Speech compressed as MP3 for Stream and Download
Reply #18 – 2003-08-22 12:40:01
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). Questions I was going to ask here already answered in your 1st post[/edit] If you use something less easy for *en*coding what'll be the difference for the ordinary-user (besides maybe better quality)? dbPowerAMP uses the lame.dll - so I'm not shure if there is a lowpassfilter, yet?! I've had a quick look at DBPoweramp's options. You can use CBR presets like --alt-preset cbr 64 which have integrated resampling and lowpass but I haven't seen a possibility to convert to mono. Maybe it'd be better to use a frontend for lame.exe. edit: part II Would you suggest to use --alt-preset <bitrate> to create a 64kbps / 32Khz / mono file ? (is it possible?) I'd suggest to use lame.exe (+ frondend if needed - or create a .bat file for easy use) and 1) Find out what lowpass setting you can tolerate (the lower the better to avoid later encoding artifacts) by encoding some of your files with lame.exe from commandline like this: "lame --alt-preset standard --lowpass <x> infile.wav" with <x> values between 5.0 and 10.0. If you find that e.g. <x> = 7 still sounds ok for you, 2) choose the next possible sampling rate above 2*<x> (e.g. 2*7 = 14 -> 16kHz) for resampling. The commandline would be --alt-preset CBR 64 --lowpass 7 --resample 16 -m m. You might want to do some testing with switches mentioned in the other thread about encoding speech. 3) Highpass won't help much to avoid artifacts - and you'd have to use an external one as lame's lowest possible highpass is far too high to be useful for speech. 4) The only thing what could be done with a sound editor is noise reduction. 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. I don't have much experience with lame + bitrates that high for speech encoding (I usually go for 32kbps abr) but probably with a good commandline (mono, resampling, lowpass) 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 Pro 2.1 (which I use and am satisfied with) is exactly the same as Adobe Audition. Can't tell how it compares to goldwave.
Last Edit: 2003-08-22 13:39:51 by tigre