[gst-devel] sending audio/video on RTP with C API

[pierre.crepieux at orange-ftgroup.com]

Thanks,

I gave a look at your code, and found it instructive. It didn't really 
highlight the problem I had (i'm using dshow and directsound src), but 
anyway, I got my sample working :-)  There is still one thing i don't 
get: my sample works if I use h263 encoding but doesn't with h264 ...

I still have many thing to learn regarding gstreamer !


Miguel Telleria de Esteban a écrit :
> Hello,
>
> Like you are doing now, I followed the same goal of using gstrtpbin and
> found many small nuances.
>
> You can find here an implementation that I did to send ogg (vorbis and
> theora) audio through RTP using gstrtpbin.
>
> http://www.ctr.unican.es/asignaturas/dec/Doc/ogg_audio_video_rtp_sender_receiver.tar.gz
>
> Hope it helps.
>
> And yes, I plan to submit it as a bugzilla addition sometime in the
> future :).
>
> 	Miguel
>
>
> On Fri, 07 Jan 2011 11:36:46 +0100 pierre.crepieux
> wrote:
>
>   
>> Hi,
>>
>> I'm trying to write a simple C program that sends webcam/microphone
>> streams on rtp with gstrtpbin. I gave a look at the provided
>> server-alsasrc-PCMA.c test program and mimics it to add h264 stream
>> to the rtpbin (just like server-v4l2-H264-alsasrc-PCMA.sh does).
>> Sadly, using VLC to receive the streams, give me a fixed image and a
>> pretty bad sound (delay, sometime stop/restart,...). Are there any
>> tricks to be aware of for doing this translation of gst-launch pipe
>> definition to C code ? Does gstrtpbin require subtle manipulations
>> (that gst-launch would do behind the scene ... ??? ) to operate with
>> 2 rtp stream ?
>>
>> Pierre
>> --
>>
>>     
>
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