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2009/5/11 <[hidden email]>
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Today's Topics:
1. Re: [gst-cvs] gst-plugins-base: theoradec: rearrange code in
preparation for 422 and 444 support. (Michael Smith)
2. Gstreamer contractor needed (Media Bu)
3. Re: rtspsrc on Windows (Michael Smith)
4. Re: Problem using gstrtpbin (Wim Taymans)
----------------------------------------------------------------------
Message: 1
Date: Mon, 11 May 2009 10:21:25 -0700
From: Michael Smith <[hidden email]>
Subject: Re: [gst-devel] [gst-cvs] gst-plugins-base: theoradec:
rearrange code in preparation for 422 and 444 support.
To: Jan Schmidt <[hidden email]>
Cc: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
Content-Type: text/plain; charset=ISO-8859-1
On Mon, May 11, 2009 at 9:54 AM, Jan Schmidt <[hidden email]> wrote:
> I'm unfreezing in a few hours anyway, so might as well leave them in.
> I'll try and make sure to leave the freeze notice in the channel topic
> next time.
Ok. Sorry for the screwup.
>
> For the record, the actual rule is that things unfreeze when I push the
> commit that bumps the version number in configure.ac back to
> development.
>
> The consensus seems to be that there's nothing wrong with our current
> practice apart from me forgetting to leave the IRC topic marked frozen,
> so that's cool.
The current practice is ok with me.
I think it'd be slightly better to branch _at release time_ (but
retain the freeze on trunk before the release). Brown-paper-bag
releases, as well as future security releases if required, would
happen from the branch. No post-release freeze would be required. So
if we were to change anything, that's what I'd suggest.
Mike
------------------------------
Message: 2
Date: Mon, 11 May 2009 10:25:19 -0700
From: Media Bu <[hidden email]>
Subject: [gst-devel] Gstreamer contractor needed
To: <[hidden email]>
Message-ID: <[hidden email]>
Content-Type: text/plain; charset="iso-8859-1"
2wire, Inc., the leader in the development of consumer electronics software for the next-generation of media platforms, has an outstanding consulting opportunity as a GStreamer Expert in San Jose, CA.
We are looking for a consultant with a successful track record and expertise in the following areas:
GStreamer development for audio & video streaming application
Set-top box development, experience working with AV decoding Soc
DVR, trick-play, time-shifting TV recording
AV standards: MPEG-2 transport, ASF, MP4, WMV, H.264, WMA, AC3
Live TV, tuner, ATSC standard
Required Core Skills:
Extensive Gstreamer application and plugin development experience
HTTP/RTP/RTSP/RTCP
Linux platform development
Extensive experience working with open source
Successful delivery of multimedia product
Please reply with resume and/or background summary of experience/expertise along with expected bill rate.
_________________________________________________________________
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Message: 3
Date: Mon, 11 May 2009 10:31:48 -0700
From: Michael Smith <[hidden email]>
Subject: Re: [gst-devel] rtspsrc on Windows
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
Content-Type: text/plain; charset=ISO-8859-1
2009/5/11 Marl?ne Hildebrand-Ehrhardt <[hidden email]>:
> Hi,
>
> I have a problem : I am trying to launch a rtsp server on Windows with
> gst-launch, but everytime, it crashes after a few minutes. The console sends
> me a warning, indicating that it could not read from source, and then it
> tells me that the server closed the connection and that it got an EOS from
> the element "pipeline0".
Did you write an rtsp server as a gstreamer element? GStreamer doesn't
come with anything to do RTSP serving that you could run with
gst-launch.
Please provide details about what you're _actually_ doing.
Also mention which versions you're using (and which version of windows).
Mike
Thank you for answering me, Mike.
I'm using gst-launch-0.10, version 0.10.22 with Gstreamer 0.10.22. I'm working on Windows XP professionnal, version 2002, SP2.
In fact, I'm building a rtsp server with the following command line :
gst-launch-0.10.exe -v rtspsrc location=rtsp://(IP adress of my IP camera)/mpeg4/media.amp ! queue ! decodebin ! queue ! videorate ! capsfilter caps="video/x-raw-yuv" ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src ! udpsink port=5000 host=(IP adress of the client) session.send_rtcp_src ! udpsink port=5001 host=(IP adress of the client)
My aim is to stream the video from an IP camera.
Don't hesitate to ask to provide more details, if necessary! Plus, I'm french, so excuse-me if I might not immediatly understand what you are asking me.
Again, thank you by advance!
Marlène
------------------------------
Message: 4
Date: Mon, 11 May 2009 19:58:59 +0200
From: Wim Taymans <[hidden email]>
Subject: Re: [gst-devel] Problem using gstrtpbin
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID: <1242064739.4809.166.camel@metal>
Content-Type: text/plain; charset="UTF-8"
On Mon, 2009-05-11 at 14:19 -0300, Tiago Katcipis wrote:
> but the ssrc is being created, why does the pad not? i get all the
> msgs the pipeline sends and print them all and i get no error, i
> already made a test with just a udpsrc ! filesink and the data is
> comming.
Because those signals are emitted when a source is created in the
session, which happens before the various pt/ssrc demuxers add the pad
to gstrtpbin.
Wim
>
> On Mon, May 11, 2009 at 2:06 PM, Wim Taymans <[hidden email]>
> wrote:
> On Mon, 2009-05-11 at 13:55 -0300, Tiago Katcipis wrote:
> > i still didnt the teste, but i found this
> >
>
>
>
> Sometimes (dynamic pads) on elements *always* get notified to
> you with
> the "pad-added" callback. This is a signal on GstElement. You
> will get
> the newly created pads from gstrtpbin with this signal. The
> other
> signals you are trying to connect to are for something else.
>
> Wim
>
>
> >
> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpssrcdemux.html#GstRtpSsrcDemux-new-ssrc-pad
> >
> > the GstRtpSsrcDemux sends the signal, i tried to get the
> signal from
> > the bin with:
> >
> > g_signal_connect (rtp_bin, "new-ssrc-pad", G_CALLBACK
> > (on_new_ssrc_pad), rtp_decoder);
> >
> > but i didnt got the signal.
> >
> > My question is, i can get the signal from an element inside
> the bin
> > from the bin or i have to retrieve the element from the bin
> and then
> > connect? and if i have to get the element from the bin (in
> this case
> > the GstRtpSsrcDemux), how is the cleanest way of doing
> this?
> >
> > thanks for all the help.
> > Best regards,
> > Katcipis
> >
> > here goes the src:
> >
> > #include <gst/gst.h>
> > #include <glib.h>
> >
> > #define PORTA_UDP_ENTRADA 5000
> >
> > static gboolean
> > bus_call (GstBus *bus,
> > GstMessage *msg,
> > gpointer data)
> > {
> > GMainLoop *loop = (GMainLoop *) data;
> >
> > switch (GST_MESSAGE_TYPE (msg)) {
> >
> > case GST_MESSAGE_EOS:
> > g_print ("End of stream\n");
> > g_main_loop_quit (loop);
> > break;
> >
> > case GST_MESSAGE_ERROR: {
> > gchar *debug;
> > GError *error;
> >
> > gst_message_parse_error (msg, &error, &debug);
> > g_free (debug);
> >
> > g_printerr ("Error: %s\n", error->message);
> > g_error_free (error);
> >
> > g_main_loop_quit (loop);
> > break;
> > }
> > default:
> > g_print("Tipo da mensagem [%d], Nome da mensagem [%
> s]\n",
> > GST_MESSAGE_TYPE (msg), GST_MESSAGE_TYPE_NAME(msg));
> > break;
> > }
> >
> > return TRUE;
> > }
> >
> >
> > static void
> > on_new_ssrc (GstElement* gstrtpbin,
> > guint session,
> > guint ssrc,
> > gpointer data)
> > {
> > GstPad* sinkpad;
> > GstPad* srcpad[1];
> > GstElement* decoder = (GstElement *) data;
> > GstIterator* iter;
> > gint done, linked, iter_count;
> >
> > g_print ("New session stabilished, linking gstrtpbin
> session src pad
> > to the rtp_decoder\n");
> >
> > sinkpad = gst_element_get_static_pad(decoder, "sink");
> > // TODO Esta dificil de pegar o pad src do gstrtpbin que
> eh criado
> > ao iniciar uma sessao nova.
> > if(!sinkpad){
> > g_warning("Error getting rtp_decoder sink pad");
> > return;
> > }
> > /*
> > unique pad recv_rtp_src_%d_%d_%d on gstrtpbin with the
> session
> > number, SSRC and payload type respectively as the pad name.
> >
> >
> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html
> > */
> >
> > iter = gst_element_iterate_src_pads(gstrtpbin);
> > if(!iter){
> > g_warning("Error getting gstrtpbin pads iterator");
> > return;
> > }
> >
> > done = FALSE;
> > linked = FALSE;
> > iter_count = 0;
> >
> > while (!done) {
> > switch (gst_iterator_next (iter, (gpointer *) srcpad))
> {
> > case GST_ITERATOR_OK:
> > if(gst_pad_link (*srcpad, sinkpad) !=
> GST_PAD_LINK_OK){
> > g_warning("Error linking gstrtpbin pad[%s]
> to
> > rtp_decoder pad[%s]", gst_pad_get_name(*srcpad),
> > gst_pad_get_name(sinkpad));
> > }else{
> > g_warning("Linked gstrtpbin pad[%s] to
> rtp_decoder
> > pad[%s] with success", gst_pad_get_name(*srcpad),
> > gst_pad_get_name(sinkpad));
> > linked = TRUE;
> > }
> > iter_count++;
> > gst_object_unref (*srcpad);
> > break;
> > case GST_ITERATOR_RESYNC:
> > gst_iterator_resync (iter);
> > break;
> > case GST_ITERATOR_ERROR:
> > done = TRUE;
> > break;
> > case GST_ITERATOR_DONE:
> > done = TRUE;
> > break;
> > }
> > }
> > if(!linked){
> > g_warning("failed to found a valid recv_src_pad on
> gstrtpbin");
> > }
> > g_debug("GstRtpBin has [%d] src pads", iter_count);
> >
> > gst_iterator_free (iter);
> > gst_object_unref (sinkpad);
> > }
> >
> > static void
> > on_new_ssrc_pad (GstElement *element,
> > GstPad *pad,
> > guint ssrc,
> > gpointer data)
> > {
> > GstPad *sinkpad;
> > GstElement *decoder = (GstElement *) data;
> >
> > /* We can now link this pad with the converter sink pad */
> > g_print ("Dynamic ssrc pad created, linking the pad to the
> > rtp_decoder\n");
> >
> > sinkpad = gst_element_get_static_pad (decoder, "sink");
> > if(gst_pad_link (pad, sinkpad) != GST_PAD_LINK_OK){
> > gchar* name = gst_pad_get_name(pad);
> > g_warning("Error linking the pad[%s] to rtp_decoder
> sinkpad",
> > name);
> > g_free(name);
> > }
> >
> > gst_object_unref (sinkpad);
> > }
> >
> > int
> > main (int argc,
> > char *argv[])
> > {
> > GMainLoop *loop;
> >
> > GstElement *pipeline, *source, *rtp_bin, *rtp_decoder,
> *sink;
> > GstPad *gstrtp_sink_pad;
> > GstBus *bus;
> >
> > /* Initialisation */
> > gst_init (&argc, &argv);
> >
> > loop = g_main_loop_new (NULL, FALSE);
> >
> > /* Create gstreamer elements */
> > pipeline = gst_pipeline_new ("audio-player");
> > source = gst_element_factory_make
> ("udpsrc","udp-source");
> > rtp_bin = gst_element_factory_make ("gstrtpbin",
> "gst_rtpbin");
> > rtp_decoder = gst_element_factory_make ("rtpL16depay",
> > "rtp_decoder");
> > sink = gst_element_factory_make ("filesink",
> "file-sink");
> >
> > if (!pipeline || !source || !sink || !rtp_decoder || !
> rtp_bin) {
> > g_printerr ("One element could not be created.
> Exiting.\n");
> > return -1;
> > }
> >
> > gstrtp_sink_pad = gst_element_get_request_pad(rtp_bin,
> > "recv_rtp_sink_0");
> > if (!gstrtp_sink_pad) {
> > g_printerr ("Sink pad could not be created.
> Exiting.\n");
> > return -1;
> > }
> >
> > /* Set up the pipeline */
> > g_object_set (G_OBJECT (source), "port",
> PORTA_UDP_ENTRADA , NULL);
> > g_object_set (G_OBJECT (sink), "location",
> "dados_recebidos_rtp" ,
> > NULL);
> >
> > /* we add a message handler */
> > bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
> > gst_bus_add_watch (bus, bus_call, loop);
> > gst_object_unref (bus);
> >
> > /* we add all elements into the pipeline */
> > /* file-source | ogg-demuxer | vorbis-decoder | converter
> |
> > alsa-output */
> > gst_bin_add_many (GST_BIN (pipeline),
> > source, sink, rtp_bin, rtp_decoder,
> NULL);
> >
> > /* we link the elements together */
> > if(gst_pad_link(gst_element_get_static_pad(source, "src"),
> > gstrtp_sink_pad) != GST_PAD_LINK_OK){
> > g_warning("Error linking source to the
> gstrtp_sink_pad");
> > gst_object_unref (GST_OBJECT (pipeline));
> > return 0;
> > }
> >
> > /*
> > After the packets are released from the jitterbuffer,
> they will be
> > forwarded to a GstRtpsSrcDemux element.
> > The GstRtpsSrcDemux element will demux the packets based
> on the
> > payload type and will create a unique pad
> > recv_rtp_src_%d_%d_%d on gstrtpbin with the session
> number, SSRC
> > and payload type respectively as the pad name.
> > Because of that we have to dinamicaly link the src pads
> on
> > runtime.
> > */
> > g_signal_connect (rtp_bin, "new-ssrc-pad", G_CALLBACK
> > (on_new_ssrc_pad), rtp_decoder);
> > g_signal_connect (rtp_bin, "on-new-ssrc", G_CALLBACK
> (on_new_ssrc),
> > rtp_decoder);
> >
> > if(!gst_element_link (rtp_decoder, sink)){
> > g_warning("Error linking the rtp_decoder to the
> sink");
> > gst_object_unref (GST_OBJECT (pipeline));
> > return -1;
> > }
> >
> > /* Set the pipeline to "playing" state*/
> > g_print ("listening on port: %d\n", PORTA_UDP_ENTRADA);
> > gst_element_set_state (pipeline, GST_STATE_PLAYING);
> >
> > /* Iterate */
> > g_print ("Running...\n");
> > g_main_loop_run (loop);
> >
> > /* Out of the main loop, clean up nicely */
> > g_print ("Returned, stopping listening on port\n");
> > gst_element_set_state (pipeline, GST_STATE_NULL);
> >
> > g_print ("Deleting pipeline\n");
> > gst_object_unref (GST_OBJECT (pipeline));
> >
> > return 0;
> > }
> >
> >
> > On Sat, May 9, 2009 at 3:40 PM, Aurelien Grimaud
> <[hidden email]>
> > wrote:
> > Sorry, I misread your code.
> > the pad-added signal is a signal of elements,
> documented in
> > the element
> > documentation.
> > Do you receive RTP ?
> > Because the pad wont be created if you do not
> receive RTP.
> > What does tcpdump tell ?
> >
> > Aurelien
> > Tiago Katcipis a ?crit :
> >
> > > i did it, the pad never is created :-(, but i get
> no message
> > of
> > > warning or error neither. And on the list of
> signals of the
> > gstrtpbin
> > > there is no "pad-added" signal, its normal to the
> signal
> > dont be there?
> > > *
> > > g_signal_connect (rtp_bin, "pad-added",
> G_CALLBACK
> > (on_pad_added),
> > > rtp_decoder);*
> > >
> > > On Sat, May 9, 2009 at 3:55 AM, Aurelien Grimaud
> > <[hidden email]
> >
> >
> > > <mailto:[hidden email]>> wrote:
> > >
> > > You should add the pad-added signal on the
> rtpbin.
> > > When it triggers, check the pad name to find
> out which
> > pad it is.
> > > If pad is a recv_rtp_src_%d_%d_%d, link your
> decoder and
> > sink in the
> > > call back.
> > >
> > > Aurelien
> > >
> > > Tiago Katcipis a ?crit :
> > > > Im trying to do a rtp stream sending data
> and another
> > side receiving
> > > > the data, the part that sends the data is
> working
> > fine, but the part
> > > > that receives is giving me a lot of trouble.
> At
> > > >
> > >
> >
> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html
> > > > i have read:
> > > >
> > > > "To use GstRtpBin
> > > >
> > >
> >
> <http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html#GstRtpBin>
> > > > as an RTP receiver, request a recv_rtp_sink_
> %|d| pad.
> > The session
> > > > number must be specified in the pad name.
> Data
> > received on the
> > > > recv_rtp_sink_%|d| pad will be processed in
> the
> > gstrtpsession
> > > manager
> > > > and after being validated forwarded on
> GstRtpsSrcDemux
> > element. Each
> > > > RTP stream is demuxed based on the SSRC and
> send to a
> > > > GstRtpJitterBuffer
> > > >
> > >
> >
> <http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpjitterbuffer.html#GstRtpJitterBuffer>.
> > > > After the packets are released from the
> jitterbuffer,
> > they will be
> > > > forwarded to a GstRtpsSrcDemux element. The
> > GstRtpsSrcDemux element
> > > > will demux the packets based on the payload
> type and
> > will create a
> > > > unique pad recv_rtp_src_%|d_|%|d_|%|d| on
> gstrtpbin
> > with the session
> > > > number, SSRC and payload type respectively
> as the pad
> > name. "
> > > >
> > > > on my application i cant get the
> recv_rtp_src_%|d_|%|
> > d_|%|d, |i
> > > > already tried on a lot of ways, my last shot
> was try
> > to iterate over
> > > > all the pads on the bin and try to conect, i
> > discovered that the src
> > > > pad never shows up. No error is given. I can
> get the
> > on-new-ssrc
> > > > signal...and other signals as |
> on-ssrc-validated...
> > but on all this
> > > > signals the | recv_rtp_src_%|d_|%|d_|%|d is
> not
> > created yet, i also
> > > > tried to get the "on-pad-added" signal but
> this signal
> > never
> > > happens|.
> > > >
> > > > My problem is, when the recv_rtp_src_%|d_|%|
> d_|%|d is
> > created|.
> > > When i
> > > > iterate over the pads i always get a
> > > > ** (teste_rtp:9516): DEBUG: GstRtpBin has
> [0] src pads
> > > >
> > > > here goes the source code, is a little messy
> because
> > im all day
> > > trying
> > > > a lot of different ways to do this. And i
> get no error
> > message.
> > > >
> > > > #include <gst/gst.h>
> > > > #include <glib.h>
> > > >
> > > > #define PORTA_UDP_ENTRADA 5000
> > > >
> > > > static gboolean
> > > > bus_call (GstBus *bus,
> > > > GstMessage *msg,
> > > > gpointer data)
> > > > {
> > > > GMainLoop *loop = (GMainLoop *) data;
> > > >
> > > > switch (GST_MESSAGE_TYPE (msg)) {
> > > >
> > > > case GST_MESSAGE_EOS:
> > > > g_print ("End of stream\n");
> > > > g_main_loop_quit (loop);
> > > > break;
> > > >
> > > > case GST_MESSAGE_ERROR: {
> > > > gchar *debug;
> > > > GError *error;
> > > >
> > > > gst_message_parse_error (msg, &error,
> &debug);
> > > > g_free (debug);
> > > >
> > > > g_printerr ("Error: %s\n",
> error->message);
> > > > g_error_free (error);
> > > >
> > > > g_main_loop_quit (loop);
> > > > break;
> > > > }
> > > > default:
> > > > g_print("Tipo da mensagem [%d], Nome
> da mensagem
> > [%s]\n",
> > > > GST_MESSAGE_TYPE (msg),
> GST_MESSAGE_TYPE_NAME(msg));
> > > > break;
> > > > }
> > > >
> > > > return TRUE;
> > > > }
> > > >
> > > >
> > > > static void
> > > > on_new_ssrc (GstElement* gstrtpbin,
> > > > guint session,
> > > > guint ssrc,
> > > > gpointer data)
> > > > {
> > > > GstPad* sinkpad;
> > > > GstPad* srcpad[1];
> > > > GstElement* decoder = (GstElement *) data;
> > > > GstIterator* iter;
> > > > gint done, linked, iter_count;
> > > >
> > > > g_print ("New session stabilished, linking
> gstrtpbin
> > session
> > > src pad
> > > > to the rtp_decoder\n");
> > > >
> > > > sinkpad =
> gst_element_get_static_pad(decoder,
> > "sink");
> > > > // TODO Esta dificil de pegar o pad src do
> gstrtpbin
> > que eh criado
> > > > ao iniciar uma sessao nova.
> > > > if(!sinkpad){
> > > > g_warning("Error getting rtp_decoder
> sink pad");
> > > > return;
> > > > }
> > > > /*
> > > > unique pad recv_rtp_src_%d_%d_%d on
> gstrtpbin
> > with the session
> > > > number, SSRC and payload type respectively
> as the pad
> > name.
> > > >
> > > >
> > >
> >
> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html
> > > > */
> > > >
> > > > iter =
> gst_element_iterate_src_pads(gstrtpbin);
> > > > if(!iter){
> > > > g_warning("Error getting gstrtpbin
> pads
> > iterator");
> > > > return;
> > > > }
> > > >
> > > > done = FALSE;
> > > > linked = FALSE;
> > > > iter_count = 0;
> > > >
> > > > while (!done) {
> > > > switch (gst_iterator_next (iter,
> (gpointer *)
> > srcpad)) {
> > > > case GST_ITERATOR_OK:
> > > > if(gst_pad_link (*srcpad,
> sinkpad) !=
> > > GST_PAD_LINK_OK){
> > > > g_warning("Error linking
> gstrtpbin
> > pad[%s] to
> > > > rtp_decoder pad[%s]",
> gst_pad_get_name(*srcpad),
> > > > gst_pad_get_name(sinkpad));
> > > > }else{
> > > > g_warning("Linked
> gstrtpbin pad[%s]
> > to rtp_decoder
> > > > pad[%s] with success",
> gst_pad_get_name(*srcpad),
> > > > gst_pad_get_name(sinkpad));
> > > > linked = TRUE;
> > > > }
> > > > iter_count++;
> > > > gst_object_unref (*srcpad);
> > > > break;
> > > > case GST_ITERATOR_RESYNC:
> > > > gst_iterator_resync (iter);
> > > > break;
> > > > case GST_ITERATOR_ERROR:
> > > > done = TRUE;
> > > > break;
> > > > case GST_ITERATOR_DONE:
> > > > done = TRUE;
> > > > break;
> > > > }
> > > > }
> > > > if(!linked){
> > > > g_warning("failed to found a valid
> recv_src_pad
> > on
> > > gstrtpbin");
> > > > }
> > > > g_debug("GstRtpBin has [%d] src pads",
> iter_count);
> > > >
> > > > gst_iterator_free (iter);
> > > > gst_object_unref (sinkpad);
> > > > }
> > > >
> > > > static void
> > > > on_pad_added (GstElement *element,
> > > > GstPad *pad,
> > > > gpointer data)
> > > > {
> > > > GstPad *sinkpad;
> > > > GstElement *decoder = (GstElement *) data;
> > > >
> > > > /* We can now link this pad with the
> converter sink
> > pad */
> > > > g_print ("Dynamic pad created, linking
> > wavparser/converter\n");
> > > >
> > > > sinkpad = gst_element_get_static_pad
> (decoder,
> > "sink");
> > > > if(gst_pad_link (pad, sinkpad) !=
> GST_PAD_LINK_OK){
> > > > g_warning("Error linking recv_rtp_src
> pad to
> > sinkpad");
> > > > }
> > > > gst_object_unref (sinkpad);
> > > > }
> > > >
> > > > int
> > > > main (int argc,
> > > > char *argv[])
> > > > {
> > > > GMainLoop *loop;
> > > >
> > > > GstElement *pipeline, *source, *rtp_bin,
> > *rtp_decoder, *sink;
> > > > GstPad *gstrtp_sink_pad;
> > > > GstBus *bus;
> > > >
> > > > /* Initialisation */
> > > > gst_init (&argc, &argv);
> > > >
> > > > loop = g_main_loop_new (NULL, FALSE);
> > > >
> > > > /* Create gstreamer elements */
> > > > pipeline = gst_pipeline_new
> ("audio-player");
> > > > source = gst_element_factory_make
> > ("udpsrc","udp-source");
> > > > rtp_bin = gst_element_factory_make
> ("gstrtpbin",
> > > "gst_rtpbin");
> > > > rtp_decoder = gst_element_factory_make
> > ("rtpL16depay",
> > > "rtp_decoder");
> > > > sink = gst_element_factory_make
> ("filesink",
> > "file-sink");
> > > >
> > > > if (!pipeline || !source || !sink || !
> rtp_decoder
> > || !rtp_bin) {
> > > > g_printerr ("One element could not be
> created.
> > Exiting.\n");
> > > > return -1;
> > > > }
> > > >
> > > > gstrtp_sink_pad =
> > gst_element_get_request_pad(rtp_bin,
> > > > "recv_rtp_sink_0");
> > > > if (!gstrtp_sink_pad) {
> > > > g_printerr ("Sink pad could not be
> created.
> > Exiting.\n");
> > > > return -1;
> > > > }
> > > >
> > > > /* Set up the pipeline */
> > > > g_object_set (G_OBJECT (source), "port",
> > PORTA_UDP_ENTRADA ,
> > > NULL);
> > > > g_object_set (G_OBJECT (sink), "location",
> > "dados_recebidos_rtp" ,
> > > > NULL);
> > > >
> > > > /* we add a message handler */
> > > > bus = gst_pipeline_get_bus (GST_PIPELINE
> > (pipeline));
> > > > gst_bus_add_watch (bus, bus_call, loop);
> > > > gst_object_unref (bus);
> > > >
> > > > /* we add all elements into the pipeline
> */
> > > > /* file-source | ogg-demuxer |
> vorbis-decoder |
> > converter |
> > > > alsa-output */
> > > > gst_bin_add_many (GST_BIN (pipeline),
> > > > source, sink, rtp_bin,
> > rtp_decoder, NULL);
> > > >
> > > > /* we link the elements together */
> > > >
> if(gst_pad_link(gst_element_get_static_pad(source,
> > "src"),
> > > > gstrtp_sink_pad) != GST_PAD_LINK_OK){
> > > > g_warning("Error linking source to the
> > gstrtp_sink_pad");
> > > > gst_object_unref (GST_OBJECT
> (pipeline));
> > > > return 0;
> > > > }
> > > >
> > > > /*
> > > > After the packets are released from the
> > jitterbuffer, they
> > > will be
> > > > forwarded to a GstRtpsSrcDemux element.
> > > > The GstRtpsSrcDemux element will demux
> the packets
> > based on the
> > > > payload type and will create a unique pad
> > > > recv_rtp_src_%d_%d_%d on gstrtpbin with
> the
> > session number, SSRC
> > > > and payload type respectively as the pad
> name.
> > > > Because of that we have to dinamicaly
> link the src
> > pads on
> > > runtime.
> > > > */
> > > > g_signal_connect (rtp_bin, "pad-added",
> G_CALLBACK
> > > (on_pad_added),
> > > > rtp_decoder);
> > > > g_signal_connect (rtp_bin, "on-new-ssrc",
> G_CALLBACK
> > > (on_new_ssrc),
> > > > rtp_decoder);
> > > >
> > > > if(!gst_element_link (rtp_decoder, sink)){
> > > > g_warning("Error linking the
> rtp_decoder to the
> > sink");
> > > > gst_object_unref (GST_OBJECT
> (pipeline));
> > > > return -1;
> > > > }
> > > >
> > > > /* Set the pipeline to "playing" state*/
> > > > g_print ("listening on port: %d\n",
> > PORTA_UDP_ENTRADA);
> > > > gst_element_set_state (pipeline,
> GST_STATE_PLAYING);
> > > >
> > > > /* Iterate */
> > > > g_print ("Running...\n");
> > > > g_main_loop_run (loop);
> > > >
> > > > /* Out of the main loop, clean up nicely
> */
> > > > g_print ("Returned, stopping listening on
> port\n");
> > > > gst_element_set_state (pipeline,
> GST_STATE_NULL);
> > > >
> > > > g_print ("Deleting pipeline\n");
> > > > gst_object_unref (GST_OBJECT (pipeline));
> > > >
> > > > return 0;
> > > > }
> > > >
> > > >
> > >
> >
> ------------------------------------------------------------------------
> > > >
> > > >
> > >
> >
> ------------------------------------------------------------------------------
> > > > The NEW KODAK i700 Series Scanners deliver
> under ANY
> > > circumstances! Your
> > > > production scanning environment may not be a
> perfect
> > world - but
> > > thanks to
> > > > Kodak, there's a perfect scanner to get the
> job done!
> > With the
> > > NEW KODAK i700
> > > > Series Scanner you'll get full speed at 300
> dpi even
> > with all image
> > > > processing features enabled.
> > http://p.sf.net/sfu/kodak-com
> > > >
> > >
> >
> ------------------------------------------------------------------------
> > > >
> > > >
> _______________________________________________
> > > > gstreamer-devel mailing list
> > > > [hidden email]
> >
> > > <mailto:[hidden email]>
> > > >
> >
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> > > >
> > >
> > >
> > >
> >
> ------------------------------------------------------------------------------
> > > The NEW KODAK i700 Series Scanners deliver
> under ANY
> > > circumstances! Your
> > > production scanning environment may not be a
> perfect
> > world - but
> > > thanks to
> > > Kodak, there's a perfect scanner to get the
> job done!
> > With the NEW
> > > KODAK i700
> > > Series Scanner you'll get full speed at 300
> dpi even
> > with all image
> > > processing features enabled.
> > http://p.sf.net/sfu/kodak-com
> > >
> _______________________________________________
> > > gstreamer-devel mailing list
> > > [hidden email]
> >
> > > <mailto:[hidden email]>
> >
> > >
> >
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> > >
> > >
> > >
> > >
> > > --
> > > "it might be a profitable thing to learn Java, but
> it has no
> > > intellectual value whatsoever" Alexander Stepanov
> > >
> >
> ------------------------------------------------------------------------
> > >
> > >
> >
> ------------------------------------------------------------------------------
> > > The NEW KODAK i700 Series Scanners deliver under
> ANY
> > circumstances! Your
> > > production scanning environment may not be a
> perfect world -
> > but thanks to
> > > Kodak, there's a perfect scanner to get the job
> done! With
> > the NEW KODAK i700
> > > Series Scanner you'll get full speed at 300 dpi
> even with
> > all image
> > > processing features enabled.
> http://p.sf.net/sfu/kodak-com
> > >
> >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > gstreamer-devel mailing list
> > > [hidden email]
> > >
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> > >
> >
> >
> >
> ------------------------------------------------------------------------------
> > The NEW KODAK i700 Series Scanners deliver under ANY
> > circumstances! Your
> > production scanning environment may not be a perfect
> world -
> > but thanks to
> > Kodak, there's a perfect scanner to get the job
> done! With the
> > NEW KODAK i700
> > Series Scanner you'll get full speed at 300 dpi even
> with all
> > image
> > processing features enabled.
> http://p.sf.net/sfu/kodak-com
> > _______________________________________________
> > gstreamer-devel mailing list
> > [hidden email]
> >
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
> >
> >
> >
> > --
> > "it might be a profitable thing to learn Java, but it has no
> > intellectual value whatsoever" Alexander Stepanov
> >
> ------------------------------------------------------------------------------
> > The NEW KODAK i700 Series Scanners deliver under ANY
> circumstances! Your
> > production scanning environment may not be a perfect world -
> but thanks to
> > Kodak, there's a perfect scanner to get the job done! With
> the NEW KODAK i700
> > Series Scanner you'll get full speed at 300 dpi even with
> all image
> > processing features enabled. http://p.sf.net/sfu/kodak-com
> > _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
> ------------------------------------------------------------------------------
> The NEW KODAK i700 Series Scanners deliver under ANY
> circumstances! Your
> production scanning environment may not be a perfect world -
> but thanks to
> Kodak, there's a perfect scanner to get the job done! With the
> NEW KODAK i700
> Series Scanner you'll get full speed at 300 dpi even with all
> image
> processing features enabled. http://p.sf.net/sfu/kodak-com
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
>
>
> --
> "it might be a profitable thing to learn Java, but it has no
> intellectual value whatsoever" Alexander Stepanov
> ------------------------------------------------------------------------------
> The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your
> production scanning environment may not be a perfect world - but thanks to
> Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700
> Series Scanner you'll get full speed at 300 dpi even with all image
> processing features enabled. http://p.sf.net/sfu/kodak-com
> _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
------------------------------
------------------------------------------------------------------------------
The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your
production scanning environment may not be a perfect world - but thanks to
Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700
Series Scanner you'll get full speed at 300 dpi even with all image
processing features enabled. http://p.sf.net/sfu/kodak-com
------------------------------
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
End of gstreamer-devel Digest, Vol 36, Issue 42
***********************************************
------------------------------------------------------------------------------
The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your
production scanning environment may not be a perfect world - but thanks to
Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700
Series Scanner you'll get full speed at 300 dpi even with all image
processing features enabled. http://p.sf.net/sfu/kodak-com_______________________________________________
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