[gst-devel] Chain building problem

Doulbou Doulbou doulbousiouf at yahoo.fr
Wed Jun 1 13:31:24 CEST 2005


Hi

I try to make a aplication using gstreamer and i have
some problems. My goal is to acquire sound using my
microphone, compress it in mp3 format, send the flow
throw the network, decompress the sound in another pc
and play it.

It seems very easy to make with gst-launch but i don't
arrive to do what i want easily. To acquire and play
sound i use the alsasrc and alsasink, to compress it i
use lame and to decode it i use mad. I have made a lot
of tests and results are very stranges, what i done is
:

First a simple test to be sure alsasrc and alsasink
work correctly : gst-launch-0.8  alsasrc ! alsasink.
And it's ok soud is correctly played when speak to my
microphone.

A second to test the data compression : gst-launch-0.8
 sinesrc ! lame ! mad ! alsasink. It work fine.

A third test to try use alsasrc instead of sinesrc. It
done a negiciation error and i solved the caps
negotiation manually : gst-launch-0.8  alsasrc !
audioconvert ! "audio/x-raw-int", endianess=1234, 
width=16, depth=16, rate=44100, channels=1,
signed=(boolean)true ! lame ! mad ! alsasink. It does
not work so, it run correctly but i head nothing ...

After a lot of other tests i arrived to make it work
by doing in one fist terminal : gst-launch-0.8 alsasrc
! audioconvert ! "audio/x-raw-int", endianess=1234, 
width=16, depth=16, rate=44100, channels=1,
signed=(boolean)true ! filesink location=pipe
And in another terminal : gst-launch filesrc
location=pipe ! "audio/x-raw-int" , rate=44100 ! lame
! mad ! alsasink

Using a named pipe to transmit data they are correctly
played, if i delete the audioconvert in the first
terminal it don't work and if i delete the pipe it's
the same things.

Normally the second test must work fine and i don't
understand where is the problem.

Does somebody can help me to solve this or maybe it's
a  gstreamer bug ?

__________________
Doulbousiouf


	

	
		
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