[gst-devel] problem sending audio/mpeg over RTP

[Antoine Tremblay]

gstrtpmpadepay.c(213) :gst_rtp_mpa_depay_chain: Unexpected
payload type 96

Ok here's your problem :

gst-plugins-base-0.10.6/gst-libs/gst/rtp/gstrtpbuffer.h:
GST_RTP_PAYLOAD_MPA = 14,             /* Audio MPEG 1-3 */

so try with

application/x-rtp, media=audio, payload=14, media=(string)audio,
clock-rate=(int)90000, encoding-name=(string)MPA'

On 6/7/06, Bebjak, Michal <michal.bebjak at siemens.com> wrote:
>
> I tied to start it with --gst-debug-level=5 and I can see this following
> errors:
>
> DEBUG (0x8122190 - 0:00:00.553758000) 
> default( 5798)
> gstrtpmpadepay.c(213):gst_rtp_mpa_depay_chain: Unexpected
> payload type 96
> LOG   (0x8122190 - 0:00:00.553790000) 
> GST_REFCOUNTING( 5798)
> gstminiobject.c(293):gst_mini_object_unref: 0x8119e70
> unref 1->0
> LOG   (0x8122190 - 0:00:00.553812000)
>           GST_BUFFER( 5798)
> gstbuffer.c(182):gst_buffer_finalize: finalize 0x8119e70
> LOG   (0x8122190 - 0:00:00.553834000)
>             GST_CAPS( 5798)
> gstcaps.c(381):gst_caps_unref: 0x8108068 3->2
> LOG   (0x8122190 - 0:00:00.553859000)
>       GST_SCHEDULING( 5798)
> gstpad.c(3177):gst_pad_chain:<rtpmpadepay0:sink> called
> chainfunction &0xb79b0a50 of pad rtpmpadepay0:sink, returned error
>
> LOG   (0x8122190 - 0:00:00.553885000) 
> GST_REFCOUNTING( 5798)
> gstobject.c(412):gst_object_unref:<rtpmpadepay0:sink>
> 0x80ed1e0 unref 2->1
> LOG   (0x8122190 - 0:00:00.553908000) 
> GST_REFCOUNTING( 5798)
> gstobject.c(412):gst_object_unref:<capsfilter0> 0x81084d8
> unref 2->1
> LOG   (0x8122190 - 0:00:00.553931000)
>       GST_SCHEDULING( 5798)
> gstpad.c(3177):gst_pad_chain:<capsfilter0:sink> called
> chainfunction &gst_base_transform_chain of pad capsfilter0:sink, returned
> error
>
> LOG   (0x8122190 - 0:00:00.553956000) 
> GST_REFCOUNTING( 5798)
> gstobject.c(412):gst_object_unref:<capsfilter0:sink>
> 0x8107f40 unref 2->1
> DEBUG (0x8122190 - 0:00:00.553978000) 
> basesrc( 5798) gstbasesrc.c(1312):gst_base_src_loop:<udpsrc0>
> pausing task, reason error
> DEBUG (0x8122190 - 0:00:00.554002000)
>                 task( 5798) gsttask.c(438):gst_task_pause:<task0>
> Pausing task 0x810f9c0
> WARN  (0x8122190 - 0:00:00.554038000) 
> basesrc( 5798) gstbasesrc.c(1318):gst_base_src_loop:<udpsrc0>
> error: Internal data flow error.
> WARN  (0x8122190 - 0:00:00.554060000) 
> basesrc( 5798) gstbasesrc.c(1318):gst_base_src_loop:<udpsrc0>
> error: streaming task paused, reason error
>
>
> Do you think problem is that the rtpmpadepay element doesn't accept the
> "payload" argument?? The rest of the logs looks OK. There are some
> additional warning messages, but I think that is not the problem. Just to be
> sure here are the messages:
>
> WARN  (0x8052240 - 0:00:00.034834000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstmms.so: undefined symbol: mmsh_connect
> WARN  (0x8052240 - 0:00:00.040353000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgsttrm.so: undefined symbol:
> gst_pad_query_peer_duration
> WARN  (0x8052240 - 0:00:00.040762000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstannodex.so: undefined symbol:
> g_intern_static_string
> WARN  (0x8052240 - 0:00:00.041090000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstapetag.so: undefined symbol:
> gst_type_find_helper_for_buffer
> WARN  (0x8052240 - 0:00:00.041383000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstauparse.so: undefined symbol:
> gst_pad_query_peer_duration
> WARN  (0x8052240 - 0:00:00.041748000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstavi.so: undefined symbol:
> gst_pad_query_peer_duration
> WARN  (0x8052240 - 0:00:00.042095000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstdebug.so: undefined symbol:
> gst_pad_query_peer_duration
> WARN  (0x8052240 - 0:00:00.042571000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgsticydemux.so: undefined symbol:
> gst_type_find_helper_for_buffer
> WARN  (0x8052240 - 0:00:00.042983000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstid3demux.so: undefined symbol:
> gst_tag_from_id3_user_tag
> WARN  (0x8052240 - 0:00:00.048695000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstossaudio.so: undefined symbol:
> g_intern_static_string
> WARN  (0x8052240 - 0:00:00.050635000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstshout2.so: undefined symbol:
> gst_base_sink_set_sync
> WARN  (0x8052240 - 0:00:00.051117000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstwavparse.so: undefined symbol:
> gst_adapter_new
> WARN  (0x8052240 - 0:00:00.053315000)   GST_PLUGIN_LOADING( 5834)
> gstplugin.c(414):gst_plugin_load_file: module_open failed:
> /usr/lib/gstreamer-0.10/libgstswfdec.so: undefined symbol:
> swfdec_decoder_get_version
>
>
> Best regards,
>
> Michal
>
> -----Original Message-----
> From: Antoine Tremblay [mailto:hexa00 at gmail.com]
> Sent: Tue 6/6/2006 11:54 PM
> To: Bebjak, Michal
> Cc: gstreamer-devel at lists.sourceforge.net
> Subject: Re: [gst-devel] problem sending audio/mpeg over RTP
>
> hehe now that I think of it it should work with the default ....but still
> it's a good thing to set it just to be sure...
>
> try to run it with --gst-debug-level=5 or any level and check what's
> happening juste before the error...
>
>
> On 6/6/06, Antoine Tremblay <hexa00 at gmail.com> wrote:
> >
> > Well you're missing some args in udpsink /udpsrc
> >
> > Should be like udpsink host=localhost port=7777
> >
> > udpsrc port=7777
> >
> > the rest seems ok at 1st glance...
> >
> > Regards
> >
> > Antoine
> >
> >
> >
> > On 6/6/06, Bebjak, Michal <michal.bebjak at siemens.com> wrote:
> > >
> > >
> > > Hi,
> > >
> > > I'm developing an client-server aplication which should use GStreamer.
> I
> > > want to endcode the audio into MP3 and send it oved RTP to the client.
> I
> > > first tried to run this following commands:
> > >
> > > server terminal:  gst-launch-0.10 -v audiotestsrc ! lame ! rtpmpapay !
> > > udpsink
> > > client terminal:  gst-launch-0.10 -v udpsrc ! 'application/x-rtp,
> > > media=audio, payload=96, media=(string)audio, clock-rate=(int)90000,
> > > encoding-name=(string)MPA' !
> > >                   rtpmpadepay !
> > > 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' ! mad !
> > > audioconvert ! volume volume=0.2 ! autoaudiosink
> > >
> > >
> > > They both work independently but when I try to send the audio from
> > > server to client the client crashes. The terminal output is:
> > >
> > >
> > > $ gst-launch-0.10 -v udpsrc ! 'application/x-rtp, media=audio,
> > > payload=96, media=(string)audio, clock-rate=(int)90000,
> > > encoding-name=(string)MPA' !
> > > rtpmpadepay ! 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100'
> !
> > > mad ! audioconvert ! volume volume=0.2 ! autoaudiosink
> > > Setting pipeline to PAUSED ...
> > > Pipeline is live and does not need PREROLL ...
> > > Setting pipeline to PLAYING ...
> > > New clock: GstSystemClock
> > > /pipeline0/capsfilter0.sink: caps = application/x-rtp,
> > > media=(string)audio, payload=(int)96, clock-rate=(int)90000,
> > > encoding-name=(string)MPA
> > > /pipeline0/capsfilter0.src: caps = application/x-rtp,
> > > media=(string)audio, payload=(int)96, clock-rate=(int)90000,
> > > encoding-name=(string)MPA
> > > /pipeline0/rtpmpadepay0.sink: caps = application/x-rtp,
> > > media=(string)audio, payload=(int)96, clock-rate=(int)90000,
> > > encoding-name=(string)MPA
> > > ERROR: from element /pipeline0/udpsrc0: Internal data flow error.
> > > Additional debug info:
> > > gstbasesrc.c(1318): gst_base_src_loop (): /pipeline0/udpsrc0:
> > > streaming task paused, reason error
> > > Execution ended after 1513470000 ns.
> > > Setting pipeline to PAUSED ...
> > > Setting pipeline to READY ...
> > > /pipeline0/rtpmpadepay0.sink: caps = NULL
> > > /pipeline0/capsfilter0.sink: caps = NULL
> > > /pipeline0/capsfilter0.src: caps = NULL
> > > Setting pipeline to NULL ...
> > > FREEING pipeline ...
> > >
> > > Can someone please help me to make it work?? Thanks a lot!!
> > >
> > > Michael
> > >
> > >
> > > _______________________________________________
> > > gstreamer-devel mailing list
> > > gstreamer-devel at lists.sourceforge.net
> > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> > >
> >
> >
>
>
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