Hi, Using the following gst-launch command, I could dump mp3
data. But I want to play it live. Can you please let me know how to achieve
that? Say my host IP: 192.168.15.174 Receiver (Rx): gst-launch -v
udpsrc ! filesink location=dump.mp3 Transmitter (Tx): gst-launch -v
filesrc location=MPEG25_11_32_m.mp3 ! udpsink Thanks in advance Regards, Ramana
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Tried the following also. But did not
work. Receiver (Rx) gst-launch filesrc
location=MPEG25_11_32_m.mp3 ! udpsink host=192.168.15.174 port=5000 sync=false Transmitter (Tx) gst-launch udpsrc port=5000 ! ffdec_mp3 !
alsasink Regards, Ramana From: thiagoss
[mailto:[hidden email]] Probably removing the
filesink and putting "mad ! alsasink" On Wed, Mar 26, 2008 at 3:09 AM, Hi,
Using
the following gst-launch command, I could dump mp3 data. But I want to play it
live. Can you please let me know how to achieve that? Say my host IP: 192.168.15.174 Receiver
(Rx): gst-launch -v udpsrc ! filesink
location=dump.mp3 Transmitter
(Tx): gst-launch -v filesrc
location=MPEG25_11_32_m.mp3 ! udpsink Thanks
in advance Regards, Ramana
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Hello, Ramana
On Wed, Mar 26, 2008 at 10:46 PM, Ramana Reddy Polaka <[hidden email]> wrote: > Tried the following also. But did not work. > > Receiver (Rx) > gst-launch filesrc location=MPEG25_11_32_m.mp3 ! udpsink host=192.168.15.174 > port=5000 sync=false > > Transmitter (Tx) > gst-launch udpsrc port=5000 ! ffdec_mp3 ! alsasink Do you have to use UDP? You might want to try with tcpserversrc/tcpclientsink. Greetings! Daniel Díaz [hidden email] > Regards, > Ramana > > ________________________________ > > > From: thiagoss [mailto:[hidden email]] > Sent: Wednesday, March 26, 2008 8:40 PM > To: Ramana Reddy Polaka > Subject: Re: [gst-devel] Network streaming using udpsrc and udpsink > > > > Probably removing the filesink and putting "mad ! alsasink" > > > > > On Wed, Mar 26, 2008 at 3:09 AM, Ramana Reddy Polaka wrote: > > > > > > Hi, > > > > Using the following gst-launch command, I could dump mp3 data. But I want to > play it live. Can you please let me know how to achieve that? Say my host > IP: 192.168.15.174 > > > > Receiver (Rx): > > gst-launch -v udpsrc ! filesink location=dump.mp3 > > > > Transmitter (Tx): > > gst-launch -v filesrc location=MPEG25_11_32_m.mp3 ! udpsink > > > > > > > > Thanks in advance > > > > Regards, > > Ramana ------------------------------------------------------------------------- Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Try this:
> gst-launch filesrc location=MPEG25_11_32_m.mp3 ! mp3parse ! udpsink host=192.168.15.174 > port=5000 > > gst-launch udpsrc port=5000 ! typefind ! ffdec_mp3 ! alsasink Setting to sync false will make udpsink push too much data, and the other process won't be able to receive that fast, probably most of them may not arrive. And using mp3parse puts timestamps on the buffers, that makes it possible to udpsink to send buffers on the right time. and on the receiver ffdec_mp3 requires that it receives "mp3" streams, already with a type, udpsrc provides "any", so typefind would help. Hope it works. Thiago
On Thu, Mar 27, 2008 at 3:01 AM, Daniel Díaz <[hidden email]> wrote: Hello, Ramana ------------------------------------------------------------------------- Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Thank you. The following works fine Server: gst-launch filesrc location=MPEG25_11_32_m.mp3 !
mp3parse ! udpsink host=192.168.15.174 port=5000 From:
[hidden email]
[mailto:[hidden email]] On Behalf Of thiagoss Try this: On Thu, Mar 27, 2008 at 3:01 AM, Daniel Díaz <[hidden email]> wrote: Hello, Ramana < wrote: > Tried the following
also. But did not work. Do you have to use UDP? You might want to try with
tcpserversrc/tcpclientsink.
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Hi All, I have tried saving the below pipeline to
xml file using “gst-launch” with –output option [example from gstrtpbin reference] gst-launch –-output=test.gst gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
But I get an error when I try to load this
pipeline using “gst-xmllaunch” command line or using “gst_xml_get_element”
in my test program. The error is basically an assert failure
for “pad != NULL”. If I construct simple pipeline like “gst-launch
--output=test.gst v4lsrc ! xvimagesink” and launch the pipeline using “gst-xmllaunch
test.gst”, it works perfectly fine. Looks like I have an issue when I use “pads”.
Is there anything wrong in my procedure? Please give me some clue to proceed. Thanks & Regards Sreejesh R B
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In reply to this post by Thiago Sousa Santos-2
Hi, How can I access external buffer to make
my pipeline to work? I tried to make use of “appsrc” plugin. Still I
could not find the way. Example is give at ../gst-plugins-bad-0.10.6/examples/app/appsrc_ex.c Basically my aim to avoid using filesrc
and give my own GstBuffer to the next element. (gst-launch filesrc
location=MPEG25_11_32_m.mp3 ! flump3dec ! alsasink ) The tried the following way in the given
example testapplication. #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <gst/gst.h> #include <gst/app/gstappsrc.h> #include <gst/app/gstappbuffer.h> #include <gst/app/gstappsink.h> #include <stdio.h> #include <string.h> #include <stdlib.h> typedef struct _App App; struct _App { GstElement *pipe; GstElement *src; GstElement *id; GstElement *mp3dec; GstElement *alsa; }; App s_app; static void dont_eat_my_chicken_wings
(void *priv); int main (int argc, char *argv[]) { App *app = &s_app; int chars_read,i=0; FILE *fp; int pos, sizeoffile; GstBuffer *buf; void *data; gst_init (&argc, &argv); app->pipe = gst_pipeline_new (NULL); g_assert (app->pipe); app->src = gst_element_factory_make
("appsrc", NULL); g_assert (app->src); gst_bin_add (GST_BIN (app->pipe),
app->src); app->mp3dec =
gst_element_factory_make ("flump3dec", NULL); g_assert (app->mp3dec); gst_bin_add (GST_BIN (app->pipe),
app->mp3dec); app->alsa = gst_element_factory_make
("alsasink", NULL); g_assert (app->alsa); gst_bin_add (GST_BIN (app->pipe),
app->alsa); app->id = gst_element_factory_make
("identity", NULL); g_assert (app->id); gst_bin_add (GST_BIN (app->pipe),
app->id); gst_element_link (app->src,
app->id); gst_element_link (app->id,
app->mp3dec); gst_element_link (app->mp3dec,
app->alsa); gst_element_set_state (app->pipe,
GST_STATE_PLAYING);
fp=fopen("MPEG25_11_32_m.mp3","rb"); pos = ftell(fp); fseek (fp, 0, SEEK_END); sizeoffile = ftell(fp); fseek (fp, pos, SEEK_SET); printf("sizeoffile :
%d\n",sizeoffile); data = (char*)malloc (sizeoffile); chars_read =
fread(data,sizeof(char),sizeoffile,fp); buf = gst_app_buffer_new (data,
sizeoffile, dont_eat_my_chicken_wings, data); gst_app_src_push_buffer (GST_APP_SRC
(app->src), buf); gst_app_src_end_of_stream (GST_APP_SRC
(app->src)); gst_buffer_unref (buf); fclose(fp); } static void dont_eat_my_chicken_wings (void *priv) { printf ("freeing buffer for pointer
%p\n", priv); free (priv); }
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This is basically manually adding or
removing data from/to a pipeline. http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-data-spoof.html From: [hidden email]
[mailto:[hidden email]] On Behalf Of Ramana Reddy Polaka Hi, How can I access external buffer to make
my pipeline to work? I tried to make use of “appsrc” plugin. Still I could
not find the way. Example is give at ../gst-plugins-bad-0.10.6/examples/app/appsrc_ex.c Basically my aim to avoid using filesrc
and give my own GstBuffer to the next element. (gst-launch filesrc
location=MPEG25_11_32_m.mp3 ! flump3dec ! alsasink )
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In reply to this post by Ramana_polaka
Hi,
I can run the same pipelines in loopback. To test it live can someone please tell me the settings for UDPSRC. Server: gst-launch filesrc location=MPEG25_11_32_m.mp3 ! mp3parse ! udpsink host=192.168.15.174 port=5000 Clinet: gst-launch udpsrc uri= ????? port=5000 ! typefind ! ffdec_mp3 ! alsasink BR Manish On Fri, Mar 28, 2008 at 10:24 AM, Ramana Reddy Polaka <[hidden email]> wrote:
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On Thu, 2008-05-08 at 19:25 +0530, Manish Rana wrote:
> Hi, > > I can run the same pipelines in loopback. > To test it live can someone please tell me the settings for UDPSRC. Unless you are receiving UDP from a multicast group, you don't need to set anything for the uri in udpsrc, just the port number to listen on is enough. Wim > > Server: gst-launch filesrc location=MPEG25_11_32_m.mp3 ! > mp3parse ! udpsink host=192.168.15.174 port=5000 > > Clinet: gst-launch udpsrc uri= ????? port=5000 ! typefind ! > ffdec_mp3 ! alsasink > > BR > Manish > > > > On Fri, Mar 28, 2008 at 10:24 AM, Ramana Reddy Polaka > <[hidden email]> wrote: > Thank you. The following works fine > > > > Server: gst-launch filesrc location=MPEG25_11_32_m.mp3 ! > mp3parse ! udpsink host=192.168.15.174 port=5000 > > Clinet: gst-launch udpsrc port=5000 ! typefind ! > ffdec_mp3 ! alsasink > > > ______________________________________________________________ > From:[hidden email] > [mailto:[hidden email]] On > Behalf Of thiagoss > Sent: Thursday, March 27, 2008 10:04 PM > To: Daniel Díaz > Cc: [hidden email]; Ramana Reddy Polaka > > > Subject: Re: [gst-devel] Network streaming using udpsrc and > udpsink > > > Try this: > > > gst-launch filesrc location=MPEG25_11_32_m.mp3 ! mp3parse ! > udpsink host=192.168.15.174 port=5000 > > > > gst-launch udpsrc port=5000 ! typefind ! ffdec_mp3 ! > alsasink > > Setting to sync false will make udpsink push too much data, > and the other process won't be able to receive that fast, > probably most of them may not arrive. And using mp3parse puts > timestamps on the buffers, that makes it possible to udpsink > to send buffers on the right time. > > and on the receiver ffdec_mp3 requires that it receives "mp3" > streams, already with a type, udpsrc provides "any", so > typefind would help. > > Hope it works. > > Thiago > > > On Thu, Mar 27, 2008 at 3:01 AM, Daniel Díaz > <[hidden email]> wrote: > > Hello, Ramana > > > On Wed, Mar 26, 2008 at 10:46 PM, Ramana Reddy Polaka > > > > < wrote: > > > > Tried the following also. But did not work. > > > > Receiver (Rx) > > gst-launch filesrc location=MPEG25_11_32_m.mp3 ! udpsink > host=192.168.15.174 > > port=5000 sync=false > > > > Transmitter (Tx) > > gst-launch udpsrc port=5000 ! ffdec_mp3 ! alsasink > > > Do you have to use UDP? You might want to try with > tcpserversrc/tcpclientsink. > > Greetings! > > Daniel Díaz > [hidden email] > > > > > > Regards, > > Ramana > > > > ________________________________ > > > > > > From: thiagoss [mailto:[hidden email]] > > Sent: Wednesday, March 26, 2008 8:40 PM > > To: Ramana Reddy Polaka > > Subject: Re: [gst-devel] Network streaming using udpsrc and > udpsink > > > > > > > > Probably removing the filesink and putting "mad ! alsasink" > > > > > > > > > > On Wed, Mar 26, 2008 at 3:09 AM, Ramana Reddy Polaka wrote: > > > > > > > > > > > > Hi, > > > > > > > > Using the following gst-launch command, I could dump mp3 > data. But I want to > > play it live. Can you please let me know how to achieve > that? Say my host > > IP: 192.168.15.174 > > > > > > > > Receiver (Rx): > > > > gst-launch -v udpsrc ! filesink location=dump.mp3 > > > > > > > > Transmitter (Tx): > > > > gst-launch -v filesrc location=MPEG25_11_32_m.mp3 ! udpsink > > > > > > > > > > > > > > > > Thanks in advance > > > > > > > > Regards, > > > > Ramana > > > > **************** CAUTION - Disclaimer ***************** > This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION > intended solely for the use of the addressee(s). If you are > not the intended recipient, please notify the sender by e-mail > and delete the original message. Further, you are not to copy, > disclose, or distribute this e-mail or its contents to any > other person and any such actions are unlawful. This e-mail > may contain viruses. Infosys has taken every reasonable > precaution to minimize this risk, but is not liable for any > damage you may sustain as a result of any virus in this > e-mail. You should carry out your own virus checks before > opening the e-mail or attachment. Infosys reserves the right > to monitor and review the content of all messages sent to or > from this e-mail address. Messages sent to or from this e-mail > address may be stored on the Infosys e-mail system. > ***INFOSYS******** End of Disclaimer ********INFOSYS*** > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Manish Rana > Senior Software Engg. > Handset Technology Solution, > Sasken Comm. Tech. > Bangalore,India > ------------------------------------------------------------------------- > This SF.net email is sponsored by the 2008 JavaOne(SM) Conference > Don't miss this year's exciting event. There's still time to save $100. > Use priority code J8TL2D2. > http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------- This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi,
Thanks a lot Wim. I am able to run and get the RTP data. But when i try getting RTCP data, i am not able to get it even in loop-back. "tcpdump -T rtcp" doesn't give ant RTCP packet. Though it is printing me some UDP and TCP packets ( may be rtcp but i am not too sure.) "as tcpdump -T rtp" clearly lists me the udp/rtpv1 packets. Also i need to get the RTCP data from Gstreamer, how can i get it??? Can i dump these RTCP packets in some filesink ??? Please Help.... Thanks and Regards Manish On Thu, May 8, 2008 at 7:37 PM, Wim Taymans <[hidden email]> wrote:
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On Tue, 2008-05-13 at 13:05 +0530, Manish Rana wrote:
> Hi, > > Thanks a lot Wim. > I am able to run and get the RTP data. > > But when i try getting RTCP data, i am not able to get it even in > loop-back. For RTCP you need to use gstrtpbin. There are some example pipelines here to get you started: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html Wim > "tcpdump -T rtcp" doesn't give ant RTCP packet. > Though it is printing me some UDP and TCP packets ( may be rtcp but i > am not too sure.) > "as tcpdump -T rtp" clearly lists me the udp/rtpv1 packets. > > Also i need to get the RTCP data from Gstreamer, how can i get it??? > Can i dump these RTCP packets in some filesink ??? > > Please Help.... > > Thanks and Regards > Manish > > > > On Thu, May 8, 2008 at 7:37 PM, Wim Taymans <[hidden email]> > wrote: > On Thu, 2008-05-08 at 19:25 +0530, Manish Rana wrote: > > Hi, > > > > I can run the same pipelines in loopback. > > To test it live can someone please tell me the settings for > UDPSRC. > > > Unless you are receiving UDP from a multicast group, you don't > need to > set anything for the uri in udpsrc, just the port number to > listen on is > enough. > > Wim > > > > > Server: gst-launch filesrc location=MPEG25_11_32_m.mp3 ! > > mp3parse ! udpsink host=192.168.15.174 port=5000 > > > > Clinet: gst-launch udpsrc uri= ????? port=5000 ! > typefind ! > > ffdec_mp3 ! alsasink > > > > BR > > Manish > > > > > > > > On Fri, Mar 28, 2008 at 10:24 AM, Ramana Reddy Polaka > > <[hidden email]> wrote: > > Thank you. The following works fine > > > > > > > > Server: gst-launch filesrc > location=MPEG25_11_32_m.mp3 ! > > mp3parse ! udpsink host=192.168.15.174 port=5000 > > > > Clinet: gst-launch udpsrc port=5000 ! typefind ! > > ffdec_mp3 ! alsasink > > > > > > > ______________________________________________________________ > > From:[hidden email] > > > [mailto:[hidden email]] On > > Behalf Of thiagoss > > Sent: Thursday, March 27, 2008 10:04 PM > > To: Daniel Díaz > > Cc: [hidden email]; Ramana > Reddy Polaka > > > > > > Subject: Re: [gst-devel] Network streaming using > udpsrc and > > udpsink > > > > > > Try this: > > > > > gst-launch filesrc location=MPEG25_11_32_m.mp3 ! > mp3parse ! > > udpsink host=192.168.15.174 port=5000 > > > > > > gst-launch udpsrc port=5000 ! typefind ! > ffdec_mp3 ! > > alsasink > > > > Setting to sync false will make udpsink push too > much data, > > and the other process won't be able to receive that > fast, > > probably most of them may not arrive. And using > mp3parse puts > > timestamps on the buffers, that makes it possible to > udpsink > > to send buffers on the right time. > > > > and on the receiver ffdec_mp3 requires that it > receives "mp3" > > streams, already with a type, udpsrc provides "any", > so > > typefind would help. > > > > Hope it works. > > > > Thiago > > > > > > On Thu, Mar 27, 2008 at 3:01 AM, Daniel Díaz > > <[hidden email]> wrote: > > > > Hello, Ramana > > > > > > On Wed, Mar 26, 2008 at 10:46 PM, Ramana Reddy > Polaka > > > > > > > > < wrote: > > > > > > > Tried the following also. But did not work. > > > > > > Receiver (Rx) > > > gst-launch filesrc location=MPEG25_11_32_m.mp3 ! > udpsink > > host=192.168.15.174 > > > port=5000 sync=false > > > > > > Transmitter (Tx) > > > gst-launch udpsrc port=5000 ! ffdec_mp3 ! alsasink > > > > > > Do you have to use UDP? You might want to try with > > tcpserversrc/tcpclientsink. > > > > Greetings! > > > > Daniel Díaz > > [hidden email] > > > > > > > > > > > Regards, > > > Ramana > > > > > > ________________________________ > > > > > > > > > From: thiagoss [mailto:[hidden email]] > > > Sent: Wednesday, March 26, 2008 8:40 PM > > > To: Ramana Reddy Polaka > > > Subject: Re: [gst-devel] Network streaming using > udpsrc and > > udpsink > > > > > > > > > > > > Probably removing the filesink and putting "mad ! > alsasink" > > > > > > > > > > > > > > > On Wed, Mar 26, 2008 at 3:09 AM, Ramana Reddy > Polaka wrote: > > > > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > Using the following gst-launch command, I could > dump mp3 > > data. But I want to > > > play it live. Can you please let me know how to > achieve > > that? Say my host > > > IP: 192.168.15.174 > > > > > > > > > > > > Receiver (Rx): > > > > > > gst-launch -v udpsrc ! filesink location=dump.mp3 > > > > > > > > > > > > Transmitter (Tx): > > > > > > gst-launch -v filesrc > location=MPEG25_11_32_m.mp3 ! udpsink > > > > > > > > > > > > > > > > > > > > > > > > Thanks in advance > > > > > > > > > > > > Regards, > > > > > > Ramana > > > > > > > > **************** CAUTION - Disclaimer > ***************** > > This e-mail contains PRIVILEGED AND CONFIDENTIAL > INFORMATION > > intended solely for the use of the addressee(s). If > you are > > not the intended recipient, please notify the sender > by e-mail > > and delete the original message. Further, you are > not to copy, > > disclose, or distribute this e-mail or its contents > to any > > other person and any such actions are unlawful. This > > may contain viruses. Infosys has taken every > reasonable > > precaution to minimize this risk, but is not liable > for any > > damage you may sustain as a result of any virus in > this > > e-mail. You should carry out your own virus checks > before > > opening the e-mail or attachment. Infosys reserves > the right > > to monitor and review the content of all messages > sent to or > > from this e-mail address. Messages sent to or from > this e-mail > > address may be stored on the Infosys e-mail system. > > ***INFOSYS******** End of Disclaimer > ********INFOSYS*** > > > > > > > ------------------------------------------------------------------------- > > Check out the new SourceForge.net Marketplace. > > It's the best place to buy or sell services for > > just about anything Open Source. > > > http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Manish Rana > > Senior Software Engg. > > Handset Technology Solution, > > Sasken Comm. Tech. > > Bangalore,India > > > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by the 2008 JavaOne(SM) > Conference > > Don't miss this year's exciting event. There's still time to > save $100. > > Use priority code J8TL2D2. > > > http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi Wim,
Thanks a lot for your prompt replies :) I am using same pipelines, but still i am not sure whether i am getting RTCP data or not that is why i used tcpdump. Also i want the the RTCP data in the Gstreamer application. Thanks n Regds Manish On Tue, May 13, 2008 at 1:47 PM, Wim Taymans <[hidden email]> wrote:
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On Tue, 2008-05-13 at 14:12 +0530, Manish Rana wrote:
> Hi Wim, > > Thanks a lot for your prompt replies :) > I am using same pipelines, but still i am not sure whether i am > getting RTCP data or not that is why i used tcpdump. > Also i want the the RTCP data in the Gstreamer application. Wireshark is a lot easier to check network activity nowadays. What are you going to do with the raw RTCP packets in the application? Wim > > Thanks n Regds > Manish > > On Tue, May 13, 2008 at 1:47 PM, Wim Taymans <[hidden email]> > wrote: > On Tue, 2008-05-13 at 13:05 +0530, Manish Rana wrote: > > Hi, > > > > > Thanks a lot Wim. > > I am able to run and get the RTP data. > > > > But when i try getting RTCP data, i am not able to get it > even in > > loop-back. > > > For RTCP you need to use gstrtpbin. There are some example > pipelines > here to get you started: > > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html > > Wim > > > > > "tcpdump -T rtcp" doesn't give ant RTCP packet. > > Though it is printing me some UDP and TCP packets ( may be > rtcp but i > > am not too sure.) > > "as tcpdump -T rtp" clearly lists me the udp/rtpv1 packets. > > > > Also i need to get the RTCP data from Gstreamer, how can i > get it??? > > Can i dump these RTCP packets in some filesink ??? > > > > Please Help.... > > > > Thanks and Regards > > Manish > > > > > > > > On Thu, May 8, 2008 at 7:37 PM, Wim Taymans > <[hidden email]> > > wrote: > > On Thu, 2008-05-08 at 19:25 +0530, Manish Rana > wrote: > > > Hi, > > > > > > I can run the same pipelines in loopback. > > > To test it live can someone please tell me the > settings for > > UDPSRC. > > > > > > Unless you are receiving UDP from a multicast group, > you don't > > need to > > set anything for the uri in udpsrc, just the port > number to > > listen on is > > enough. > > > > Wim > > > > > > > > Server: gst-launch filesrc > location=MPEG25_11_32_m.mp3 ! > > > mp3parse ! udpsink host=192.168.15.174 port=5000 > > > > > > Clinet: gst-launch udpsrc uri= ????? > port=5000 ! > > typefind ! > > > ffdec_mp3 ! alsasink > > > > > > BR > > > Manish > > > > > > > > > > > > On Fri, Mar 28, 2008 at 10:24 AM, Ramana Reddy > Polaka > > > <[hidden email]> wrote: > > > Thank you. The following works fine > > > > > > > > > > > > Server: gst-launch filesrc > > location=MPEG25_11_32_m.mp3 ! > > > mp3parse ! udpsink host=192.168.15.174 > port=5000 > > > > > > Clinet: gst-launch udpsrc port=5000 ! > typefind ! > > > ffdec_mp3 ! alsasink > > > > > > > > > > > > ______________________________________________________________ > > > > From:[hidden email] > > > > > > [mailto:[hidden email]] On > > > Behalf Of thiagoss > > > Sent: Thursday, March 27, 2008 10:04 PM > > > To: Daniel Díaz > > > Cc: [hidden email]; > Ramana > > Reddy Polaka > > > > > > > > > Subject: Re: [gst-devel] Network streaming > using > > udpsrc and > > > udpsink > > > > > > > > > Try this: > > > > > > > gst-launch filesrc > location=MPEG25_11_32_m.mp3 ! > > mp3parse ! > > > udpsink host=192.168.15.174 port=5000 > > > > > > > > gst-launch udpsrc port=5000 ! typefind ! > > ffdec_mp3 ! > > > alsasink > > > > > > Setting to sync false will make udpsink > push too > > much data, > > > and the other process won't be able to > receive that > > fast, > > > probably most of them may not arrive. And > using > > mp3parse puts > > > timestamps on the buffers, that makes it > possible to > > udpsink > > > to send buffers on the right time. > > > > > > and on the receiver ffdec_mp3 requires > that it > > receives "mp3" > > > streams, already with a type, udpsrc > provides "any", > > so > > > typefind would help. > > > > > > Hope it works. > > > > > > Thiago > > > > > > > > > On Thu, Mar 27, 2008 at 3:01 AM, Daniel > Díaz > > > <[hidden email]> wrote: > > > > > > Hello, Ramana > > > > > > > > > On Wed, Mar 26, 2008 at 10:46 PM, Ramana > Reddy > > Polaka > > > > > > > > > > > > < wrote: > > > > > > > > > > Tried the following also. But did not > work. > > > > > > > > Receiver (Rx) > > > > gst-launch filesrc > location=MPEG25_11_32_m.mp3 ! > > udpsink > > > host=192.168.15.174 > > > > port=5000 sync=false > > > > > > > > Transmitter (Tx) > > > > gst-launch udpsrc port=5000 ! > ffdec_mp3 ! alsasink > > > > > > > > > Do you have to use UDP? You might want to > try with > > > tcpserversrc/tcpclientsink. > > > > > > Greetings! > > > > > > Daniel Díaz > > > [hidden email] > > > > > > > > > > > > > > > > Regards, > > > > Ramana > > > > > > > > ________________________________ > > > > > > > > > > > > From: thiagoss > [mailto:[hidden email]] > > > > Sent: Wednesday, March 26, 2008 8:40 PM > > > > To: Ramana Reddy Polaka > > > > Subject: Re: [gst-devel] Network > streaming using > > udpsrc and > > > udpsink > > > > > > > > > > > > > > > > Probably removing the filesink and > putting "mad ! > > alsasink" > > > > > > > > > > > > > > > > > > > > On Wed, Mar 26, 2008 at 3:09 AM, Ramana > Reddy > > Polaka wrote: > > > > > > > > > > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > > > Using the following gst-launch command, > I could > > dump mp3 > > > data. But I want to > > > > play it live. Can you please let me know > how to > > achieve > > > that? Say my host > > > > IP: 192.168.15.174 > > > > > > > > > > > > > > > > Receiver (Rx): > > > > > > > > gst-launch -v udpsrc ! filesink > location=dump.mp3 > > > > > > > > > > > > > > > > Transmitter (Tx): > > > > > > > > gst-launch -v filesrc > > location=MPEG25_11_32_m.mp3 ! udpsink > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Thanks in advance > > > > > > > > > > > > > > > > Regards, > > > > > > > > Ramana > > > > > > > > > > > > **************** CAUTION - Disclaimer > > ***************** > > > This e-mail contains PRIVILEGED AND > CONFIDENTIAL > > INFORMATION > > > intended solely for the use of the > addressee(s). If > > you are > > > not the intended recipient, please notify > the sender > > by e-mail > > > and delete the original message. Further, > you are > > not to copy, > > > disclose, or distribute this e-mail or its > contents > > to any > > > other person and any such actions are > unlawful. This > > > may contain viruses. 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I used WireShark also.It was giving me only UDP Packets.
I wat to decide in Application e.g Packet loss is huge then i can modify the JitterBuffer settings or if the Bandwidth is too low i can reduce the bitrate. So that way it will be helpful. On Tue, May 13, 2008 at 2:24 PM, Wim Taymans <[hidden email]> wrote:
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