• - The 'clients' are not MP3-experts, they simly want to download the speech an hear it (as mp3 in portable-players, or they burn it as a audio-cd). So the files have to be compatible (thats why I can not use speex or ogg).
• - download should also be interesting for modem/isdn-users (about: do not download longer than the playtime of the file is!)

I recommend you use WMA with it's ACELP.net encoder.

Usage is as no-brainer as it is with MP3.

All you need is any program that supports WMA.

Never heard about 'no brainer' ?!

Hu, that sound so easy    Is there a similar method to Lame&RazorLame to compress WAV to WMA with the ACELP codec?

Do you know how much I have to pay for using acelp.net / wma?

"easy". You need no brain to use it.

Is there a possibility to use the ACELP.net WMA Encoder without the Windows Media Encoder (I'd like something like RazorLame)?

Try --preset medium (VBR) in LAME - for speech files, it might produce bitrates you want
or --alt-preset <bitrate> (ABR), --alt-preset cbr <bitrate>
Or use --preset help and pick one that suits you - beware, they're mostly old.

MMJB employs modifies FhG codec, which had many bugs, that affected its quality.

You can use dBpowerAMP, it's a very good encoder.

I'm very happy with the sound quailty of aceltpå net.

you can go down to 8kbits and it still sound lisenable for story telling

First you mentioned the  ACELP.net / WMA Encoder.. Do you think dBpowerAMP is equivalently good or even better? So I'd need  the dMC + WMA PlugIn?

Whats about the difference between MP3 @ 80kbps CBR // VBR ?

And I would recommend using either MP3enc 3.1 or MusicMatch's Fastenc

Go with CBR. It will probably be overkill anyway given you are encoding 32kHz/Mono...

MP3Enc  (info @ dBpowerAmp Website) is about 200$. Uff. Does anybody know if differences between MP3Enc and FastEnc or LAME are large?
You wrote 'MusicMatch's Fastenc' - are there different versions of FastEnc? Does Music Match's FastEnc differ from FHGs FastEncc?

MP3enc is based on FhG's professional encoder, and I would guess that Musicmatch's slow setting uses that encoder.

MusicMatch Jukebox uses multiple Fraunhofer encoders based on the mode
and settings used to record your audio tracks.  Which encoder is used
for which modes is proprietary information that we can not disclose.

edit: (Perhaps the noise also counts as (unwanted) information which has to be saved and this way there is less space left for the wanted signal?!
...
)

Does a lowpass filter help?

Again it's a trade-off as the more noise you take away the more likely you get artifacts that probably get worse by encoding.

I thought you already used a lowpass

Now I switched to dbPowerAMP (because it supports WMA and is VERY easy to use - exactly what I need, because I have to explain that to some ordinary-user).

dbPowerAMP uses the lame.dll - so I'm not shure if there is a lowpassfilter, yet?!

Would you suggest to use --alt-preset <bitrate> to create a 64kbps / 32Khz / mono file ? (is it possible?)

If you use something less easy for *en*coding what'll be the difference for the ordinary-user (besides maybe better quality)?

1) Find out what lowpass setting you can tolerate [...]

(the lower [the highpass-filter value,] the better to avoid later encoding artifacts)

2) choose the next possible sampling rate above 2*<x> (e.g. 2*7 = 14 -> 16kHz) for resampling.

24kbps speech:
--alt-preset 24 -a --resample 22 --lowpass 7 -Z

I'd test to find a level of noise reduction small enough to avoid artifacts - and choose a noise reduction level even lower than that (e.g. 2dB) to have some headroom for encoding.

it's not necessary to perform noise reduction before encoding unless noise is just too loud for your taste

I wouldn't use compression/limiting as it increases noise level or causes noise modulation (bumping) which is more annoying than constant noise IMO

Cool Edit is a safer choice than Goldwave in terms of quality. Don't have much experience but I do know that for instance the resampling in Goldwave is pretty bad

Quote

(the lower [the highpass-filter value,] the better to avoid later encoding artifacts)

Again, because there is less information, which has to fit into the given kbps-limitation!?

Sample rates above '2x[value of lowpassfilter]' do not make much sense?

If so, I dont understand, why @ Lame Settings For Speech they recommendet

Quote

24kbps speech:
--alt-preset 24 -a --resample 22 --lowpass 7 -Z

Yet, I used 32kbps, because it still is mpeg1 Layer3 (and this more compatible?!)..

For my requirements in terms of the quality, in my own experiments I did not like bitrates less than 64kbps. Would you also resemple to 16kHz (with lowpass 7) for 64kbps - or is 32kbps all right?

Sorry,  I don't really understand what all this means [span style='font-size:8pt;line-height:100%'](When I use the noise-reduction filter, I copy a part of the noise and than aply the filter. So there is no db value - see GoldWave Screenshot)[/span]

How does the noise reduction influence artifacts? What does 'headroom for encoding' mean?

OK: You've said you want MPEG1 so you can't resample to anything lower than 32kHz. So as 64kbps setting (ABR) I'd try
--alt-preset 64 -m m --resample 32 --lowpass 8

-b 32 -B 128 -a --abr 80 -F --resample 32 --lowpass 16 -h -c  --athtype 2 -X3