Hi all,
2) an "add" signal???? But how add clients using a signal? Nicola ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In data sabato 30 maggio 2009 15:49:32, MailingList SVR ha scritto:
: > Hi all, > > there is something not much clear to me about multiupdsink: I would like to dinamycally add clients to multiudpsink, based on the documentation (http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-multiudpsink.html) there are: > > 1) a clients property I can populate with the desidered clients, ok is fine > 2) an "add" signal???? But how add clients using a signal? > > I tried to modify the clients property while the pipeline is running but this didn't work, so the only way if one is to use the add signal but I don't know how to use a signal to add a client can you give me some examples please? I'm using the python bindings, > > thanks > Nicola > Ok I managed to change the property using the signal but this doesn't work dinamically, for example if "a" is my multiudpsink element I can do something like: ------------------------------------------------------------------------------ OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In reply to this post by Mailing List SVR
In data sabato 30 maggio 2009 15:49:32, MailingList SVR ha scritto:
: > Hi all, > > there is something not much clear to me about multiupdsink: I would like to dinamycally add clients to multiudpsink, based on the documentation (http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-multiudpsink.html) there are: > > 1) a clients property I can populate with the desidered clients, ok is fine > 2) an "add" signal???? But how add clients using a signal? > > I tried to modify the clients property while the pipeline is running but this didn't work, so the only way if one is to use the add signal but I don't know how to use a signal to add a client can you give me some examples please? I'm using the python bindings, > > thanks > Nicola > Nicola ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In reply to this post by Mailing List SVR
Hi, I was trying to transfer video and audio using network. I used teh examples from the net to do that and succeeded. But now I wanted to save the stream into file and faced with some problem. Please look at the following command:
gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! ffdec_h263 ! xvimagesink ----- this is what i used to accept and display a video stream. So, to save the stream into a file I changed the last two elements (the ffmpeg decoder and xvimake sink). I thought that since the packet coming from the other machine is already encoded in h263p codec, replacing these two elements with the following elements would solve my problem: I used these elments: avimux ! filesink location=testnet.avi . That is, i connected the rtph263pdepay element to the avimux element and to the file sink element sequentially as follows. gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! avimux ! filesink location=test.avi But I got an error, that says: streaming task paused, reason not-negotiated (-4) Please help me on how I can save a stream. Thank you. - Zelalem S. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi, Hi ,, Try providing caps between rtph263pdepay and avimux . On Sat, May 30, 2009 at 8:28 PM, Zelalem Sintayehu <[hidden email]> wrote:
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Hi Sudarshan, thank you for your prompt response. But, I am new to gstream and I think I don't understand your comment. Sorry to bother you again. I tried to set the caps property of the rtpdepayloader by putting caps="application/x-rtp, media...." for the rtph263pdepay element, but I get the following error: WARNING: erroneous pipeline: no property "caps" in element "rtph263pdepay0". So, do i need to include a new element in between the two elements (rtp263depay and avimux) or how can i provide caps?
Thank you again for your assistance. - Zelalem S. Date: Sun, 31 May 2009 10:58:31 +0530 From: [hidden email] To: [hidden email] Subject: Re: [gst-devel] How to save a stream from a network into a file Hi, Hi ,, Try providing caps between rtph263pdepay and avimux . On Sat, May 30, 2009 at 8:28 PM, Zelalem Sintayehu <[hidden email]> wrote:
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Hi ,
for exapmle try these :-
gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263,width=176,height=144,framerate=25 ' ! avimux ! filesink location=test.avi
or gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263 ' ! avimux ! filesink location=test.avi
On Mon, Jun 1, 2009 at 1:00 PM, Zelalem Sintayehu <[hidden email]> wrote:
-- Regards, Sudarshan Bisht ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi Sudarshan, thank you again for the prompt response. I tried your suggestion but still got error. When i try the first suggestion, I got the following error:
WARNING: erroneous pipeline: could not parse caps "video/x-h263,width=176,height=144,framerate=25\" And then I removed the space after 25 (the frame rate - please look at the suggested caps below) and got the following error: WARNING: erroneous pipeline: could not link rtph263pdepay0 to avimux0. It considers as if the caps is not there. When I use the second suggestion again with the space after 263, i got the same kind of error as above. When i delete the space, I got the following error: streaming task paused, reason not-negotiated (-4) I don't know what is hapening. BTW, on the sender side I used the following command: gst-launch v4l2src ! video/x-raw-rgb,width=320,height=240 ! queue ! videorate ! video/x-raw-rgb,rate=15/1 ! ffmpegcolorspace! ffenc_h263p ! rtph263ppay ! udpsink port=5000 I hope this will give you the clue about the problem. I run the command on Ubuntu Interpid. Thank you. - Zelalem S. Date: Mon, 1 Jun 2009 13:52:34 +0530 From: [hidden email] To: [hidden email] Subject: Re: [gst-devel] How to save a stream from a network into a file Hi ,
for exapmle try these :-
gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263,width=176,height=144,framerate=25 ' ! avimux ! filesink location=test.avi
or gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263 ' ! avimux ! filesink location=test.avi
On Mon, Jun 1, 2009 at 1:00 PM, Zelalem Sintayehu <[hidden email]> wrote:
-- Regards, Sudarshan Bisht See all the ways you can stay connected to friends and family ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In reply to this post by Sudarshan Bisht
Hi Sudarshan, thank you again for the prompt response. I tried your
suggestion but still got error. When i try the first suggestion, I got
the following error:
WARNING: erroneous pipeline: could not parse caps "video/x-h263,width=176,height=144,framerate=25\" And then I removed the space after 25 (the frame rate - please look at the suggested caps below) and got the following error: WARNING: erroneous pipeline: could not link rtph263pdepay0 to avimux0. It considers as if the caps is not there. When I use the second suggestion again with the space after 263, i got the same kind of error as above. When i delete the space, I got the following error: streaming task paused, reason not-negotiated (-4) I don't know what is hapening. BTW, on the sender side I used the following command: gst-launch v4l2src ! video/x-raw-rgb,width=320,height=240 ! queue ! videorate ! video/x-raw-rgb,rate=15/1 ! ffmpegcolorspace! ffenc_h263p ! rtph263ppay ! udpsink port=5000 I hope this will give you the clue about the problem. I run the command on Ubuntu Interpid. Thank you. - Zelalem S. - Zelalem S. Date: Mon, 1 Jun 2009 13:52:34 +0530 From: [hidden email] To: [hidden email] Subject: Re: [gst-devel] How to save a stream from a network into a file Hi ,
for exapmle try these :-
gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263,width=176,height=144,framerate=25 ' ! avimux ! filesink location=test.avi
or gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263 ' ! avimux ! filesink location=test.avi
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Hi ,
Actually i treid following pipeline on my machine , and it works . ( I dont have webcam so tried it using videotestsrc) .
gst-launch videotestsrc ! 'video/x-raw-yuv,width=320,heigh=240,framerate=(fraction)25/1,format=(fourcc)I420' ! ffenc_h263p ! rtph263ppay ! udpsink port=5000
gst-launch udpsrc port=5000 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! ffdec_h263 ! avimux ! filesink location=test.avi
And ensure that u have latest Gstreamer installed .
On Mon, Jun 1, 2009 at 5:43 PM, Zelalem Sintayehu <[hidden email]> wrote:
-- Regards, Sudarshan Bisht ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi Sudarshan, thank you very much! It is now working. BTW, you have added the element, ffdec_263, after the depayloader. That is the reason why it started to work. You know I had thought of that as well, but the reason why i didn't try it is that the rtph263pdepay gives us a stream that is encoded with h263 codec (that is what we recieved from teh sender). So, i thought it only needs to give this stream to the muxer and then the muxer can store it into file. I still don't understand why (especially a decoder - which I think will decode it back to raw format) is required. Could you please explain for me the idea behind. BTW, the first few frames are very distorted (blurred).
I am using Gsreater 0.10 version. I think that is the latest. Thank you again. It is really a great help. - Zelalem S. Date: Mon, 1 Jun 2009 19:08:15 +0530 From: [hidden email] To: [hidden email] Subject: Re: [gst-devel] How to save a stream from a network into a file Hi ,
Actually i treid following pipeline on my machine , and it works . ( I dont have webcam so tried it using videotestsrc) .
gst-launch videotestsrc ! 'video/x-raw-yuv,width=320,heigh=240,framerate=(fraction)25/1,format=(fourcc)I420' ! ffenc_h263p ! rtph263ppay ! udpsink port=5000
gst-launch udpsrc port=5000 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! ffdec_h263 ! avimux ! filesink location=test.avi
And ensure that u have latest Gstreamer installed .
check out the rest of the Windows Live™. More than mail–Windows Live™ goes way beyond your inbox. More than messages ------------------------------------------------------------------------------ Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT is a gathering of tech-side developers & brand creativity professionals. Meet the minds behind Google Creative Lab, Visual Complexity, Processing, & iPhoneDevCamp as they present alongside digital heavyweights like Barbarian Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In reply to this post by Sudarshan Bisht
Hi Zelalem
Use the following pipeline, here i have set the proper caps between rtph263pdepay and avimux . Make sure at whatever width, height and framerate you are capturing at source side use the same numbers here on receiver side also. This works on my machine. ( I think this suggestion i have already given earlier but i missed (fraction) part in framerate caps )
gst-launch udpsrc port=5000 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263,width=320,height=240,framerate=(fraction)25/1' ! avimux ! filesink location=test.avi
On Mon, Jun 1, 2009 at 7:08 PM, sudarshan bisht <[hidden email]> wrote:
-- Regards, Sudarshan Bisht ------------------------------------------------------------------------------ OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi Sudarshan, thank you very much. It is working. I really appreciate your assistance. I think this is the proper way of doing it as it doesn't require the decoding and encoding of an already encoded video stream. The quality would deteriorate. BTW, the error message in Gstream is not user friendly, it should have said that it is incorect syntax or something similar. Anyway, I am happy. BTW, have you seen the first frame from the file? It is very blurred. When I play the video from the file it starts with a very blurred frame. What do you think is the reason?
Thanks again for your support. Best regards, - Zelalem S. Date: Tue, 2 Jun 2009 10:07:20 +0530 From: [hidden email] To: [hidden email] Subject: Re: [gst-devel] How to save a stream from a network into a file Hi Zelalem
Use the following pipeline, here i have set the proper caps between rtph263pdepay and avimux . Make sure at whatever width, height and framerate you are capturing at source side use the same numbers here on receiver side also. This works on my machine. ( I think this suggestion i have already given earlier but i missed (fraction) part in framerate caps )
gst-launch udpsrc port=5000 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263,width=320,height=240,framerate=(fraction)25/1' ! avimux ! filesink location=test.avi
On Mon, Jun 1, 2009 at 7:08 PM, sudarshan bisht <[hidden email]> wrote:
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Hi Sudarshan, by the way, the receiver continues to run even when the sender is stopped. Is there a way that i can tell to udpsrc to stop receiving if there is no data on the port? Actually, when i see the file there is no empty part. I mean it contains only the data sent from the sender side and not an empty frame as I had expected. Since I want to use in a program later, i want to stop the recorder when the sender finishes sending the video. I hope there could be a way to do that.
Thank you. Best regards, - Zelalem S. Windows Live™: Keep your life in sync. Check it out! ------------------------------------------------------------------------------ OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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