 Subject: [Live-devel] live-devel Digest, Vol 35, Issue 19 In-Reply-To: References: Message-ID: Hi all, I am able to stream and receive ES,PS, TS MPEG2 stream using live555 Testmpeg1or2Videostreamer.cpp and Testmpeg1or2VideoReceiver.cpp demo applications programs. But when I analyze the received PS stream at client side using MPEG stream analyzer( MANZITA ) it is showing Transmitted PS stream from server as ES stream at client side. Do the LIVE server is converting PS stream to ES stream before transmitting ? Any clues on this. Regards Nitin On 9/22/06, live-devel-request at ns.live555.com < live-devel-request at ns.live555.com> wrote: > > Send live-devel mailing list submissions to > live-devel at lists.live555.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.live555.com/mailman/listinfo/live-devel > or, via email, send a message with subject or body 'help' to > live-devel-request at lists.live555.com > > You can reach the person managing the list at > live-devel-owner at lists.live555.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of live-devel digest..." > > > Today's Topics: > > 1. Reading from File (Sachin S N) > 2. Re: ~H264VideoRTPSink bug (Emiliano Parasassi) > 3. Problem streaming AAC in Windows (Neel Palrecha) > 4. Re: ~H264VideoRTPSink bug - SOLUTION (Alex Rier) > 5. Re: Looking for known working m4v file (Tranzmatt) > 6. Re: Looking for known working m4v file (Ross Finlayson) > 7. Re: ~H264VideoRTPSink bug - SOLUTION (Ross Finlayson) > 8. Packet Size (xcsmith at rockwellcollins.com) > 9. Re: Packet Size (Ross Finlayson) > 10. are there any reference about some members' initialization > (freegnu) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 21 Sep 2006 10:19:43 +0530 > From: "Sachin S N" > Subject: [Live-devel] Reading from File > To: > Message-ID: > <607194575BD8CE46BCC8D747996FA7F901E167F7 at SI-RD10-MS01.LGE.NET> > Content-Type: text/plain; charset="us-ascii" > > Hi all, > > > > Im new to the this fourm . > > Im trying to write a parser for H 264, so I need info on how to read > from the file or buffer in Live555 stack. > > > > I came to know that we can use functions like > test4Bytes(),get4Bytes(),getByte(), SkipBytes() > > defined in StreamParser.hh can be used for the file operations. > > > > Are there any other methods to read from file or buffer? > > > > How can we find that EOF is reached while reading from file or buffer? > > > > Thanks in advance. > > > > Regards > > Sachin > > ######################################################### > THIS EMAIL MESSAGE IS FOR THE SOLE USE OF THE INTENDED > RECIPIENT(S) AND MAY CONTAIN CONFIDENTIAL AND PRIVILEGED > INFORMATION. ANY UNAUTHORIZED REVIEW, USE, DISCLOSURE OR > DISTRIBUTION IS PROHIBITED.BEFORE OPENING ANY ATTACHMENTS > PLEASE CHECK FOR VIRUSES AND DEFECTS.IF YOU ARE NOT THE > INTENDED RECIPIENT, PLEASE NOTIFY US IMMEDIATELY BY REPLY > E-MAIL AND DELETE THE ORIGINAL MESSAGE. > ########################################################## > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.live555.com/pipermail/live-devel/attachments/20060920/d714c09b/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 21 Sep 2006 10:30:04 +0200 > From: Emiliano Parasassi > Subject: Re: [Live-devel] ~H264VideoRTPSink bug > To: LIVE555 Streaming Media - development & use > > Message-ID: <45124D8C.8090504 at gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Ross Finlayson wrote: > > Oops - possibly not. > > > > Try changing line 53 of H264VideoRTPSink.cpp from > > delete fOurFragmenter; > > to > > Medium::close(fOurFragmenter); > > > I have mistaken, the code is right with > fSource = NULL; > > Sorry > Emiliano > > > ------------------------------ > > Message: 3 > Date: Thu, 21 Sep 2006 15:23:00 +0100 > From: "Neel Palrecha" > Subject: [Live-devel] Problem streaming AAC in Windows > To: live-devel at ns.live555.com > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > I have compiled the live555 libraries along with the > testOnDemandRTSPServer test application in windows. When attempting to > stream an AAC file, the applications crashes (upon the DESCRIBE command). > > I've been able to isolate it to > sdpDescription = session->generateSDPDescription; > > in RTSPServer.cpp see: > > void RTSPServer::RTSPClientSession > ::handleCmd_DESCRIBE(char const* cseq, char const* urlSuffix, char const* > fullRequestStr). > > I have no problems streaming MP3 and WAV. Has anyone seen this problem, > or > have any fixes? > > Neel Palrecha > neel.palrecha at gmail.com > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.live555.com/pipermail/live-devel/attachments/20060921/e8e95cc8/attachment-0001.html > > ------------------------------ > > Message: 4 > Date: Thu, 21 Sep 2006 17:24:56 +0200 > From: "Alex Rier" > Subject: Re: [Live-devel] ~H264VideoRTPSink bug - SOLUTION > To: > Message-ID: > <683BC86C0162454BAC43B789A7FB1964393BC6 at herlios.adyoron.com> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > Bug Description: > > RTP package causes Segmentation Fault at ~H264VideoRTPSink invokation. > The fault is caused when trying to call doStopGettingFrames() from > FramedSource::stopGettingFrames() . > > Reason: > > FramedSource object was destroyed before calling that method. > > Solution: > > // According to Ross recommendation, > // replace "delete fOurFramgmenter;" in ~H264VideoRTPSink() with > Medium::close (fOurFragmenter); > fOurFragmenter = NULL; > > // Added to Ross recommendation and solved the bug: > fSource = NULL; > > Regards, > Alex > > > This mail passed through VIGILANT TECHNOLOGY Mail-SeCure. > > > ************************************************************************************ > This footnote confirms that this email message has been scanned by > PineApp Mail-SeCure for the presence of malicious code, vandals & computer > viruses. > > ************************************************************************************ > > > > ------------------------------ > > Message: 5 > Date: Thu, 21 Sep 2006 10:39:42 -0400 > From: Tranzmatt > Subject: Re: [Live-devel] Looking for known working m4v file > To: live-devel at ns.live555.com > Message-ID: > <423100740609210739h6a401ea1kf03041bf7090b350 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Unfortunately, Apple coopted (misused) the '.m4v' filename extension > > to mean "MPEG-4 video contained within a MPEG-4 format file", rather > > than "a MPEG-4 Video Elementary Stream file", which is how everyone > > else had been using the extension before. > > Well, that would explain it. So liveMedia can't handle the mpeg4 > container/file format? The reason I ask is that I've successfully > built liveMedia on a PPC board running VxWorks 5.5 and wanted > to use it to stream some processed video data off to a media client > like vlc. The mp3/mpeg1or2 tests work fine, but I wanted to use > mpeg4 because of it's apparent ability to contain "tracks" of various > types, > plus I really want to stream using only an intra-frame lossless codec, > so the whole DCT stuff in mpeg1/2 doesn't work for me. Am I out > of luck in trying to use liveMedia? > > Matt > > > ------------------------------ > > Message: 6 > Date: Thu, 21 Sep 2006 10:13:18 -0700 > From: Ross Finlayson > Subject: Re: [Live-devel] Looking for known working m4v file > To: LIVE555 Streaming Media - development & use > > Message-ID: > Content-Type: text/plain; charset="us-ascii" ; format="flowed" > > > > Unfortunately, Apple coopted (misused) the '.m4v' filename extension > >> to mean "MPEG-4 video contained within a MPEG-4 format file", rather > >> than "a MPEG-4 Video Elementary Stream file", which is how everyone > >> else had been using the extension before. > > > >Well, that would explain it. So liveMedia can't handle the mpeg4 > >container/file format? > > There is currently no code in the "LIVE555 Streaming Media" libraries > for demultiplexing and reading from MPEG-4 format files. > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ > > > ------------------------------ > > Message: 7 > Date: Thu, 21 Sep 2006 10:15:35 -0700 > From: Ross Finlayson > Subject: Re: [Live-devel] ~H264VideoRTPSink bug - SOLUTION > To: LIVE555 Streaming Media - development & use > > Message-ID: > Content-Type: text/plain; charset="us-ascii" ; format="flowed" > > >Bug Description: > > > > RTP package causes Segmentation Fault at ~H264VideoRTPSink invokation. > > The fault is caused when trying to call doStopGettingFrames() from > > FramedSource::stopGettingFrames() . > > This is fixed in the latest version (2006.09.20) of the "LIVE555 > Streaming Media" code. Please upgrade. > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ > > > ------------------------------ > > Message: 8 > Date: Thu, 21 Sep 2006 13:43:47 -0500 > From: > Subject: [Live-devel] Packet Size > To: live-devel at ns.live555.com > Message-ID: > < > OF2D278F16.47F41295-ON862571F0.00668481-862571F0.0066F883 at rockwellcollins.com > > > > Content-Type: text/plain; charset=US-ASCII > > > Hi LIVE! > > I am trying to use LIVE streaming server on a device that cannot send or > receive network packets larger than 576 bytes. Any packets larger than > this are dropped. I think this leaves 536 bytes for data after TCP header > and 548 bytes after removing UDP header. Is there a way for me to limit > the size of the packets LIVE makes? > > Thx! > ~Medra > > > > ------------------------------ > > Message: 9 > Date: Thu, 21 Sep 2006 13:42:43 -0700 > From: Ross Finlayson > Subject: Re: [Live-devel] Packet Size > To: LIVE555 Streaming Media - development & use > > Message-ID: > Content-Type: text/plain; charset="us-ascii" ; format="flowed" > > >Hi LIVE! > > > >I am trying to use LIVE streaming server on a device that cannot send or > >receive network packets larger than 576 bytes. Any packets larger than > >this are dropped. I think this leaves 536 bytes for data after TCP header > >and 548 bytes after removing UDP header. Is there a way for me to limit > >the size of the packets LIVE makes? > > Yes. After creating each "RTPSink" (actually, "MultiFramedRTPSink") > object, call > setPacketSizes(preferredPacketSize, maxPacketSize); > on it. > > In your case, maxPacketSize would be 548. I suggest a value of 500 > for "preferredPacketSize". (Both values should be multiples of 4, to > avoid potential problems when streaming PCM audio.) > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ > > > ------------------------------ > > Message: 10 > Date: Fri, 22 Sep 2006 11:35:49 +0800 > From: "freegnu" > Subject: [Live-devel] are there any reference about some members' > initialization > To: "LIVE555 Streaming Media - development & use" > > Message-ID: <000b01c6ddf8$3331fde0$0501a8c0 at zwenqiang> > Content-Type: text/plain; charset="ISO-8859-1" > > hi,all, i want to implement a class called > MPEG4ProgramSteamFileServerMediaSubsession > to support mpeg4 stream, but when i read the code of MPEG1or2Demux.cpp i > find there are many variables and functions are not open source and dont > give the function prototype, and i dont how to use them. > is any one give me and advice for how to use the functions or variables? > > for example: > MPEG1or2Demux > ::MPEG1or2Demux(UsageEnvironment& env, > FramedSource* inputSource, Boolean reclaimWhenLastESDies) > : Medium(env), > fInputSource(inputSource), fMPEGversion(0), > fNextAudioStreamNumber(0), fNextVideoStreamNumber(0), > fReclaimWhenLastESDies(reclaimWhenLastESDies), fNumOutstandingESs(0), > fNumPendingReads(0), fHaveUndeliveredData(False) { > fParser = new MPEGProgramStreamParser(this, inputSource); > for (unsigned i = 0; i < 256; ++i) { > fOutput[i].savedDataHead = fOutput[i].savedDataTail = NULL; > fOutput[i].isPotentiallyReadable = False; > fOutput[i].isCurrentlyActive = False; > fOutput[i].isCurrentlyAwaitingData = False; > } > } > > //how can i know how to use fInputSource,fMPEGversion...... > > > > ------------------------------ > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > End of live-devel Digest, Vol 35, Issue 19 > ****************************************** > -- Nitin Jain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060922/8523800b/attachment-0001.html From xcsmith at rockwellcollins.com Fri Sep 22 08:20:11 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Fri, 22 Sep 2006 10:20:11 -0500 Subject: [Live-devel] are there any reference about some members' initialization In-Reply-To: <000b01c6ddf8$3331fde0$0501a8c0@zwenqiang> Message-ID: hi,all, i want to implement a class called MPEG4ProgramSteamFileServerMediaSubsession to support mpeg4 stream, but when i read the code of MPEG1or2Demux.cpp i find there are many variables and functions are not open source and dont give the function prototype, and i dont how to use them. is any one give me and advice for how to use the functions or variables? for example: MPEG1or2Demux ::MPEG1or2Demux(UsageEnvironment& env, FramedSource* inputSource, Boolean reclaimWhenLastESDies) : Medium(env), fInputSource(inputSource), fMPEGversion(0), fNextAudioStreamNumber(0), fNextVideoStreamNumber(0), fReclaimWhenLastESDies(reclaimWhenLastESDies), fNumOutstandingESs(0), fNumPendingReads(0), fHaveUndeliveredData(False) { fParser = new MPEGProgramStreamParser(this, inputSource); for (unsigned i = 0; i < 256; ++i) { fOutput[i].savedDataHead = fOutput[i].savedDataTail = NULL; fOutput[i].isPotentiallyReadable = False; fOutput[i].isCurrentlyActive = False; fOutput[i].isCurrentlyAwaitingData = False; } } //how can i know how to use fInputSource,fMPEGversion...... Hi, free. In the Library, you will find that the member data begins with a lowercase "f." So these things "fInputSource", "fMPEGversion" are member data for that class. If you can't find them in the class you are looking at, try checking if they were inherited from a parent class. The parent class of MPEG1or2Demux should be OnDemandServerMediaSession, who's parent in turn is ServerMediaSubsession. The piece of code you pointed out that looks like it's calling functions is a constructor initialization list. ~Medra From finlayson at live555.com Fri Sep 22 08:30:32 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 22 Sep 2006 08:30:32 -0700 Subject: [Live-devel] live-devel Digest, Vol 35, Issue 19 In-Reply-To: References: Message-ID: > Do the LIVE server is converting PS stream to ES stream before >transmitting ? Yes. This should be very clear if you look at the effect of defining SOURCE_IS_PROGRAM_STREAM in "testMPEG1or2VideoStreamer.cpp" -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From xcsmith at rockwellcollins.com Fri Sep 22 08:34:08 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Fri, 22 Sep 2006 10:34:08 -0500 Subject: [Live-devel] Conversion to ES (nitin) In-Reply-To: Message-ID: But when I analyze the received PS stream at client side using MPEG stream analyzer( MANZITA ) it is showing Transmitted PS stream from server as ES stream at client side. Do the LIVE server is converting PS stream to ES stream before transmitting ? Yes. A PS is sent by demultiplexing it into individual streams and sending them separately. A PS is not designed to be delivered over a network with possible "non-negligible" transport errors. ~Medra From xcsmith at rockwellcollins.com Fri Sep 22 08:46:48 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Fri, 22 Sep 2006 10:46:48 -0500 Subject: [Live-devel] Ref: testOnDemandRTSPServer In-Reply-To: <200609221015.k8MAFI1N000242@ns.live555.com> Message-ID: In the code for testOnDemandRTSPServer.cpp, I have made changes to streamName and inputFileName (in the portion which deals with MPEG-TS streams). After compiling and running I get the RTSP URL and can see my file using VLC. However, I am not able to Pause the stream. It Plays the file for the full duration and then sends a Teardown Message after it is finished. Please suggest where I am wrong. Is Pause functionality available on MPEG-TS streams? Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. I'm not totally sure, but I'd like to suggest that the reason VLC does not even send a pause message is that the SDP for the TS that the LIVE server sends has no playtime showing. Note the line in SDP: "a=range:npt=0-" VLC possibly assumes the TS file is a live feed because there is no max time. (When the SDP is sent for a PS demux, LIVE fills in the max time.) As it is not very logical to pause a live feed, VLC would have the option disabled. I see the same behaviour for TS from my VLC also. As to whether or not PAUSE is allowed on TS, you'll have to wait for Ross or someone else to answer. ~Medra From rahul.ray at gmail.com Fri Sep 22 10:48:19 2006 From: rahul.ray at gmail.com (Rahul Ray) Date: Fri, 22 Sep 2006 10:48:19 -0700 Subject: [Live-devel] openRTSP not working with -c Message-ID: Hi, I need to run openRTSP in a loop continuously. I ran it like: openRTSP -i -w 704 -h 480 -f 15 -e 5 -c -b 80000 rtsp://11.210.18.18/mpeg4?q=15&cam=0/ It says: Failed to start playing session: PLAY: cannot handle response: RTSP/1.0 455 Method Not Valid In This State Any idea how to fix this ? The full log is: [ipicsspm2 182] ~/live/testProgs > openRTSP -i -w 704 -h 480 -f 15 -e 5 -c -b 80000 rtsp://64.210.18.18/mpeg4?q=15&cam=0/ [1] 7407 Opened URL "[10:44 - 1.11] [ipicsspm2 183] ~/live/testProgs > rtsp://64.210.18.18/mpeg4?q=15", returning a SDP description: v=0 o=MangoDSP 9487543690017569303 9487543690017569303 IN IP4 64.210.18.18 s=Mango DSP Audio/Video c=IN IP4 0.0.0.0 m=video 0 RTP/AVP 96 a=rtpmap:96 MP4V-ES/90000 a=fmtp:96 profile-level-id=1;config=000001B003000001B50900000101000001200086E00024E22003E80068824800F628B021E0A31F000001B3001007 a=control:__StreamID=2212056448 Created receiver for "video/MP4V-ES" subsession (client ports 32812-32813) Setup "video/MP4V-ES" subsession (client ports 32812-32813) Started playing session Receiving streamed data (for up to 5.000000 seconds)... Failed to start playing session: PLAY: cannot handle response: RTSP/1.0 455 Method Not Valid In This State ...........................................................................[1]+ Exit 1 openRTSP -i -w 704 -h 480 -f 15 -e 5 -c -b 80000 rtsp://64.210.18.18/mpeg4?q=15 From xcsmith at rockwellcollins.com Fri Sep 22 09:16:32 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Fri, 22 Sep 2006 11:16:32 -0500 Subject: [Live-devel] are there any reference about some members' initialization In-Reply-To: Message-ID: The parent class of MPEG1or2Demux should be OnDemandServerMediaSession, who's parent in turn is ServerMediaSubsession. ~Medra Sorry, I've made a mistake/typo. I thought you were looking at MPEG1or2DemuxedFileServerMediaSubsession (who's parent is OnDemandServerMediaSubsession, not OnDemandServerMediaSession). The parent of MPEG1or2Deumux is "Medium." From finlayson at live555.com Fri Sep 22 12:43:30 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 22 Sep 2006 12:43:30 -0700 Subject: [Live-devel] Ref: testOnDemandRTSPServer In-Reply-To: References: Message-ID: >I'm not totally sure, but I'd like to suggest that the reason VLC does not >even send a pause message is that the SDP for the TS that the LIVE server >sends has no playtime showing. >Note the line in SDP: "a=range:npt=0-" VLC possibly assumes the TS file >is a live feed because there is no max time. (When the SDP is sent for a >PS demux, LIVE fills in the max time.) As it is not very logical to pause >a live feed, VLC would have the option disabled. This is correct. Our RTSP server implementation currently does not calculate a max time for a Transport Stream file, and so does not report one in the SDP description. VLC therefore does not implement the RTSP "PAUSE" command on the stream. (In contrast, you'll notice that for some other file types - e.g., WAV - a max time is reported, and pausing in VLC works OK.) This will be fixed around the end of the year when 'trick play' support (seek, ff, rw) is added for Transport Stream files. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Fri Sep 22 14:44:50 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 22 Sep 2006 14:44:50 -0700 Subject: [Live-devel] are there any reference about some members' initialization In-Reply-To: References: Message-ID: >The parent class of MPEG1or2Demux should be OnDemandServerMediaSession, >who's parent >in turn is ServerMediaSubsession. > >~Medra > >Sorry, I've made a mistake/typo. I thought you were looking at >MPEG1or2DemuxedFileServerMediaSubsession (who's parent is >OnDemandServerMediaSubsession, not OnDemandServerMediaSession). The parent >of MPEG1or2Deumux is "Medium." Note that questions like these are easily answered by browsing the code's 'Doxygen' online documentation: http://www.live555.com/liveMedia/doxygen/html/classes.html -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060922/5df699e9/attachment.html From finlayson at live555.com Fri Sep 22 17:10:30 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 22 Sep 2006 17:10:30 -0700 Subject: [Live-devel] openRTSP not working with -c In-Reply-To: References: Message-ID: >Hi, > I need to run openRTSP in a loop continuously. > > I ran it like: > > openRTSP -i -w 704 -h 480 -f 15 -e 5 -c -b 80000 >rtsp://11.210.18.18/mpeg4?q=15&cam=0/ > > It says: > > Failed to start playing session: PLAY: cannot handle response: >RTSP/1.0 455 Method Not Valid In This State > > Any idea how to fix this ? Use a different server that doesn't return this (bogus, IMHO) error. (Both our RTSP server implementation, and the Darwin Streaming Server, work OK with "openRTSP -c".) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From ambuj.sinha at iptvlabs.biz Sat Sep 23 03:15:58 2006 From: ambuj.sinha at iptvlabs.biz (Ambuj Sinha) Date: Sat, 23 Sep 2006 15:45:58 +0530 Subject: [Live-devel] Ref: testOnDemandRTSPServer Message-ID: <200609231016.k8NAG4dh058033@ns.live555.com> Hi Ross Thanks for the reply. Now I want to test testOnDemandRTSPServer sending TS files to communicate with Amino STB. Could you please give me a link to download a valid MPEG-TS file for this purpose which has been tested to communicate with Amino STB and testOnDemandRTSPServer? What are the steps involved for this communication do I need to make some changes in the code also for it? Thanks Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060923/6e2999c8/attachment-0001.html From herurori at yahoo.com Sat Sep 23 23:06:50 2006 From: herurori at yahoo.com (Heru Rori) Date: Sat, 23 Sep 2006 23:06:50 -0700 (PDT) Subject: [Live-devel] Need Help Message-ID: <20060924060650.32261.qmail@web30709.mail.mud.yahoo.com> Hello, can anyone help me give a step by step instructions. i have live555 streaming media source code but i still can't configure and build it on my windows. I has following the instructions but it still can't be opened on my visual studio. I use microsoft visual studio 6 and microsoft visual studio .net. thank you very much. Heru Yuri Rori Herurori at yahoo.com 081-759-0008 --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060923/5add8b01/attachment.html From sachin.sn at lgsoftindia.com Sun Sep 24 20:36:20 2006 From: sachin.sn at lgsoftindia.com (Sachin S N) Date: Mon, 25 Sep 2006 09:06:20 +0530 Subject: [Live-devel] info needed on input buffer Message-ID: <607194575BD8CE46BCC8D747996FA7F901E685AB@SI-RD10-MS01.LGE.NET> Hi all, I need info on how to read from the file or input buffer in Live555 stack. I came to know that we can use functions like test4Bytes(),get4Bytes(),getByte(), SkipBytes() defined in StreamParser.hh. Are there any other methods to read from file or buffer? How can we find out that EOF is reached while reading from input buffer? Thanks Sachin ######################################################### THIS EMAIL MESSAGE IS FOR THE SOLE USE OF THE INTENDED RECIPIENT(S) AND MAY CONTAIN CONFIDENTIAL AND PRIVILEGED INFORMATION. ANY UNAUTHORIZED REVIEW, USE, DISCLOSURE OR DISTRIBUTION IS PROHIBITED.BEFORE OPENING ANY ATTACHMENTS PLEASE CHECK FOR VIRUSES AND DEFECTS.IF YOU ARE NOT THE INTENDED RECIPIENT, PLEASE NOTIFY US IMMEDIATELY BY REPLY E-MAIL AND DELETE THE ORIGINAL MESSAGE. ########################################################## -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060924/01b4b49e/attachment.html From finlayson at live555.com Sun Sep 24 21:17:36 2006 From: finlayson at live555.com (Ross Finlayson) Date: Sun, 24 Sep 2006 21:17:36 -0700 Subject: [Live-devel] info needed on input buffer In-Reply-To: <607194575BD8CE46BCC8D747996FA7F901E685AB@SI-RD10-MS01.LGE.NET> References: <607194575BD8CE46BCC8D747996FA7F901E685AB@SI-RD10-MS01.LGE.NET> Message-ID: > I came to know that we can use functions like >test4Bytes(),get4Bytes(),getByte(), SkipBytes() >defined in StreamParser.hh. > > Are there any other methods to read from file or buffer? "StreamParser" is used to parse data from a byte stream (which will usually be a "ByteStreamFileSource"). If, instead, you simply want to read data without parsing it, or if you want to read data in fixed-size chunks only, then you can do so by reading directly from a "ByteStreamFileSource". > > How can we find out that EOF is reached while reading from input buffer? The "onInputCloseFunc" function (passed as a parameter to the "StreamParser" constructor) will get called. (Usually, this will be the "onCloseFunc" function that's passed to "FramedSource::getNextFrame()".) Look at the code for the demo applications to see several examples of this. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060924/9d42bd1f/attachment.html From Claudio.Cavalera at siemens.com Mon Sep 25 00:21:04 2006 From: Claudio.Cavalera at siemens.com (Cavalera Claudio) Date: Mon, 25 Sep 2006 09:21:04 +0200 Subject: [Live-devel] RTSP/RTP Proxy Message-ID: Hello, I would like to build an RTSP/RTP Proxy using liveMedia. Concerning RTSP, I think that the proxy could forward RTSP commands with minor adjustments (IP Addresses, UDP Ports, Content-Length in DESCRIBE response). Concerning RTP, I think that the proxy could just forward UDP packets to the right ports without knowing that they are RTP. I've searched ML archives and found this message: http://lists.live555.com/pipermail/live-devel/2005-April/002359.html Could the proxy be a sort of merge of openRTSP and testOnDemandRTSPServer? Thanks for any advice, Claudio From ambuj.sinha at iptvlabs.biz Mon Sep 25 03:27:01 2006 From: ambuj.sinha at iptvlabs.biz (Ambuj Sinha) Date: Mon, 25 Sep 2006 15:57:01 +0530 Subject: [Live-devel] Ref: MPEG -TS file Message-ID: <200609251027.k8PARBNJ096878@ns.live555.com> Hi I want to calculate the Transport Bit rate/Transfer Rate for MPEG-TS file stored on the disc so that I can calculate the total File Duration the file will be played. Which part of the live555 library should I refer to and if changes are to be made which specific files should I work on? Thanks Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060925/cc102bcb/attachment.html From rajeshkumar.r at imimobile.com Mon Sep 25 05:14:52 2006 From: rajeshkumar.r at imimobile.com (Rajesh Kumar Raju) Date: Mon, 25 Sep 2006 17:44:52 +0530 Subject: [Live-devel] unable to recv the streaming Data Message-ID: Hi All I am trying to make RTSP client. After issuing play commands How to recv the streaming Data from server. when I am issuing the PLAY Command,the Response is OK. But I am unable to recv the Streaming Data. plz tell me How to do this with regards rajesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060925/ca59fd10/attachment.html From wei.francois at gmail.com Mon Sep 25 06:06:06 2006 From: wei.francois at gmail.com (=?GB2312?B?s8K2q86w?=) Date: Mon, 25 Sep 2006 21:06:06 +0800 Subject: [Live-devel] Fwd: Question about port Live555 to uClinux In-Reply-To: <503760b30609250603h1ed8105bxed0c25a507f887e8@mail.gmail.com> References: <503760b30609250603h1ed8105bxed0c25a507f887e8@mail.gmail.com> Message-ID: <503760b30609250606o16935042g80dac74adaa9cb80@mail.gmail.com> ---------- Forwarded message ---------- From: ??? Date: 2006-9-25 ??9:03 Subject: Question about port Live555 to uClinux To: live-devel at lists.live555.com Hello, everyone Now I have a target board (named TMS320DM270 with ARM and DSP), and I have ported the uClinux 2.4.19-uc1 on ARM. Then I want to port the Live555 to the target board in order to receive and send streaming video to the LAN/WAN. My question is how to cross-compile the Live555 and then port it to uClinux on board? Following is my original methods: 1) Compile the Live555 on host Linux(RH9) with #./genMakefile uClinux #make 2) copy the bin file to uClinux filesystem 3) then download the kernel and filesystem to the board 4) run the Live555's bin files on board I am looking forward to your reply. Any suggestions will be appreciated! Thank you very much! -- Regards, francois_wei -- Regards, francois_wei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060925/128a9f0b/attachment-0001.html From wei.francois at gmail.com Mon Sep 25 06:03:08 2006 From: wei.francois at gmail.com (=?GB2312?B?s8K2q86w?=) Date: Mon, 25 Sep 2006 21:03:08 +0800 Subject: [Live-devel] Question about port Live555 to uClinux Message-ID: <503760b30609250603h1ed8105bxed0c25a507f887e8@mail.gmail.com> Hello, everyone Now I have a target board (named TMS320DM270 with ARM and DSP), and I have ported the uClinux 2.4.19-uc1 on ARM. Then I want to port the Live555 to the target board in order to receive and send streaming video to the LAN/WAN. My question is how to cross-compile the Live555 and then port it to uClinux on board? Following is my original methods: 1) Compile the Live555 on host Linux(RH9) with #./genMakefile uClinux #make 2) copy the bin file to uClinux filesystem 3) then download the kernel and filesystem to the board 4) run the Live555's bin files on board I am looking forward to your reply. Any suggestions will be appreciated! Thank you very much! -- Regards, francois_wei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060925/0d2fee3f/attachment.html From dsusa at cestel.es Mon Sep 25 07:04:01 2006 From: dsusa at cestel.es (Diego Susa) Date: Mon, 25 Sep 2006 16:04:01 +0200 Subject: [Live-devel] streaming 'redirection' Message-ID: <4517E1D1.50106@cestel.es> Hello, & congrats for such a good library ! I'm evaluating liveMedia library in order to use it in the followig situation: I need to control a media-server from my application, and putting an RTSP client inside it, looks like a good, standard and easy choice (the "easy" part thanks to your RTSPClient class). But my app is "3rd party call control" one, so I need kind a "third party media control"; i.e. what I need is the media server to deliver audio/video streams to a DIFERENT host than the one which is asking those streams as an RTSP client. I've browsed liveMedia sources and found the following symbol: RTSP_ALLOW_CLIENT_DESTINATION_SETTING with this clear explanation from Ross http://lists.live555.com/pipermail/live-devel/2004-February/000324.html This makes pretty clear I have to recompile "libliveMedia.lib" with this flag enabled, in order to achieve my goal. Obviously, I also have to re-link it in videoLAN (this is the Media Server I'm using). Here's where the question comes in: do I have to change RTSPClient's class behavior too ? I mean: do I have to change it to include the "destination" parameter into the "Transport:" line ? ... maybe I could introduce the destination IP in the subsession object, and add it as "destination" with subsession.connectionEndpointAddress(), somewhere inside the block commented "normal RTP streaming" (this happens to be RTSPClient/line=806 in my copy of your sources) NOTICE: I should clarify I don't mind the potential D-O-S attacks, they're are not an issue for me at this stage :-) Thanks in advance for any help, tip, or hint whatsoever... ! Diego From shaswata at alumnux.com Mon Sep 25 08:57:40 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Mon, 25 Sep 2006 21:27:40 +0530 Subject: [Live-devel] unable to recv the streaming Data References: Message-ID: <002d01c6e0bb$5465ed30$2e0aa8c0@NITU> Have you opened the UDP sockets in the client side (at the port configured by the client in SETUP request). Though opening UDP is the most usual one, but if you want to get the media stream over TCP, then the media will come along the same TCP connection opened by the client to send RTSP requests. Instead of port, client must send channel configuration information for RTP interleaved over TCP. ----- Original Message ----- From: Rajesh Kumar Raju To: live-devel at ns.live555.com Sent: Monday, September 25, 2006 5:44 PM Subject: [Live-devel] unable to recv the streaming Data Hi All I am trying to make RTSP client. After issuing play commands How to recv the streaming Data from server. when I am issuing the PLAY Command,the Response is OK. But I am unable to recv the Streaming Data. plz tell me How to do this with regards rajesh ------------------------------------------------------------------------------ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060925/7fbca9e4/attachment.html From xcsmith at rockwellcollins.com Mon Sep 25 08:57:49 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Mon, 25 Sep 2006 10:57:49 -0500 Subject: [Live-devel] LIVE logo Message-ID: What are the rules for using the LIVE logo on a website? I have a company internal site that describes my project, and I wanted to add the LIVE logo to the site since we are using LIVE for media streaming. ~Medra From shaswata at alumnux.com Mon Sep 25 09:21:21 2006 From: shaswata at alumnux.com (Shaswata Jash) Date: Mon, 25 Sep 2006 21:51:21 +0530 Subject: [Live-devel] RTSP/RTP Proxy References: Message-ID: <006101c6e0be$a34c6a20$2e0aa8c0@NITU> Hi Claudio, Essentially the Proxy has two sides, one side which directly communicates with the clients and other side communicates with the real RTSP/RTP server. So for the side which is handling the clients' RTSP request, it must act as testOnDemandRTSPServer. So you have to probably change the code within RTSPServer class. Just check those functions which handles DESCRIBE, SETUP, PLAY etc. In these functions, now you have to first parse the client's RTSP request and then further you need to forward the request to a real RTSP Server modifying some of the parameters according to your requirement. So you have to use implementation of openRTSP here. After getting the response back, the proxy will further reply back to the original requesting clients. It should be noted that multiple instance of openRTSP will be created. Now about forwarding the RTP packets - here each of those openRTSP client must open proper RTPSource. For your requirement, you can think of modifying the SimpleRTPSource. The above is absolutely my own understanding. It may not be very appropriate. It will be better if Ross makes further clarification. With regards, Shaswata Jash ----- Original Message ----- From: "Cavalera Claudio" To: "LIVE555 Streaming Media - development & use" Sent: Monday, September 25, 2006 12:51 PM Subject: [Live-devel] RTSP/RTP Proxy > Hello, > I would like to build an RTSP/RTP Proxy using liveMedia. > > Concerning RTSP, I think that the proxy could forward RTSP commands with > minor adjustments (IP Addresses, UDP Ports, Content-Length in DESCRIBE > response). > Concerning RTP, I think that the proxy could just forward UDP packets to > the right ports without knowing that they are RTP. > > I've searched ML archives and found this message: > http://lists.live555.com/pipermail/live-devel/2005-April/002359.html > > Could the proxy be a sort of merge of openRTSP and > testOnDemandRTSPServer? > > Thanks for any advice, > Claudio > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > From tranzmatt at gmail.com Mon Sep 25 09:59:46 2006 From: tranzmatt at gmail.com (Tranzmatt) Date: Mon, 25 Sep 2006 12:59:46 -0400 Subject: [Live-devel] Looking for known working m4v file Message-ID: <423100740609250959t3f356be8rcaf5d7664dbcec3d@mail.gmail.com> > There is currently no code in the "LIVE555 Streaming > Media" libraries for demultiplexing and reading from > MPEG-4 format files. This probably isn't the right place to ask, but what I'm really after is an infrastructure that will let me take a video "frame" (ideally off of a DSP board, but let's KISS it and say a grayscale oscillating 0-255) and serve it up in a format that's readable by vlc or similar media client. The container/file format doesn't really matter, only that there's no inter-frame encoding. (This runs inside my lab at 1000T, so bandwidth isn't an issue.) If Live isn't the way to go, is there something else that'll fit the bill? The simpler the better. Matt From xcsmith at rockwellcollins.com Mon Sep 25 11:00:37 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Mon, 25 Sep 2006 13:00:37 -0500 Subject: [Live-devel] MediaSession cannot be inherited from Message-ID: Hi LIVE I notice the classes "MediaSession" and "MediaSubsession" have private constructors and private, non-virtual destructors. Is there a reason this class cannot be inherited from? ~Medra From rahul.ray at gmail.com Mon Sep 25 17:20:55 2006 From: rahul.ray at gmail.com (Rahul Ray) Date: Mon, 25 Sep 2006 17:20:55 -0700 Subject: [Live-devel] openRTSP -m .... Message-ID: I have a trivial question. Could you please tell me where exactly this frames are created -m option for openRTSP ? The name of function is enough for me. I am not familiar with debugging on linux. That is why ... Thanks for your help. -best, Rahul From ymreddy at ssdi.sharp.co.in Tue Sep 26 00:22:53 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Tue, 26 Sep 2006 12:52:53 +0530 Subject: [Live-devel] Server and Client on the same machine Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CFC9@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Everybody, We are using testMPEG1or2VideoStreamer.cpp and testMPEG1or2VideoReceiver.cpp for streaming and receiving the MPEG ES and PS streams on the same machine. We are running the server and client on the same machine. We observed that sometimes client will never receive any packets i.e. the function networkReadHandler() in MultiFramedRTPSource.cpp will never be called and some times, it receives the packets. Even the RTCP reports are not sent/received properly. The same functionality works fine, if the server and client is running on different machines. Any clues on this. Thanks and Regards Y. Mallikharjuna Reddy From Oussama.Layaida at inrialpes.fr Tue Sep 26 00:54:50 2006 From: Oussama.Layaida at inrialpes.fr (Oussama.Layaida at inrialpes.fr) Date: Tue, 26 Sep 2006 09:54:50 +0200 Subject: [Live-devel] detect video resolution of an MPEG4 Stream In-Reply-To: References: Message-ID: <1159257290.4518dcca8705c@listes-serv.inrialpes.fr> Hi folks, I have a newbie question. I'm using Live555 library to stream an MPEG4 video stream from a vlc server. (almost) Everything is working well: I create an RTSP client, setup the session, get data frames into memory and successfully decode them using xvid or ffmpeg (thanks to Live55 developpers for the great job). The only obstructing thing is that I have to know the video resolution in advance, especially when it is not included in the SDP (x-resolution fmtp). Is it possible to get it from the RTP or the codec specific headers ? how this can be included in the Live555 ? thanks From Xian.XU at alcatel-sbell.com.cn Tue Sep 26 01:15:34 2006 From: Xian.XU at alcatel-sbell.com.cn (MCG XU Xian) Date: Tue, 26 Sep 2006 16:15:34 +0800 Subject: [Live-devel] (no subject) Message-ID: <8C661356BADDFE49A5EAC8A1B7CDD638029D0753@asbmail1.sbell.com.cn> Hi all, Could you please share some MPEG-4 ES video files or some web site to download? I can't get it in the way of FAQ because I use http proxy to access to the internet and only 8080 port is available. Thanks for you help. Best Regards, XU Xian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060926/0fc8a1b5/attachment.html From sdhays.neon.com.tw at gmail.com Tue Sep 26 01:26:37 2006 From: sdhays.neon.com.tw at gmail.com (Scott Hays) Date: Tue, 26 Sep 2006 16:26:37 +0800 Subject: [Live-devel] detect video resolution of an MPEG4 Stream In-Reply-To: <1159257290.4518dcca8705c@listes-serv.inrialpes.fr> References: <1159257290.4518dcca8705c@listes-serv.inrialpes.fr> Message-ID: <9866ce4f0609260126ref0c86dh8f619f8fbdc55c5@mail.gmail.com> This data is included in the VOL of the MPEG4 ES. That should be included as part of the "a=fmtp...." line. It's the config part. Check out RFC 3016 for more information. You'll have to then parse that data (as specified by the MPEG4 ES standard) for the resolution. Scott On 9/26/06, Oussama.Layaida at inrialpes.fr wrote: > > > Hi folks, > > I have a newbie question. > > I'm using Live555 library to stream an MPEG4 video stream from a vlc > server. > (almost) Everything is working well: I create an RTSP client, setup the > session, get data frames into memory and successfully decode them using > xvid or > ffmpeg (thanks to Live55 developpers for the great job). > > The only obstructing thing is that I have to know the video resolution in > advance, especially when it is not included in the SDP (x-resolution > fmtp). Is > it possible to get it from the RTP or the codec specific headers ? how > this can > be included in the Live555 ? > > thanks > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060926/e811396f/attachment.html From finlayson at live555.com Tue Sep 26 02:26:51 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 02:26:51 -0700 Subject: [Live-devel] LIVE logo In-Reply-To: References: Message-ID: >What are the rules for using the LIVE logo on a website? There are currently no hard and fast rules laid out for this. > I have a company >internal site that describes my project, and I wanted to add the LIVE logo >to the site since we are using LIVE for media streaming. This should be OK, as long as there's no implication that your project is something actually developed by our company. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Tue Sep 26 02:34:10 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 02:34:10 -0700 Subject: [Live-devel] Server and Client on the same machine In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520337CFC9@ssdimailsrvnt01.ssdi.sharp.co.in > References: <7FB4685EA93D014C8E30AA087B66E7520337CFC9@ssdimailsrvnt01.ssdi.sharp.co.in > Message-ID: >Hi Everybody, > >We are using testMPEG1or2VideoStreamer.cpp and testMPEG1or2VideoReceiver.cpp >for streaming and receiving the MPEG ES and PS streams on the same machine. >We are running the server and client on the same machine. We observed that >sometimes client will never receive any packets i.e. the function >networkReadHandler() in MultiFramedRTPSource.cpp will never be called and >some times, it receives the packets. Even the RTCP reports are not >sent/received properly. > >The same functionality works fine, if the server and client is running on >different machines. > >Any clues on this. This sounds like a problem with your OS. If you're running Windows, I suggest that you upgrade to version 2006.09.18 or later of the "LIVE555 Streaming Media" code, and recompile. If that still doesn't work, then - if you're running an old version of Windows - you may need to upgrade to Windows XP for this to work. If you're not running Windows, then I don't know what might be wrong (although it's probably still an OS problem). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Tue Sep 26 02:37:31 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 02:37:31 -0700 Subject: [Live-devel] (no subject) In-Reply-To: <8C661356BADDFE49A5EAC8A1B7CDD638029D0753@asbmail1.sbell.com.cn> References: <8C661356BADDFE49A5EAC8A1B7CDD638029D0753@asbmail1.sbell.com.cn> Message-ID: >content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C6E143.F0818FB9" > >Hi all, > > Could you please share some MPEG-4 ES video files or some web >site to download? I can't get it in the way of FAQ because I use >http proxy to access to the internet and only 8080 port is available. You can try http://www.live555.com/test.m4v (I will leave this online for the next few days.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060926/acc1c631/attachment.html From finlayson at live555.com Tue Sep 26 02:40:46 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 02:40:46 -0700 Subject: [Live-devel] detect video resolution of an MPEG4 Stream In-Reply-To: <9866ce4f0609260126ref0c86dh8f619f8fbdc55c5@mail.gmail.com> References: <1159257290.4518dcca8705c@listes-serv.inrialpes.fr> <9866ce4f0609260126ref0c86dh8f619f8fbdc55c5@mail.gmail.com> Message-ID: >This data is included in the VOL of the MPEG4 ES. That should be >included as part of the "a=fmtp...." line. It's the config part. > Check out RFC 3016 for more information. You'll have to then parse >that data (as specified by the MPEG4 ES standard) for the resolution. Note that the "LIVE555 Streaming Media" libraries contain a function parseGeneralConfigStr() that will do this parsing. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Tue Sep 26 02:44:50 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 02:44:50 -0700 Subject: [Live-devel] MediaSession cannot be inherited from In-Reply-To: References: Message-ID: >Hi LIVE > >I notice the classes "MediaSession" and "MediaSubsession" have private >constructors and private, non-virtual destructors. Is there a reason this >class cannot be inherited from? Probably not - they could be "protected" instead. (I'll make this change in the next release of the software.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Tue Sep 26 02:50:56 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 02:50:56 -0700 Subject: [Live-devel] streaming 'redirection' In-Reply-To: <4517E1D1.50106@cestel.es> References: <4517E1D1.50106@cestel.es> Message-ID: >I'm evaluating liveMedia library in order to use it in the followig >situation: >I need to control a media-server from my application, and putting an >RTSP client inside it, looks like a good, standard and easy choice (the >"easy" part thanks to your RTSPClient class). >But my app is "3rd party call control" one, so I need kind a "third >party media control"; i.e. what I need is the media server to deliver >audio/video streams to a DIFERENT host than the one which is asking >those streams as an RTSP client. > >I've browsed liveMedia sources and found the following symbol: > >RTSP_ALLOW_CLIENT_DESTINATION_SETTING > >with this clear explanation from Ross >http://lists.live555.com/pipermail/live-devel/2004-February/000324.html > >This makes pretty clear I have to recompile "libliveMedia.lib" with this >flag enabled, in order to achieve my goal. Obviously, I also have to >re-link it in videoLAN (this is the Media Server I'm using). > >Here's where the question comes in: do I have to change RTSPClient's >class behavior too ? >I mean: do I have to change it to include the "destination" parameter >into the "Transport:" line ? Yes. I think that's the only change you will need to make to "RTSPClient". -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From alexr at vigilanttechnology.com Tue Sep 26 04:20:45 2006 From: alexr at vigilanttechnology.com (Alex Rier) Date: Tue, 26 Sep 2006 13:20:45 +0200 Subject: [Live-devel] SRTP (Secure RTP) Message-ID: <683BC86C0162454BAC43B789A7FB1964393CAC@herlios.adyoron.com> Hi, Is it possible to integrate Live555 RTP with SRTP (http://srtp.sourceforge.net/srtp.html)? What should be the best way to do that? Thanks, Alex This mail passed through VIGILANT TECHNOLOGY Mail-SeCure. ************************************************************************************ This footnote confirms that this email message has been scanned by PineApp Mail-SeCure for the presence of malicious code, vandals & computer viruses. ************************************************************************************ From brett at oneway.co.nz Tue Sep 26 03:47:48 2006 From: brett at oneway.co.nz (Brett Waterson) Date: Tue, 26 Sep 2006 22:47:48 +1200 Subject: [Live-devel] MediaSession cannot be inherited from In-Reply-To: Message-ID: <20060926104802.DILO11710.fep04.xtra.co.nz@ONEWAY> Hi Live/Ross, I was hoping that someone can help me out with a hopefully small problem. It appears that MPEG-2 TS ECM packets are being dropped from streams when served from the "testOnDemandRTSPServer". Is there a way to allow these to be passed through to the RTP stream? Best regards Brett Waterson | Founder and Technical ONEWAY NZ LIMITED PO Box 8659 | Symonds Street | Auckland | New Zealand Email: brett at oneway.co.nz Web: www.oneway.co.nz From finlayson at live555.com Tue Sep 26 07:06:52 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 07:06:52 -0700 Subject: [Live-devel] SRTP (Secure RTP) In-Reply-To: <683BC86C0162454BAC43B789A7FB1964393CAC@herlios.adyoron.com> References: <683BC86C0162454BAC43B789A7FB1964393CAC@herlios.adyoron.com> Message-ID: >Is it possible to integrate Live555 RTP with SRTP It is 'possible', but will be a lot of work. It is on our 'to do' list, though... -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Tue Sep 26 07:15:43 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 07:15:43 -0700 Subject: [Live-devel] Your Transport Stream problem In-Reply-To: <20060926104802.DILO11710.fep04.xtra.co.nz@ONEWAY> References: <20060926104802.DILO11710.fep04.xtra.co.nz@ONEWAY> Message-ID: >I was hoping that someone can help me out with a hopefully small problem. Yes, but in the future, please use a relevant Subject: line > >It appears that MPEG-2 TS ECM packets are being dropped from streams when >served from the "testOnDemandRTSPServer". No. The Transport Stream data is passed, as is, from the input file to the outgoing RTP stream. As long as the data is contained within properly-formed Transport Stream data (i.e., 188 bytes, beginning with a sync character), then it will be streamed, as is, within RTP. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From ambuj.sinha at iptvlabs.biz Tue Sep 26 07:28:29 2006 From: ambuj.sinha at iptvlabs.biz (Ambuj Sinha) Date: Tue, 26 Sep 2006 19:58:29 +0530 Subject: [Live-devel] Ref: File Duration for MPEG -TS file Message-ID: <200609261428.k8QESeLJ000938@ns.live555.com> Hi Ross I want to calculate the Transport Bit rate/Transfer Rate for MPEG-TS file stored on the disc so that I can calculate the total File Duration the file will be played. Which part of the live555 library should I refer to and if changes are to be made which specific files should I work on? I have also studied code related to transfer of MPEG-PS files. Currently I am focusing on following files for TS files MPEG2TransportStreamFramer.cpp MPEG2TransportFileServerMediaSubsession.cpp MPEG2TransportStreamMultiplexor.cpp I want to know this to implement Pause on MPEG - TS file. I also found that for computing a transport rate as a number of bytes between PCRs divided by the delta time; and delta time=difference between the PCR values transport rate="number of bytes between PCRs"/"delta time" Please suggest, Thanks Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060926/b01700a0/attachment.html From darnold at futurec.net Tue Sep 26 09:45:19 2006 From: darnold at futurec.net (David Arnold) Date: Tue, 26 Sep 2006 09:45:19 -0700 Subject: [Live-devel] H.264 and MPEG2 Transport Stream In-Reply-To: <200609261428.k8QESeLJ000938@ns.live555.com> Message-ID: We would like to integrate our existing Video Server, which is based on LIVE555, with some LEADTOOLS (www.leadtools.com) components. For RTSP streaming, we have written some custom source modules (H264VideoSource and PCMUAudioSource), which are derived from FramedSource. Audio/Video is stream using these modules and code similar to testOnDemandRTSPServer. We are considering writing a viewer/conferencing application based on some leadtools components. From the LEADTOOLS doc: "The LEAD MPEG2 Transport UDP Source is a DirectShow(tm) filter for streaming data from a UDP port. The data is usually in an MPEG-2 Transport format, as defined in ISO/IEC 13818-1. The filter is also capable of receiving MPEG-2 Transport data packetized with RTP packets conforming to RFC 1889 and RFC 1890." My question is this: What do we need to do to modify our video server so that it can stream the H.264 and PCMU sources within a MPEG-2 Transport Stream so that it can be received by the LEADTOOLS DirectShow filter? Will modifications be required of any existing LIve555 to do this? Thank you, Dave Arnold Future Concepts, La Verne The information contained in this electronic mail transmission is intended only for the use of the individual or entity named above and is privileged and confidential. If you are not the intended recipient, please do not read, copy, use or disclose this communication to others. Any dissemination, distribution or copying of this communication other than to the person or entity named above is strictly prohibited. If you have received this communication in error, please immediately delete it from your system. From finlayson at live555.com Tue Sep 26 10:49:45 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 10:49:45 -0700 Subject: [Live-devel] H.264 and MPEG2 Transport Stream In-Reply-To: References: Message-ID: >My question is this: What do we need to do to modify our video server so >that it can stream the H.264 and PCMU sources within a MPEG-2 Transport >Stream You should be able to do this by writing your own subclass of "MPEG2TransportStreamMultiplexor". It would probably be similar to the existing "MPEG2TransportStreamFromESSource" class, which is used to create a MPEG-2 Transport Stream that contains MEPG Elementary Stream data. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Tue Sep 26 11:03:49 2006 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Sep 2006 11:03:49 -0700 Subject: [Live-devel] Ref: File Duration for MPEG -TS file In-Reply-To: <200609261428.k8QESeLJ000938@ns.live555.com> References: <200609261428.k8QESeLJ000938@ns.live555.com> Message-ID: > I want to calculate the Transport Bit rate/Transfer Rate >for MPEG-TS file stored on the disc so that I can calculate the >total File Duration the file will be played. Which part of the >live555 library should I refer to There's currently nothing in the library that does this specifically. So, you will need to write your own code that reads a MPEG-2 TS file and parses its headers to figure out the bit rate information that you're looking for. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060926/56e1cef2/attachment.html From herurori at yahoo.com Wed Sep 27 00:26:54 2006 From: herurori at yahoo.com (Heru Rori) Date: Wed, 27 Sep 2006 00:26:54 -0700 (PDT) Subject: [Live-devel] asking Message-ID: <20060927072654.51585.qmail@web30709.mail.mud.yahoo.com> i found OC1.exe - 11 error(s), 4 warning(s) while i rebuild all. can help me fix this problems --> live\livemedia\h263plusvideostreamparser.cpp(116) : warning C4390: ';' : empty controlled statement found; is this the intent? --> live\livemedia\rtcp.cpp(125) : warning C4355: 'this' : used in base member initializer list --> live\livemedia\rtpsink.cpp(51) : warning C4355: 'this' : used in base member initializer list --> live\livemedia\rtpsource.cpp(55) : warning C4355: 'this' : used in base member initializer list playCommon.obj : error LNK2001: unresolved external symbol "unsigned int controlConnectionUsesTCP" (?controlConnectionUsesTCP@@3IA) playCommon.obj : error LNK2001: unresolved external symbol "unsigned int allowProxyServers" (?allowProxyServers@@3IA) playCommon.obj : error LNK2001: unresolved external symbol "unsigned int supportCodecSelection" (?supportCodecSelection@@3IA) playCommon.obj : error LNK2001: unresolved external symbol "char * __cdecl getSDPDescriptionFromURL(class Medium *,char const *,char const *,char const *,char const *,unsigned short,unsigned short)" (?getSDPDescriptionFromURL@@YAPADPAVMedium@@PBD111 GG at Z) playCommon.obj : error LNK2001: unresolved external symbol "char * __cdecl getOptionsResponse(class Medium *,char const *,char *,char *)" (?getOptionsResponse@@YAPADPAVMedium@@PBDPAD2 at Z) playCommon.obj : error LNK2001: unresolved external symbol "char const * const clientProtocolName" (?clientProtocolName@@3PBDB) playCommon.obj : error LNK2001: unresolved external symbol "class Medium * __cdecl createClient(class UsageEnvironment &,int,char const *)" (?createClient@@YAPAVMedium@@AAVUsageEnvironment@@HPBD at Z) playCommon.obj : error LNK2001: unresolved external symbol "unsigned int __cdecl clientSetupSubsession(class Medium *,class MediaSubsession *,unsigned int)" (?clientSetupSubsession@@YAIPAVMedium@@PAVMediaSubsession@@I at Z) playCommon.obj : error LNK2001: unresolved external symbol "unsigned int __cdecl clientStartPlayingSession(class Medium *,class MediaSession *)" (?clientStartPlayingSession@@YAIPAVMedium@@PAVMediaSession@@@Z) playCommon.obj : error LNK2001: unresolved external symbol "unsigned int __cdecl clientTearDownSession(class Medium *,class MediaSession *)" (?clientTearDownSession@@YAIPAVMedium@@PAVMediaSession@@@Z) Debug/OC1.exe : fatal error LNK1120: 10 unresolved externals Error executing link.exe. --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060927/5b5b60d2/attachment-0001.html From finlayson at live555.com Wed Sep 27 01:38:04 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 27 Sep 2006 01:38:04 -0700 Subject: [Live-devel] H.264 and MPEG2 Transport Stream Message-ID: >>My question is this: What do we need to do to modify our video server so >>that it can stream the H.264 and PCMU sources within a MPEG-2 Transport >>Stream > >You should be able to do this by writing your own subclass of >"MPEG2TransportStreamMultiplexor". It would probably be similar to >the existing "MPEG2TransportStreamFromESSource" class, which is used >to create a MPEG-2 Transport Stream that contains MEPG Elementary >Stream data. One more thing: You may find the code for the "wis-streamer" application useful. This code makes use of the "MPEG2TransportStreamFromESSource" class, in a way that may be similar to your intended application. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From xcsmith at rockwellcollins.com Wed Sep 27 14:48:23 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Wed, 27 Sep 2006 16:48:23 -0500 Subject: [Live-devel] What is StreamState class used for? Message-ID: Hi LIVE! I am having trouble understanding what RTSPServer::RTSPClientSession::streamState and the StreamState are used for. Can you please explain? I see they are being used by the RTSPServer::RTSPClientSession. I am trying to understand how the RTSPServer::RTSPClientSession uses these "states" to handle the requests on the sessions. Thanks! ~Medra From finlayson at live555.com Wed Sep 27 17:10:08 2006 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 27 Sep 2006 17:10:08 -0700 Subject: [Live-devel] What is StreamState class used for? In-Reply-To: References: Message-ID: >I am having trouble understanding what >RTSPServer::RTSPClientSession::streamState and the StreamState are used >for. Can you please explain? This data structure represents the state of a single media stream (e.g., audio or video) within a single session requested by a remote client. > I see they are being used by the >RTSPServer::RTSPClientSession. I am trying to understand how the >RTSPServer::RTSPClientSession uses these "states" to handle the requests on >the sessions. Notice how the "streamToken" variable is assigned by the call to "getStreamParameters()" (which is called to handle the RTSP "SETUP" command), and notice how later this same "streamToken" variable is used when performing operations on the stream (e.g., starting it, or pausing it). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From ymreddy at ssdi.sharp.co.in Thu Sep 28 02:47:28 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Thu, 28 Sep 2006 15:17:28 +0530 Subject: [Live-devel] Server and Client on the same machine Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CFD3@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Ross, Thanks for the information. We observed the same behavior in the new version of the code. We upgraded "LIVE555 Streaming Media" to version 2006.09.18 and carried the tests on Windows XP Version 5.1, Service Pack 2. On the client side, the function networkReadHandler() in MultiFramedRTPSource.cpp is never called and because of this, client could not able to receive the packets. Any clues on this. Thanks and Regards Y. Mallikharjuna Reddy -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com]On Behalf Of Ross Finlayson Sent: Tuesday, September 26, 2006 3:04 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Server and Client on the same machine >Hi Everybody, > >We are using testMPEG1or2VideoStreamer.cpp and testMPEG1or2VideoReceiver.cpp >for streaming and receiving the MPEG ES and PS streams on the same machine. >We are running the server and client on the same machine. We observed that >sometimes client will never receive any packets i.e. the function >networkReadHandler() in MultiFramedRTPSource.cpp will never be called and >some times, it receives the packets. Even the RTCP reports are not >sent/received properly. > >The same functionality works fine, if the server and client is running on >different machines. > >Any clues on this. This sounds like a problem with your OS. If you're running Windows, I suggest that you upgrade to version 2006.09.18 or later of the "LIVE555 Streaming Media" code, and recompile. If that still doesn't work, then - if you're running an old version of Windows - you may need to upgrade to Windows XP for this to work. If you're not running Windows, then I don't know what might be wrong (although it's probably still an OS problem). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel From ymreddy at ssdi.sharp.co.in Thu Sep 28 04:25:29 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Thu, 28 Sep 2006 16:55:29 +0530 Subject: [Live-devel] Server and Client on the same machine Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CFD5@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Ross, Further to the tests carried on Windows XP for running the server and client on the same machine, I just want to share the following information. We observed that when a MPEG II file is streamed from server to client, the client could able to receive the data for first 5 to 6 times. After this, the function networkReadHandler() in MultiFramedRTPSource.cpp is never called. When we debug this in SingleStep() function in BasicTaskSchedule.cpp file, the handler() functions in two while loops point to RTCP incomingReportHandler() function. Because of this, networkReadHandler() is not called. But after waiting for some time, and start streaming from server to client, the client will receive the data. Any clues on why this is happening. Thanks and Regards Y. Mallikharjuna Reddy -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com]On Behalf Of Mallikharjuna Reddy (NAVT) Sent: Thursday, September 28, 2006 3:17 PM To: 'LIVE555 Streaming Media - development & use' Subject: Re: [Live-devel] Server and Client on the same machine Hi Ross, Thanks for the information. We observed the same behavior in the new version of the code. We upgraded "LIVE555 Streaming Media" to version 2006.09.18 and carried the tests on Windows XP Version 5.1, Service Pack 2. On the client side, the function networkReadHandler() in MultiFramedRTPSource.cpp is never called and because of this, client could not able to receive the packets. Any clues on this. Thanks and Regards Y. Mallikharjuna Reddy -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com]On Behalf Of Ross Finlayson Sent: Tuesday, September 26, 2006 3:04 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Server and Client on the same machine >Hi Everybody, > >We are using testMPEG1or2VideoStreamer.cpp and testMPEG1or2VideoReceiver.cpp >for streaming and receiving the MPEG ES and PS streams on the same machine. >We are running the server and client on the same machine. We observed that >sometimes client will never receive any packets i.e. the function >networkReadHandler() in MultiFramedRTPSource.cpp will never be called and >some times, it receives the packets. Even the RTCP reports are not >sent/received properly. > >The same functionality works fine, if the server and client is running on >different machines. > >Any clues on this. This sounds like a problem with your OS. If you're running Windows, I suggest that you upgrade to version 2006.09.18 or later of the "LIVE555 Streaming Media" code, and recompile. If that still doesn't work, then - if you're running an old version of Windows - you may need to upgrade to Windows XP for this to work. If you're not running Windows, then I don't know what might be wrong (although it's probably still an OS problem). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel From finlayson at live555.com Thu Sep 28 08:21:09 2006 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 28 Sep 2006 08:21:09 -0700 Subject: [Live-devel] Server and Client on the same machine In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520337CFD5@ssdimailsrvnt01.ssdi.sharp.co.in > References: <7FB4685EA93D014C8E30AA087B66E7520337CFD5@ssdimailsrvnt01.ssdi.sharp.co.in > Message-ID: >We observed that when a MPEG II file is streamed from server to client, the >client could able to receive the data for first 5 to 6 times. After this, >the function networkReadHandler() in MultiFramedRTPSource.cpp is never >called. This means that your application is no longer receiving network packets. If you didn't modify the code at all, then this suggests a network and/or OS problem. (If you did modify the code, then you're on your own - sorry.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From rahul.ray at gmail.com Thu Sep 28 09:32:38 2006 From: rahul.ray at gmail.com (Rahul Ray) Date: Thu, 28 Sep 2006 09:32:38 -0700 Subject: [Live-devel] openRTSP -m ... Message-ID: Hi, I would like to extract the JPEG file infos which -m option writes on to disk , before writing. I want to avoid writing on disk and create an IplImage (opencv) on the fly. Could you please mention, which function to examine for this ? Thanks for your help. -best, Ray From ymreddy at ssdi.sharp.co.in Fri Sep 29 00:18:02 2006 From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT)) Date: Fri, 29 Sep 2006 12:48:02 +0530 Subject: [Live-devel] Interoperability tests for LIVE Client Message-ID: <7FB4685EA93D014C8E30AA087B66E7520337CFD9@ssdimailsrvnt01.ssdi.sharp.co.in> Hi Everybody, Just want to find out whether anybody has carried interoperability tests for LIVE Client with any standard RTP server. Appreciate, if you can share the information. Thanks and Regards Y. Mallikharjuna Reddy From chenw at blrcsv.china.bell-labs.com Fri Sep 29 01:54:52 2006 From: chenw at blrcsv.china.bell-labs.com (chenwei) Date: Fri, 29 Sep 2006 16:54:52 +0800 Subject: [Live-devel] A question About sending stream to Darwin Message-ID: <20060929083932.72AB78719@blrcsv.china.bell-labs.com> Dear All: I used liveMedia send a stream to Darwin(testMPEG1or2AudioVideoToDarwin),and I can receive it use realplay using rtsp://myserver/test.sdp. Now I want to know: (1) What kind of stream can Darwin receives and forwards? Is there any documentation? (2) Can I receive the forwarded stream using "relay"? So I can use the multicast function. And how? Many thanks! ????????chenwei ????????chenw at blrcsv.china.bell-labs.com ??????????2006-09-29 From ambuj.sinha at iptvlabs.biz Fri Sep 29 02:44:07 2006 From: ambuj.sinha at iptvlabs.biz (Ambuj Sinha) Date: Fri, 29 Sep 2006 15:14:07 +0530 Subject: [Live-devel] File Duration Message-ID: <200609290944.k8T9iDvj060905@ns.live555.com> Hi Ross I have seen a implementation of the function MPEG1or2ProgramStreamFileDuration() in the MPEG1or2FileServerDemux.cpp file which calculates the Program Stream File Duration which refers to getMPEG1or2TimeCode() and computeSCRTimeCode() function . Could you please elaborate how getMPEG1or2TimeCode and computeSCRTimeCode() works. I am also writing a similar function MPEG2TransportStreamFileDuration() to calculate the time duration of MPEG TS file duration. Please suggest the approach I need to calculate it. Thanks Ambuj Kumar Sinha iptv Scientist iptvLabs (India) Pvt. Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060929/c6c7e854/attachment-0001.html From xcsmith at rockwellcollins.com Fri Sep 29 15:36:30 2006 From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com) Date: Fri, 29 Sep 2006 17:36:30 -0500 Subject: [Live-devel] Cannot create a recording Message-ID: Hi LIVE! My colleague and I have been struggling to record a stream. Our program is similar to the openRTSP test program. But we have this problem: When subsession->sink->startplaying() is called, the application appears to step through 1 series of getNextFrame() and other calls on the MediaSubsession socket (source) and during that time, turn on background read handling. Then we enter the environment loop, and these functions to use the data are never called again. BUT -> Only when I run on one of our 3 Fedora5 machines. When I run the program on one of our two Fedora4 servers, both are able to make a recording. I have the same problem if I use openRTSP to try to make the recording. The file permissions appear to be the same on all of these machines. All of the machines allow my program to create a file, but the Fedora5 machines never add any data to the file. With debugging set, Groupsock::handleRead is able to print that it sees 1300 bytes of data regularly on the Fedora5 machines. My colleague is running a live version from mid august, and I am running the latest version available as of 9/29 on your site. Where could I start looking in the LIVE to find out why writes are never happening on some machines? Thanks! ~Medra From p4olo_prete at yahoo.it Fri Sep 29 16:47:39 2006 From: p4olo_prete at yahoo.it (Paolo Prete) Date: Sat, 30 Sep 2006 01:47:39 +0200 (CEST) Subject: [Live-devel] redirecting a mpeg4 rtsp stream Message-ID: <20060929234739.60141.qmail@web28007.mail.ukl.yahoo.com> Hi all, I have an IP cam whith an embedded mpeg4 rtsp streaming server; unfortunately, the max number of allowed clients is small. In order to solve this limit, I would like to create a new simple streaming server, on my machine, which only redirectes the ip-cam's stream. Is there a support, or sample codes for this operation in an easy way, or should I do it using the OpenRTSP (--->RTSPClient) and/or testMPEG4VideoStreamer programs as a base for my purpose (but it wouldn't be trivial...)? thanks, Paolo Prete __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it From finlayson at live555.com Fri Sep 29 16:46:36 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 29 Sep 2006 16:46:36 -0700 Subject: [Live-devel] Cannot create a recording In-Reply-To: References: Message-ID: >My colleague and I have been struggling to record a stream. Our program is >similar to the openRTSP test program. But we have this problem: >When subsession->sink->startplaying() is called, the application appears to >step through 1 series of getNextFrame() and other calls on the >MediaSubsession socket (source) and during that time, turn on background >read handling. Then we enter the environment loop, and these functions to >use the data are never called again. BUT -> Only when I run on one of our >3 Fedora5 machines. When I run the program on one of our two Fedora4 >servers, both are able to make a recording. I have the same problem if I >use openRTSP to try to make the recording. The file permissions appear to >be the same on all of these machines. All of the machines allow my program >to create a file, but the Fedora5 machines never add any data to the file. >With debugging set, Groupsock::handleRead is able to print that it sees >1300 bytes of data regularly on the Fedora5 machines. My colleague is >running a live version from mid august, and I am running the latest version >available as of 9/29 on your site. Where could I start looking in the LIVE >to find out why writes are never happening on some machines? To simplify your testing, you should begin with the original, unmodified "openRTSP" application, using the latest version of the source code. (We can't give you any support once you've modified the original code, or if you are using an old version.) Also, you should begin by having "openRTSP" attempt regular file output, and *not* to stdout (i.e., *don't* use the "-a", "-v", "-q", "-4", or "-i" options). I.e., first find the simplest example that does not work. Then, look at the "FileSink::afterGettingFrame1()" function. This function should get called each time a new piece of data arrives via RTP. If this function doesn't get called at all (or only gets called once), then send us the diagnostic output from "openRTSP" (i.e., generated by adding the "-V" flag), and we'll take a look at it. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Fri Sep 29 16:53:43 2006 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 29 Sep 2006 16:53:43 -0700 Subject: [Live-devel] redirecting a mpeg4 rtsp stream In-Reply-To: <20060929234739.60141.qmail@web28007.mail.ukl.yahoo.com> References: <20060929234739.60141.qmail@web28007.mail.ukl.yahoo.com> Message-ID: >I have an IP cam whith an embedded mpeg4 rtsp >streaming server; >unfortunately, the max number of allowed clients is >small. >In order to solve this limit, I would like to create a >new simple streaming server, on my machine, which only >redirectes the ip-cam's stream. > >Is there a support, or sample codes for this operation No, not really - sorry. This is something that you would need to develop yourself. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From pony8420 at gmail.com Fri Sep 29 23:04:04 2006 From: pony8420 at gmail.com (cn PACO) Date: Sat, 30 Sep 2006 14:04:04 +0800 Subject: [Live-devel] testOnDemandRTSPServer.cpp question Message-ID: <5ec8aa350609292304w39d1546dg1ef6e2baff5ac07c@mail.gmail.com> I rewrite the testOnDemandRTSPServer.cpp as follows. The testOnDemandRTSPServer.CPP will produce two URLs, each URL include a m4v video separately. I use the Ethereal tools to analyze the packet. According to the Ethereal analytical result, each URL has a pair port is the RTP port and RTCP port. So this RTSP server will have two each RTP ports, two each RTCP ports and a RTSP port. Do these two URLs have the way to use a pair port( one RTP port and one RTCP port ) together ? Ignoring the RTSP contain a few URLs, I hope the RTSP server only uses one RTSP port, one RTP, and one RTCP port. ***************************************************************************************************** int main(int argc, char** argv) { // Begin by setting up our usage environment: TaskScheduler* scheduler = BasicTaskScheduler::createNew(); env = BasicUsageEnvironment::createNew(*scheduler); // Create the RTSP server: RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, NULL, 45); if (rtspServer == NULL) { *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n"; exit(1); } char const* descriptionString = "Session streamed by \"testOnDemandRTSPServer\""; // 1: A MPEG-4 video elementary stream: { char const* streamName = "Video01"; char const* inputFileName = "test1.m4v"; ServerMediaSession* sms = ServerMediaSession::createNew(*env, streamName, streamName, descriptionString); sms->addSubsession(MPEG4VideoFileServerMediaSubsession ::createNew(*env, inputFileName, reuseFirstSource)); rtspServer->addServerMediaSession(sms); } // 2: A MPEG-4 video elementary stream: { char const* streamName = "Video02"; char const* inputFileName = "test2.m4v"; ServerMediaSession* sms = ServerMediaSession::createNew(*env, streamName, streamName, descriptionString); sms->addSubsession(MPEG4VideoFileServerMediaSubsession ::createNew(*env, inputFileName, reuseFirstSource)); rtspServer->addServerMediaSession(sms); } env->taskScheduler().doEventLoop(); return 0; } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20060929/fd7a526e/attachment.html From p4olo_prete at yahoo.it Sat Sep 30 12:59:30 2006 From: p4olo_prete at yahoo.it (Paolo Prete) Date: Sat, 30 Sep 2006 21:59:30 +0200 (CEST) Subject: [Live-devel] redirecting a mpeg4 rtsp stream In-Reply-To: Message-ID: <20060930195930.69443.qmail@web28009.mail.ukl.yahoo.com> I would like to > create a > >new simple streaming server, on my machine, which > only > >redirectes the ip-cam's stream. > > > >Is there a support, or sample codes for this > operation > > No, not really - sorry. This is something that you > would need to > develop yourself. > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ Anyway, before starting a development, i ask: 1)would it be a good idea starting with playCommon.cpp? I have seen that the code makes a RTSPClient for the incoming streaming and then creates some different fileSinks (QuickTimeFileSink, AVIFileSink etc.); should i proceed using a MPEG4ESVideoRTPSink instead? 2) if I use a MPEG4ESVideoRTPSink, and then I call the startPlaying function which kind of MediaSource should I pass to the function's first argument? In testMPEG4Streamer example, a ByteStreamFileSource is passed: in my case, should I pass a MPEG4(generic or ES)RTPSource ? Thanks again for your help Paolo Prete Italy __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it