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3
Support - (fb2k) / How about the "Library files could not be written" dialog ?
Last post by wcs13 -
Hi, here's the dialog. I get it EVERY TIME I close foobar.
Often I have to click two or three times on "Try again" until it finally works and I can quit foobar properly.
AFAIK I've had it since foobar 1.4 (now I'm in 1.4.1 and the problem persists).
I'm running Windows 7 SP1 x64.
Any thoughts ?
4
Vinyl / Re: 192/32 needed for digitized vinyl to sound as analogue as possible
Last post by old tech -
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However I have noticed on cheaper T/T cart combinations there does seem to be a somewhat artificial widening of the sound stage.  It's hard to describe, almost like the fake surround sound some soundbars put out which of course, DSPs can emulate if that is one's thing.

I have never given this much thought but I presume it is due to phase shifts, which vinyl playback always has but more so with less well engineered or correctly aligned set-ups.  Any thoughts on this?
If the cartridge is wired wrong the left & right channels can be out-of-phase (one channel inverted).   You'll get the same "weird widening" effect if you reverse the wires to one speaker (or if you invert one channel in an audio editor).    You'll also notice a loss of bass as the bass soundwaves cancel, and if you mix-down to mono (electrically or digitally) you'll get a "vocal removal"  effect where the "center channel" information (the information common to left & right) gets canceled..

Thanks for the info but I and many others have noticed this under normal conditions where the wiring and polarity are correct.  However, it does become less noticeable with higher quality cartridges/arms and proper alignment.

Could it just be an artefact of the one stylus trying to track two channels causing phase timing errors?  If so, it would stand to reason that a better engineered cartridge/arm and alignment would bring the sound stage closer to the more precise digital reproduction.
5
Vinyl / Re: 192/32 needed for digitized vinyl to sound as analogue as possible
Last post by old tech -
If I take a look closer to my new Cds (EAC accurately ripped and so on), I can't understand why the "factory default" CD sound is likely flat, louder and way beyond the full scale (0 dB). All right, sounds good for a boombox but for my PC Hi-Fi system it's a little bit more harshly. I have to de-amplify to -12dB and then to make some compression and normalization to -3dB to get an acceptable sound from CD. I tested CDs from various artists, Pink Floyd, AC/DC, Genesis, Alan Parsons, The Band, Fleetwood Mac, Manfred Mann's Earth Band and all sounds are way beyond the full scale (0 dB). Is this a CD standard? Not happend when I record the same LPs from new vinyls to DSD 5.6 MHz and then converted to wav 196/32, 96/32, 48/32 or 44/16.

You're definitately doing something wrong there.  I have CDs of these bands and they sound great, though it probably depends on which issue or remaster.  For example, I have the 1983 Dark Side of the Moon (non TO) black triangle CD which has the same tape source and Sony mastering as the excellent 1977 Japan Pro Use LP, which I also have. Comparing the two side by side they sound almost identical, with the edge to the CD being slightly more dynamic and consistent across the album, ie no IGD as the LP plays towards the centre.  This CD, along with many other early Japan CDs has pre-emphasis, so if you do not de-emphasise the CD rip it will sound a little bit shrill.
6
General Audio / Re: fb2k's foo_hdcd not as good as hdcd.exe?
Last post by kode54 -
Sheesh. I only did that to prevent false positives, which are a dime a dozen when it comes to the HDCD format. If you don't like it, I can just make it always process, and never auto detect, and you'll just notice that all of your lossless formats decode slower than 500-600 times real time, skewing any benchmarks that people love to tout about their machines. Because this decoder is slow, even with these optimizations.
9
Audio Hardware / Re: Objective measurements of portable players using df-metric
Last post by Serge Smirnoff -
Is it even possible to be *audibly* better? (if "better" = objectively better, as in less distortion, flatter FR, etc.)
In case of traditional audio metric (that you mentioned) relationship between perceived audio quality and objective measurements is poor and your question is understandable. In SE audio metric such relationship is more defined, so in some cases we can safely predict subjective quality from objective measurements.

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Still most of audio files that we listen are 44.1.
Not true, with the rise of Opus which basically makes everything 48kHz.
You'll be right when Opus become mainstream.

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Most real-life audio material (especially if it is perfectly mixed and mastered) has natural high frequency roll-off
Most real-life audio material (especially if it is perfectly mixed and mastered) has quite a lot of energy above 20 kHz (that is, between 20 kHz and the upper limit which is usually 22050 Hz), even with roll-off it's still quite a lot; and sometimes people use extreme settings for noise shaping (for 16 bit conversion) which adds to that band too. Humans, though, don't hear that band at all, so taking it into account at all would introduce unnecessary mistakes.
The test set “Variety” (2 hours of music), which is used for df-measurements has the following overall amplitude-frequency characteristic:



Not so much energy between 20kHz and 22.5kHz indeed.
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