[streaming] speex question

[FAIGL Zoltan]

Hi!

I would like to stream audio with speex codec, but I can't make
any effect on its parameters from command line. It's working well when I
am streaming voice, but I couldn't force it to use other bandwith than
16-20 kbyte/sec (two channels). In the documentation
(http://www.videolan.org/doc/streaming-howto/en/ch03.html) I found only
that  speex must be used strictly with ogg multiplexer, and it's for low
bitrate speech trasport.
I downloaded a user manual for speex, and I saw that it can be very well
configured with a lot of parameters,but I
couldn't manage to pass them to
the codec in vlc.


Please, could you help me, how is it possible to give parameters to the
speex encoder? And can it be used with other mux than ogg?

I copy here my last try:

vlc --advanced --extraintf dummy -vvv dshow:// :dshow-vdev="None"
:dshow-adev="NVIDIA(R) nForce(TM) Audio" :dshow-config
:sout=#transcode{acodec=spx,ab=20,aenc=speex{comp=6,quality=1}}:\
duplicate{dst=display,dst=std{access=http,mux=ogg,url=:8080}}

I thought that it works in a way that I can give the parameters for the
speex encoder with the aid of "aenc={}" option. But it has no effect on
it. So how to solve this problem?

I have two other questions. We can lower
the used bandwidth of the audio stream if we use one channel instead of
two. But, in
my case, if I give "transcode{...,channels=1}" the sound become very deep
and noisy, I think,it merges the left and right channels badly. Is it the
fault of
Windows (SP2 made strange things),how to solve the problem ?

I tried also to resample the incoming sound but I didn't manage to do it.
For example I tried:
... :sout=#transcode{acodec=mp3,ab=32,samplerate=22050,channels=2}:\
duplicate{dst=display,dst=std{access=http,mux=ogg,url=:1234}}

My experience was that in this case I gave to the encoder that it
interpret the
incoming audio as it would be 22kHz. But it doesn't resample the audio.
At the receiver side the audio became deep (my sound card get samples with
44kHz).
So my last question is, how can we resample the audio, which audio codec
does enable this, could you show an example also for this?


Thanks your help in advance,
best regards,
Zoltán

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