[gst-devel] help: about rtp

shirie shizibo at gmail.com
Thu Apr 16 04:30:52 CEST 2009


How to set the queue size and how large it should be ok?

I don't know what's going on. I just do what the doc says. I want to test
transmitting video use rtp. So how should I built a proper pipeline?
Thanks.


thiagoss wrote:
> 
> On Wed, Apr 15, 2009 at 11:11 PM, thiagoss <thiagossantos at gmail.com>
> wrote:
> 
>>
>>
>> On Wed, Apr 15, 2009 at 10:44 PM, shirie <shizibo at gmail.com> wrote:
>>
>>>
>>> sorry, I mispaste the output. Here is the correct:
>>> THe pipeline is :
>>> gst-launch -v filesrc location=shake.mp4 ! qtdemux name=d \
>>> d. ! { queue ! rtpmp4gpay ! udpsink port=5000 }  \
>>> d. ! { queue ! rtpmp4vpay ! udpsink port=5002 }
>>>
>>> and the output is;
>>> Setting pipeline to PAUSED ...
>>> /pipeline0/queue0.sink: caps = video/mpeg, mpegversion=(int)4,
>>> systemstream=(boolean)false,
>>>
>>> codec_data=(buffer)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> width=(int)320, height=(int)240, framerate=(fraction)15/1
>>> /pipeline0/queue0.src: caps = video/mpeg, mpegversion=(int)4,
>>> systemstream=(boolean)false,
>>>
>>> codec_data=(buffer)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> width=(int)320, height=(int)240, framerate=(fraction)15/1
>>> /pipeline0/rtpmp4gpay0.src: caps = application/x-rtp,
>>> media=(string)video,
>>> payload=(int)96, clock-rate=(int)90000,
>>> encoding-name=(string)MPEG4-GENERIC,
>>> ssrc=(guint)80355967, clock-base=(guint)309711442,
>>> seqnum-base=(guint)5287,
>>> streamtype=(string)4, profile-level-id=(string)1, mode=(string)generic,
>>>
>>> config=(string)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3
>>> /pipeline0/rtpmp4gpay0.sink: caps = video/mpeg, mpegversion=(int)4,
>>> systemstream=(boolean)false,
>>>
>>> codec_data=(buffer)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> width=(int)320, height=(int)240, framerate=(fraction)15/1
>>> /pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
>>> payload=(int)96, clock-rate=(int)90000,
>>> encoding-name=(string)MPEG4-GENERIC,
>>> ssrc=(guint)80355967, clock-base=(guint)309711442,
>>> seqnum-base=(guint)5287,
>>> streamtype=(string)4, profile-level-id=(string)1, mode=(string)generic,
>>>
>>> config=(string)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3
>>> Pipeline is PREROLLING ...
>>>
>>> There is really only one udpsink. What's problem?
>>
>>
>> Try setting the queue's max size to a greater value.
>>
> Also, it seems that your second queue (the audio one) doesn't receive any
> data. Did you try another file? Maybe one that data is interleaved?
> 
> I guess that the first udpsink is waiting to sync with the other one that
> never receives data because the video queue has blocked (already is full)
> and consequently might block the demuxer. But I might be wrong, I haven't
> gone far in gst in clock/sync issues.
> 
> 
>>
>>>
>>>
>>>
>>> thiagoss wrote:
>>> >
>>> > On Wed, Apr 15, 2009 at 12:07 AM, Zibo SHI <shizibo at gmail.com> wrote:
>>> >
>>> >> hi, all. I'm trying the rtp plugin refer to the doc at:
>>> >>
>>> >>
>>> >>
>>> http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/gst/rtp/README?revision=1.13&view=markup
>>> >>
>>> >> Here's my simple pipeline:
>>> >> gst-launch -v filesrc location=shake.mp4 ! qtdemux name=d \
>>> >> d. ! { queue ! rtpmp4vpay ! udpsink port=5000 } \
>>> >> d. ! { queue ! rtpmp4gpay ! udpsink port=5002 }
>>> >>
>>> >> but after it's run, here is the output:
>>> >>
>>> >> Setting pipeline to PAUSED ...
>>> >> /pipeline0/queue0.sink: caps = video/mpeg, mpegversion=(int)4,
>>> >> systemstream=(boolean)false,
>>> >>
>>> codec_data=(buffer)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> >> width=(int)320, height=(int)240, framerate=(fraction)15/1
>>> >> /pipeline0/queue0.src: caps = video/mpeg, mpegversion=(int)4,
>>> >> systemstream=(boolean)false,
>>> >>
>>> codec_data=(buffer)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> >> width=(int)320, height=(int)240, framerate=(fraction)15/1
>>> >> /pipeline0/rtpmp4vpay0.src: caps = application/x-rtp,
>>> >> media=(string)video,
>>> >> payload=(int)96, clock-rate=(int)90000,
>>> encoding-name=(string)MP4V-ES,
>>> >> ssrc=(guint)3831296950, clock-base=(guint)4106776350,
>>> >> seqnum-base=(guint)53267, profile-level-id=(string)1,
>>> >>
>>> config=(string)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34
>>> >> /pipeline0/rtpmp4vpay0.sink: caps = video/mpeg, mpegversion=(int)4,
>>> >> systemstream=(boolean)false,
>>> >>
>>> codec_data=(buffer)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34,
>>> >> width=(int)320, height=(int)240, framerate=(fraction)15/1
>>> >> /pipeline0/queue1.sink: caps = audio/mpeg, mpegversion=(int)4,
>>> >> framed=(boolean)true, codec_data=(buffer)1290, rate=(int)32000,
>>> >> channels=(int)2
>>> >> /pipeline0/queue1.src: caps = audio/mpeg, mpegversion=(int)4,
>>> >> framed=(boolean)true, codec_data=(buffer)1290, rate=(int)32000,
>>> >> channels=(int)2
>>> >> /pipeline0/rtpmp4gpay0.src: caps = application/x-rtp,
>>> >> media=(string)audio,
>>> >> payload=(int)96, clock-rate=(int)32000,
>>> >> encoding-name=(string)MPEG4-GENERIC,
>>> >> ssrc=(guint)3458294825, clock-base=(guint)2594580451,
>>> >> seqnum-base=(guint)42905, encoding-params=(string)2,
>>> >> streamtype=(string)5,
>>> >> profile-level-id=(string)2, mode=(string)AAC-hbr,
>>> config=(string)1290,
>>> >> sizelength=(string)13, indexlength=(string)3,
>>> indexdeltalength=(string)3
>>> >> /pipeline0/rtpmp4gpay0.sink: caps = audio/mpeg, mpegversion=(int)4,
>>> >> framed=(boolean)true, codec_data=(buffer)1290, rate=(int)32000,
>>> >> channels=(int)2
>>> >
>>> >
>>> >
>>> > here
>>> >
>>> >>
>>> >> /pipeline0/udpsink1.sink: caps = application/x-rtp,
>>> media=(string)audio,
>>> >> payload=(int)96, clock-rate=(int)32000,
>>> >> encoding-name=(string)MPEG4-GENERIC,
>>> >> ssrc=(guint)3458294825, clock-base=(guint)2594580451,
>>> >> seqnum-base=(guint)42905, encoding-params=(string)2,
>>> >> streamtype=(string)5,
>>> >> profile-level-id=(string)2, mode=(string)AAC-hbr,
>>> config=(string)1290,
>>> >> sizelength=(string)13, indexlength=(string)3,
>>> indexdeltalength=(string)3
>>> >
>>> >
>>> >
>>> > and here
>>> >
>>> >>
>>> >> /pipeline0/udpsink0.sink: caps = application/x-rtp,
>>> media=(string)video,
>>> >> payload=(int)96, clock-rate=(int)90000,
>>> encoding-name=(string)MP4V-ES,
>>> >> ssrc=(guint)3831296950, clock-base=(guint)4106776350,
>>> >> seqnum-base=(guint)53267, profile-level-id=(string)1,
>>> >>
>>> config=(string)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335312e34302e34
>>> >
>>> >
>>> >
>>> >>
>>> >> Pipeline is PREROLLING ...
>>> >> Pipeline is PREROLLED ...
>>> >> Setting pipeline to PLAYING ...
>>> >> New clock: GstSystemClock
>>> >>
>>> >> According to the doc, there should be udpsink0 and udpsink1, but I
>>> only
>>> >> have one udpsink in the output.
>>> >> Is there anything wrong  with my pipeline? I'm fairly new with
>>> gstreamer.
>>> >> Can anyone hlep me? Thanks very much.
>>> >>
>>> >
>>> > Look again, there they are!
>>> >
>>> >
>>> >>
>>> >>
>>> >>
>>> ------------------------------------------------------------------------------
>>> >> This SF.net email is sponsored by:
>>> >> High Quality Requirements in a Collaborative Environment.
>>> >> Download a free trial of Rational Requirements Composer Now!
>>> >> http://p.sf.net/sfu/www-ibm-com
>>> >> _______________________________________________
>>> >> gstreamer-devel mailing list
>>> >> gstreamer-devel at lists.sourceforge.net
>>> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>> >>
>>> >>
>>> >
>>> >
>>> > --
>>> > Thiago Sousa Santos
>>> >
>>> > Embedded Systems and Pervasive Computing Lab (Embedded)
>>> > Center of Electrical Engineering and Informatics (CEEI)
>>> > Federal University of Campina Grande (UFCG)
>>> >
>>> >
>>> ------------------------------------------------------------------------------
>>> > This SF.net email is sponsored by:
>>> > High Quality Requirements in a Collaborative Environment.
>>> > Download a free trial of Rational Requirements Composer Now!
>>> > http://p.sf.net/sfu/www-ibm-com
>>> > _______________________________________________
>>> > gstreamer-devel mailing list
>>> > gstreamer-devel at lists.sourceforge.net
>>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>> >
>>> >
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/help%3A-about-rtp-tp23051783p23070080.html
>>> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Stay on top of everything new and different, both inside and
>>> around Java (TM) technology - register by April 22, and save
>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco.
>>> 300 plus technical and hands-on sessions. Register today.
>>> Use priority code J9JMT32. http://p.sf.net/sfu/p
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> gstreamer-devel at lists.sourceforge.net
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>
>>
>>
>> --
>> Thiago Sousa Santos
>>
>> Embedded Systems and Pervasive Computing Lab (Embedded)
>> Center of Electrical Engineering and Informatics (CEEI)
>> Federal University of Campina Grande (UFCG)
>>
> 
> 
> 
> -- 
> Thiago Sousa Santos
> 
> Embedded Systems and Pervasive Computing Lab (Embedded)
> Center of Electrical Engineering and Informatics (CEEI)
> Federal University of Campina Grande (UFCG)
> 
> ------------------------------------------------------------------------------
> Stay on top of everything new and different, both inside and 
> around Java (TM) technology - register by April 22, and save
> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco.
> 300 plus technical and hands-on sessions. Register today. 
> Use priority code J9JMT32. http://p.sf.net/sfu/p
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> 
> 

-- 
View this message in context: http://www.nabble.com/help%3A-about-rtp-tp23051783p23070414.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.





More information about the gstreamer-devel mailing list