Hi All,
We are facing an issue with playbin while playing AVI file.
We have tested following gst pipeline using gst-launch application:
gst-launch filesrc location=MPEG4SP_MP3_512kbps_25fps_QVGA_WilsonWar.avi !
avidemux name=demux {demux.audio_00 ! queue ! omx_mp3dec ! alsasink } {
demux.video_00 ! queue ! omx_mpeg4dec ! xvimagesink }
This pipeline works fine. Here we have used GST-OMX component for
audio-video decoding, avidemux as avi demuxer and alsasink/xvimagesink for
audio/video rendering, respectively.
On the same system when we tested the playback using following playbin, it
starts playback and displays 2-3 frames after that it stops.
gst-launch playbin uri=file://MPEG4SP_MP3_512kbps_25fps_QVGA_WilsonWar.avi
When we checked the pipeline it is similar to the one mentioned above with
gst-launch. We have observed with playbin that it is buffering initial video
frames. We have limited number of video frame buffer and they are getting
buffered in playbin which is causing this issue in playback.
Can we avoid this buffering, so that it will work like gst-launch
application?
Or
Is this buffering happens because of some thread synchronization?
The version of GST components which are used:
Gstreamer : 0.10.9
Gst plugins base : 0.10.9
Gst plugins good : 0.10.4
Gst openmax : 0.10.0.4
*************************
Regards,
Gulshan
On 8/18/08, Tiago Katcipis <katcipis at inf.ufsc.br> wrote:
>
> i will, if i find anything usefull for you guys of course...im just
> starting on voip...so im very newbie, it will be hard to me to find
> something interessting or new to you guys, but if i find i will let you know
> of course.
>
> im very thankfull for the help... if theres anything that i can do to
> help...ill do.
>
> better start readind :-)
>
> best regards
>
> Tiago
>
> On Sun, Aug 17, 2008 at 2:40 PM, Manish Rana <manish.rana at gmail.com>wrote:
>
>>
>> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html
>>
>>
>> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpsession.html
>>
>> will help you
>> also you can use the README file of gstrtpbin.
>> Think that will be good to start...after that you can read RFC's
>>
>> All the best....Please share your findings as well as querries, that is
>> how most of us learns :)
>>
>> Thanks a lot
>> Manish
>>
>> On Sun, Aug 17, 2008 at 8:21 PM, Tiago Katcipis <katcipis at inf.ufsc.br>wrote:
>>
>>> hmm thanks for the help guys, my last question is where to start? i know
>>> that i can always try google, but maybe i will loose time reading the wrong
>>> stuff, if you point me on the right direction i would appreciate.
>>>
>>> best regards
>>>
>>> tiago
>>>
>>>
>>> On Sun, Aug 17, 2008 at 7:02 AM, Manish Rana <manish.rana at gmail.com>wrote:
>>>
>>>> hey generally it depends on the network dealys............
>>>> but i think generally 1000ms is an Ok ok but RTCP will genreally let u
>>>> know the jitter value
>>>>
>>>>
>>>> On Sat, Aug 16, 2008 at 7:23 PM, Tiago Katcipis <katcipis at inf.ufsc.br
>>>> > wrote:
>>>>
>>>>> and how bigger would have to be the jitter buffer? i cant use on the
>>>>> lib a too big jitter.
>>>>>
>>>>> thanks everyone for the help
>>>>>
>>>>>
>>>>> On Sat, Aug 16, 2008 at 5:25 AM, Manish Rana <manish.rana at gmail.com>wrote:
>>>>>
>>>>>> hey but i setting the jitter-buffer latency we can take care of the
>>>>>> round trip delays
>>>>>> and alsa will get mostly sequenced audio packets....
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Aug 16, 2008 at 9:04 AM, gulshan karmani <
>>>>>> gulshan.karmani at gmail.com> wrote:
>>>>>>
>>>>>>> hi all,
>>>>>>> Only issue cud be voice rendering which has constraints of round trip
>>>>>>> delays due to naec and in that case use of alsa plugin cud be a
>>>>>>> problem.
>>>>>>> Rgds,
>>>>>>> Gulshan
>>>>>>>
>>>>>>> On 8/15/08, Manish Rana <manish.rana at gmail.com> wrote:
>>>>>>> > Hi,
>>>>>>> >
>>>>>>> > as far as the real time constrain is concerned, this should be
>>>>>>> taken care by
>>>>>>> > RTP and add the required delays shall be added by RTP, that is
>>>>>>> gstrtpbin
>>>>>>> > plugin in gstreamer.
>>>>>>> > On addition to this gstreamer will give u flexiblity to create the
>>>>>>> pipelines
>>>>>>> > as your requirements. You can have minimal elements as well or add
>>>>>>> more to
>>>>>>> > get the better audio. (like audio resample and audioconvert can be
>>>>>>> optional)
>>>>>>> >
>>>>>>> > And if i am not wrong Gstreamer is used successfully in maemo for
>>>>>>> the VoIP
>>>>>>> > application, and there is Farsight plugin available, which is
>>>>>>> optimised.
>>>>>>> >
>>>>>>> > I am sorry if I have any wrong info... Please correct me....
>>>>>>> >
>>>>>>> > Also please add more on the same...........
>>>>>>> >
>>>>>>> > BR
>>>>>>> > Manish
>>>>>>> > On Fri, Aug 15, 2008 at 3:58 PM, Tiago Katcipis <
>>>>>>> katcipis at inf.ufsc.br>wrote:
>>>>>>> >
>>>>>>> >> I'm working in a project using voip on a software and in embedded
>>>>>>> systems,
>>>>>>> >> its more like a lib, that is used by a high level software for PC
>>>>>>> and in
>>>>>>> >> embedded systems. Actually everything is done in one single
>>>>>>> gigantic
>>>>>>> >> function, now we are working on creating a more readable and
>>>>>>> expandable
>>>>>>> >> lib,
>>>>>>> >> so we started to build the lib using pipes and filters patterns.
>>>>>>> That's
>>>>>>> >> when
>>>>>>> >> the idea of using gstreamer came, but since gstreamer is usually
>>>>>>> used on
>>>>>>> >> media players i would like to know if it is good to be used on
>>>>>>> real time
>>>>>>> >> voip systems that rely heavily on time to work properly. Is
>>>>>>> Gstreamer a
>>>>>>> >> good
>>>>>>> >> lib to build this type of application? If it is who would be the
>>>>>>> best
>>>>>>> >> place
>>>>>>> >> for me to start reading about it (using gstreamer on voip) ?
>>>>>>> >>
>>>>>>> >> sorry if i asked something stupid... I'm just starting on the job
>>>>>>> and
>>>>>>> >> don't
>>>>>>> >> have to much experience, sorry for the lousy English too :-)
>>>>>>> >>
>>>>>>> >> Best regards
>>>>>>> >>
>>>>>>> >> Tiago C?sar Katcipis
>>>>>>> >>
>>>>>>> >>
>>>>>>> -------------------------------------------------------------------------
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>>>>>>> >> _______________________________________________
>>>>>>> >> Gstreamer-embedded mailing list
>>>>>>> >> Gstreamer-embedded at lists.sourceforge.net
>>>>>>> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded
>>>>>>> >>
>>>>>>> >>
>>>>>>> >
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>>
>>
>>
>>
>
>
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