[gst-devel] Best encoder for streaming video?

[Merrick Fonnesbeck]

Oliver, you know I haven't tried it recently with the built-in Google
Talk (Video) Beta application that it comes with to see how they
compare.  Do you know if they are using GStreamer or what encoder they
are using?  I am trying to something similar and am mainly concerned
about the video than the audio.

And Eric, if RTP has issues, what should I use for better transporting
of the video?  Is there something better or more reliable?  Please let
me know so that I might learn and improve this application, since I am
very new to developing on Maemo and these mobile devices and want to
become better.  Thanks to the both of you for your responses.

MFonnesbeck

------------------------------
>On Wed, 2008-08-27 at 14:18 -0600, Merrick Fonnesbeck wrote:
>> I am writing a video conferencing application for the Nokia N800/N810

>> devices that stream video between two devices, and am currently using

>> the hantro4200enc encoder in my GStreamer pipeline to encode the
video 
>> before sending it out.  The video appears on the other side very 
>> choppy and pixelated, and I am wondering if I am using the right 
>> encoder or whether there is a better encoder that would compress my 
>> video better to give me a cleaner video?  I got the code initially 
>> from the following wiki post telling me how to stream video from the 
>> Nokia devices.
>> 
>> http://maemo.org/community/wiki/howtostreamvideo/
>> 
>> If there is a better way let me know, thanks.
>
>Is it any worse than the built-in call application? Because thats close
to the best you can get out of this device.
>
>You can probably play with the parameters of the hantro4200enc element,
but it shouldn't change much...
>
>--
>Olivier Cr?te
>olivier.crete at collabora.co.uk
>Collabora Ltd
>-------------- next part --------------
>
>Hi, Merrick:
>
>    I think this is not an encoder issue. I noticed the wiki articles
you mentioned and I found it uses RTP as transport protocol. RTP is a
realtime transport protocol so if packets aren't arrived during  a
specified `latency', the packets will be considered as `late' and be
dropped. That's the main reason to make your video choppy and
pixelated.
>
>    RTP stuffs in gstreamer are still in `bad' plugin repository and
they need to be improved, such as lip-synchronization, jitterbuffer and
etc.
>
>Eric Zhang
>
>