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foobar "output" and what to set it at

post #1 of 12
Thread Starter 
what kind of specs do i need to know about my computer in order to determine what the buffer length and output data format should be at? what do these two options do exactly?
post #2 of 12
Buffer length can/should be at the lowest possible value before it starts to skip on playback.

Output data format is dependent on your music files. If your files are ripped from a CD, 16 bits is good.
post #3 of 12
Buffer length shouldn't really matter, but the larger the buffer, the more memory the player will use. The buffer is a memory area that the player uses to make sure that there's always data available to play, even if the disk is slow. The player reads MP3 (or whatever) data, decodes it and places it in the buffer memory ready for playing. Then, when the soundcard driver indicates that it needs data, all the player must do is copy the data from the buffer area to the soundcard. There are a number of reasons why this is necessary, but I won't get into them. Suffice to say that the buffer length basically determines how far 'ahead' of the sound coming out of the soundcard the player software is. It won't affect sound quality as long as it's large enough, though an excessively large buffer may cause delays when starting playback.

Output data format should be 16bit unless you have a 24bit capable soundcard and 24bit source files to play (or want to dither).
post #4 of 12
I disagree on 16 bit depth. You should use 24 or 32 if your sound card will support it. Although CDs only have 16 bits per sample, if you use any DSPs in foobar, including the volume control, then the extra bits become relevant.
post #5 of 12
Quote:
Originally Posted by Rempert View Post
I disagree on 16 bit depth. You should use 24 or 32 if your sound card will support it. Although CDs only have 16 bits per sample, if you use any DSPs in foobar, including the volume control, then the extra bits become relevant.
My understanding is that foobar's internal processing pipeline is entirely 32bit regardless of input and output formats, which is then either dithered or truncated to the output format depending on your configuration. There's a fairly decent chance that I'm wrong though. Since we're all using 32bit processors there's no penalty for doing this, in fact it may be 'cheaper' since checking for overflow is simpler. Makes sense from an audio quality standpoint as well, so I'm pretty sure this is how it's done. Correct me if I'm wrong though.
post #6 of 12
Sounds right... At least in XP, all calculations should be 32-bit, and this is the way most professional audio software works.
post #7 of 12
Quote:
Originally Posted by error401 View Post
My understanding is that foobar's internal processing pipeline is entirely 32bit regardless of input and output formats, which is then either dithered or truncated to the output format depending on your configuration.
But if you have 32 bit audio after processing, and you have a 24 bit soundcard, why would you choose to dither or truncate down to 16 bits? Certainly it is a lot better than doing the processing at 16 bit, but you are still sacrificing those least significant bits for no particular reason.
post #8 of 12
Quote:
Originally Posted by Rempert View Post
But if you have 32 bit audio after processing, and you have a 24 bit soundcard, why would you choose to dither or truncate down to 16 bits? Certainly it is a lot better than doing the processing at 16 bit, but you are still sacrificing those least significant bits for no particular reason.
Good point, I clearly didn't think this through to the end (and don't have any 24bit equipment :P).
post #9 of 12
The question is, is the difference percievable? For 24 bits, Foobar doesnt offer dithering, probably because its far beyond human perception. So it just lops off the LSBs in 24 bit mode. I do however agree with your "if you have it flaunt it" method.
post #10 of 12
Quote:
Originally Posted by Alleyman View Post
The question is, is the difference percievable? For 24 bits, Foobar doesnt offer dithering, probably because its far beyond human perception. So it just lops off the LSBs in 24 bit mode. I do however agree with your "if you have it flaunt it" method.
Human perception in this case is somewhat relative. The more your DSPs are attenuating the signal, the easier those LSBs will be to hear. But at some point you hit your equipment's noise floor. I think with typical settings and reasonable volume levels, most people would not hear any difference at all between 16 bit and 24 bit. But I would say the same thing for DirectSound vs. ASIO, and probably get burnt at the stake for it around here.
post #11 of 12
Im with you on 'DS vs ASIO' too. Its not worth the extra CPU cycles for ASIO to me when I can't hear a damned difference. Those who can, well, more power to them .
post #12 of 12
Quote:
Originally Posted by Alleyman View Post
Im with you on 'DS vs ASIO' too. Its not worth the extra CPU cycles for ASIO to me when I can't hear a damned difference. Those who can, well, more power to them .
It matters as long as your soundcard cannot output bit-perfectly through DS mode, since then all your equipment upgrades will be hindered by some low-grade upsampling algorithm that the soundcard produces (e.g. most Creative soundcards ).
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