Hi Ross Our company policy doesn't allow external excess to our network sorry for it :-), but would like to know what can be the problem behind this incomplete file streaming, as bit band player is streaming File till end while our openRTSP is not streaming till end. With our openRTSP test application, Server is exiting after streaming Half of the stream. Regards Ravinder
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 29, 2007 4:05 PM To: [EMAIL PROTECTED] Subject: live-devel Digest, Vol 43, Issue 21 Send live-devel mailing list submissions to live-devel@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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of live-devel digest..." Today's Topics: 1. Re: Read multiple frames? (Ross Finlayson) 2. Re: Problem with openRTSP and testMPEG4VideoStreamer (Igor Trevisan) 3. SeqNo ([EMAIL PROTECTED]) 4. H264 Framerate (Abe Friesen) 5. Re: SeqNo (Ross Finlayson) 6. How to Stream different media format(mpeg4 & pcm) at the same time? (???) 7. Re: How to Stream different media format(mpeg4 & pcm) at the same time? (Ross Finlayson) 8. Re: How can I change the received packet at the client by using openRTSP? (jerry zhao) 9. stream not played till end (singh, Ravinder) 10. Re: stream not played till end (Ross Finlayson) ---------------------------------------------------------------------- Message: 1 Date: Mon, 28 May 2007 01:53:10 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] Read multiple frames? To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" ; format="flowed" >>Therefore, if you are feeding input from a "MPEG1or2VideoRTPSource" >>into a decoder, and your decoder is not smart enough to decode one >>slice at a time, then you must aggregate the input data into >>complete video frames before feeding them to your decoder. > >Can this be done with live555 stuff? No. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ ------------------------------ Message: 2 Date: Mon, 28 May 2007 11:03:52 +0200 From: "Igor Trevisan" <[EMAIL PROTECTED]> Subject: Re: [Live-devel] Problem with openRTSP and testMPEG4VideoStreamer To: "LIVE555 Streaming Media - development & use" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" > You didn't properly read the "openRTSP" documentation (in particular, > the meaning of the "-o" option) Thanks Ross! I apologize for my questions: it was a typical case in which a "RTFM" as answer would have been appropriate! ;) I. -- "Much less doesn't mean zero" -- E.Benetti -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20070528/931e5 d5f/attachment-0001.html ------------------------------ Message: 3 Date: Mon, 28 May 2007 23:43:26 +0200 (CEST) From: [EMAIL PROTECTED] Subject: [Live-devel] SeqNo To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain;charset=iso-8859-1 Hello *, I have a question regarding the interface between the live library and mplayer. As expected, in the case of bigger frames, they are split in more RTP packages. These packages, are put together in live and assembled in a bigger package, that contains the whole frame. How could I have access from mplayer (ReadBufferQueue class, of the demux_rtp.cpp) to the number of rtp packages each frame has been composed from? I can tell if there is loss at mplayer level by inspecting the last sequence number from bufferQueue->rtpSource->curPacketRTPSeqNum() before and after calling the getBuffer(...) in the demux_rtp_fill_buffer function of demux_rtp.cpp file. But I can not know if there are multiple frames lost or just one. If I could know how many RTPpackets each frame is composed on at live level it would be enough to find out how many frames were lost. Any information would be helpful. Regards, Silviu Homoceanu. ------------------------------ Message: 4 Date: Mon, 28 May 2007 16:52:10 -0700 From: "Abe Friesen" <[EMAIL PROTECTED]> Subject: [Live-devel] H264 Framerate To: "'LIVE555 Streaming Media - development & use'" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Is it possible to determine the frame rate of an H264 encoded asset (encoded as an Annex B bitstream (NAL units preceded by 0x00 0x00 0x00 0x01)) from the data stream so that I can accurately set fDurationInMicroseconds in the framer? Also, from the H264 spec it looks like the only way to determine whether the current NAL unit ends an access unit is by examining the following NAL unit to determine if it starts an access unit - or am I missing something? Thanks, Abe Friesen ------------------------------ Message: 5 Date: Mon, 28 May 2007 17:33:16 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] SeqNo To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" ; format="flowed" >But I can not know if there are multiple frames >lost or just one. If I could know how many RTPpackets each frame is >composed on at live level it would be enough to find out how many frames >were lost. As you noted, the "RTPSource" abstraction delivers complete 'frames'. (However, the term 'frames' is somewhat of a misnomer, because they do not always correspond to entire video frames - e.g., for MPEG-1 or 2, it delivers complete 'slices'; for H.264, it delivers complete NAL units). The number of RTP packets that made up each frame is (deliberately) not exposed outside the "RTPSource" interface. Data receivers (e.g., audio or video decoders) should not care about this information. However, if you just want to find out packet loss rates, then you can do so by looking at the "RTPReceptionStats" data. (Alternatively, the data sender (i.e., server) can look at RTCP Reception Report ("RR") coming back from the receiver; this data is recorded in a "RTPTransmissionStats" object.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ ------------------------------ Message: 6 Date: Tue, 29 May 2007 12:03:36 +0900 From: ??? <[EMAIL PROTECTED]> Subject: [Live-devel] How to Stream different media format(mpeg4 & pcm) at the same time? To: "live Media" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="ks_c_5601-1987" Hi~ ^^ Thanks for your answer~ ^^ testonDemandRTSPServer can stream multiple media files at the same time,then how can I use/modify the testOnDemandRTSPServer or livemedia server to stream Mpeg4 video & pcm (wave) audio at the same time to single client(just one connection), i.e how can I multiplex & stream separate mpeg4 video data & PCM (wave) audio data.?.. is it possible? Actually our system have seperate stream for mpeg4 video & PCM audio which I wanna stream using livemedia to play at the same time by single media player. Using livemedia source Im able to stream mpeg4 video & pcm wave audio seperately . but how can i stream them together to play at the same time by same player. ------------------------------ Message: 7 Date: Mon, 28 May 2007 23:06:59 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] How to Stream different media format(mpeg4 & pcm) at the same time? To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" ; format="flowed" >testonDemandRTSPServer can stream multiple media files at the same >time,then how can I use/modify the testOnDemandRTSPServer or >livemedia server to stream Mpeg4 video & pcm (wave) audio at the >same time to single client(just one connection), i.e how can I >multiplex & stream separate mpeg4 video data & PCM (wave) audio >data.?.. is it possible? Yes. You would create a "ServerMediaSession" object, and add two separate "ServerMediaSubsession" objects to it - one for the MPEG-4 video stream, one for the PCM audio stream. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ ------------------------------ Message: 8 Date: Tue, 29 May 2007 10:55:40 +0200 From: "jerry zhao" <[EMAIL PROTECTED]> Subject: Re: [Live-devel] How can I change the received packet at the client by using openRTSP? To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" >>I used the testOnDemandRTSPServer to stream video und openRTSP to >>receive the streamed data. My question is: where I should change the >>received packet data before the client stores the received packet on >>the disk, e.g. adding some data or a special header to the received >>packets. >You would need to insert a new filter object (i.e., a subclass of >"FramedFilter" - that you would write), and insert it in the data >chain, in front of the "FileSink" object. (You would do this in >"testProgs/playCommon.cpp".) Hi, Ross, thanks for your reply. If one frame consists of several packets and I need to add some additional data to this frame until all the packets of this frame are received, can I still implement this by writing a subclass of FramedFilter? Is there any reference code for this kind of implementation? Thanks again. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20070529/b38de 147/attachment-0001.html ------------------------------ Message: 9 Date: Tue, 29 May 2007 14:50:25 +0530 From: "singh, Ravinder" <[EMAIL PROTECTED]> Subject: [Live-devel] stream not played till end To: <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Hi All I am trying to use your openRTSP application to stream data from our Bit Band VOD server, what I have Observed is openRTSP never streams complete video data, if the file is around 50MB only 16MB is streamed, Would like to know why this is happening. I am writing to file, and using following command . /openRTSP -V -Q "rtsp://172.24.141.104:554/Video/nature_mp_544x480_2000.ts" Server is correctly reporting npt time (Total Duration of Call) as a=range: npt=0.0-356.774, while only 60 sec streams Is streamed. regards Ravinder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.live555.com/pipermail/live-devel/attachments/20070529/625b2 f1e/attachment-0001.html ------------------------------ Message: 10 Date: Tue, 29 May 2007 03:34:52 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] stream not played till end To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" >I am trying to use your openRTSP application to stream data from our >Bit Band VOD server, what I have >Observed is openRTSP never streams complete video data, if the file >is around 50MB only 16MB is streamed, >Would like to know why this is happening. >I am writing to file, and using following command >. /openRTSP -V -Q "rtsp://172.24.141.104:554/Video/nature_mp_544x480_2000.ts" Sorry, but your host "172.24.141.104" is not reachable from our network, therefore I can't test this for myself. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... 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