Date: Fri, 29 Mar 2002 19:01:04 -0600 (CST) From: John Utz <john@utzweb.net> To: Rahul Siddharthan <rsidd@online.fr> Cc: "Adam D. Gorski" <agorski@engin.umich.edu>, <freebsd-multimedia@FreeBSD.ORG> Subject: Re: SB problem (was: Cat'ing /dev/audio) Message-ID: <Pine.LNX.4.44.0203291848540.11448-100000@jupiter.linuxengine.net> In-Reply-To: <20020330004311.GA1858@lpt.ens.fr>
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On Sat, 30 Mar 2002, Rahul Siddharthan wrote: > > 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. ahh, good point. but i have both more sophisticated and less sophisticated cards then that card and neither of them have the problem. *but* that's not the same thing as having the exact card, so i take your point. but he has neighbors running the same card without trouble. > 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.... ahh, here's where we are talking past each other. and maybe i missed something. your assuming that an mp3 or ogg is the same type of file as a .wav or a .au file. they are not. in order to play an ogg file or an mp3 file, *or* to *convert* them to a .wav or a .au or a .aiff file. so, if you start with an ogg or mp3 and turn it into a low or high bitrate mp3, and you have math problems, you will get nasty sounds when you play it. that would tell me it's a math problem if you start with a high bitrate wav and a lo bitrate wave and the lo bitrate wave doesnt have noise and the highbitrate one does, then your right, math is irrelevant. but my recoolection was the original complaint was that wavs played ok, but mp3's didnt > Methinks it's time one of the experts on the list spoke up. they've had plenty of time and they havent shared.... > Rahul > -- John L. Utz III john@utzweb.net Idiocy is the Impulse Function in the Convolution of Life To Unsubscribe: send mail to majordomo@FreeBSD.org with "unsubscribe freebsd-multimedia" in the body of the message
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