Date: Sat, 30 Mar 2002 01:43:11 +0100 From: Rahul Siddharthan <rsidd@online.fr> To: John Utz <john@utzweb.net> Cc: "Adam D. Gorski" <agorski@engin.umich.edu>, freebsd-multimedia@FreeBSD.ORG Subject: Re: SB problem (was: Cat'ing /dev/audio) Message-ID: <20020330004311.GA1858@lpt.ens.fr> In-Reply-To: <Pine.LNX.4.44.0203291744280.11448-100000@jupiter.linuxengine.net> References: <20020330004107.B88610@lpt.ens.fr> <Pine.LNX.4.44.0203291744280.11448-100000@jupiter.linuxengine.net>
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> i would also concede that your suggestion that the problem is a sampling > rate problem is possible. because it would be concievable that linux/alsa > might have a more sophisticated approach to automagically resampling > things. > > *but*, i simply dont run into content that has been sampled that way! i > havent ever run into stuff that i cant play via xmms on my 4.5 box. > evidently that isnt the case for you. Is yours the same card (SB16)? There are lots of cards which accept variable sampling rates; the problem arises when they don't. If you have absolutely no problems with an SB16 whatever the sampling rate of your soundfile, I'd agree that sampling rate is probably not the issue in this case. Note that with upsampling, you won't get problems nearly as bad. So if your card accepts 44100 Hz and all your soundfiles are 44100 Hz or less, you won't hear the kind of noise Adam complains of; there may be some slight distortion (that was my experience) but you need good speakers to make it out. The problem comes with downsampling, eg playing a 48000 Hz sound into a 44100 Hz card. Given that other people apparently use the SB16 on FreeBSD happily, my guess would be either Adam's computer clock is too fast (a signal which should be 44100 Hz ends up at 42000 Hz, say) or his cards accept only 22050Hz for some reason. And yes, alsa is probably pretty sophisticated about things like sample rate conversion. And now that it's become standard in the linux kernel, more and more linux-based sound applications are going to require it (some do already) and it will be difficult to port them to FreeBSD :-( > my current supposition is a math error, because i know that gcc with the > more specialized cpu architecture setups isnt particularly well tested, > and it's tested even less on freebsd! furthermore, preparing oggs and > mp3's for play is a math intensive process, so any goof ups in the math > code would be painfully evident. Well, gcc3 may well have problems, but the stock gcc (2.95.x) is quite well tested with vorbis and other audio software. Also it's not terribly obvious to me why a math error would affect high sampling rates and not low ones.... Methinks it's time one of the experts on the list spoke up. Rahul To Unsubscribe: send mail to majordomo@FreeBSD.org with "unsubscribe freebsd-multimedia" in the body of the message
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