Date: Wed, 25 Apr 2018 08:12:04 +0200 From: Ralf Mardorf <ralf.mardorf@rocketmail.com> To: freebsd-questions@freebsd.org Subject: Re: audiophile sound on FreeBSD ? Message-ID: <20180425081204.34bb8047@archlinux> In-Reply-To: <20180424235410.5e175bc6@gumby.homeunix.com> References: <20180423224242.7299f430@WorkMachine> <20180424113308.52f35f93@WorkMachine> <20180424200924.12c648bf@archlinux> <20180424235410.5e175bc6@gumby.homeunix.com>
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On Wed, 25 Apr 2018 00:21:31 +0100, RW via freebsd-questions wrote: >On Tue, 24 Apr 2018 20:09:24 +0200 Ralf Mardorf wrote: >> Btw. an "audio CD" already provides not that good audio quality. What >> you need for high end audio quality are 48 KHz, > >I'd be astonished if you could tell the difference under double blind >conditions - even if you are a child. I'd be surprised, if most adults with trained hearing would be unable to perceive a difference. To keep it short, I'll quote a short and raw explanation: "> frankly, 48k may be a good enough for distribution, but it is > sub-optimal not for production ... and it is horrible for digital > synthesis. Only if you use 'primitive' algorithms. Unfortunately there's a lot of those around. In summary, 96 or 192 kHz will allow you to use simpler algorithms. That may be a good reason for higher sampler rates, but it doesn't mean you can't have the same performance at 48 kHz. Another good reason for higher sampling rates is that the antialising filters in the converters can have a much wider transition band (assuming you don't actually use the higher bandwidth), leading to much reduced latency. It's the reason why 'digital snakes' used in PA system usually work at 96 kHz. By starting the transition band at 24 kHz or so they can use very short filters, a fraction of a millisecond for some. The same matter makes all the difference between 44.1 and 48 kHz." - http://linux-audio.4202.n7.nabble.com/jack-oversampling-td89817.html Loosely speaking: It all depends on the tuning. You could assume that most gear and software works best at 48 KHz, this is professional audio studio standard and apart from audio CD, consumer standard, too. However, depending on your amp, your speakers or any other part of the audio chain or depending on the recorded signal, such as e.g. an electric guitar played via a Celestion speaker, you might be even unable to hear any loss when using DAT longplay (32 KHz, 12 bit non-linear). Since the subject is "audiophile sound on FreeBSD" I recommend to go with a RME audio interface or another professional audio interface. The OP should ensure that an audio interface that claims to be "professional" isn't just a "prosumer" audio interface. Without doubts, 48 KHz is the best choice. When using a professional audio interface, with an elCheapo Hifi amp and supermarket speakers, you wont be able to hear the good analog audio quality of your professional audio interface. If you can't hear the good analog audio quality, you can't notice any issues of different sample rates.
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