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Interesting.  I've never noticed this issue using the "Resampler (Sox)" component in foobar2000.  Typically, I'm converting FLAC 24/96 or 24/192 to 16/44.
I guess we came across that problem more than once. Better always check the gaps.
I remember bandpass once suggested to use shorter filters so not using VHQ for no reason to minimize this effect.
General - (fb2k) / Re: New HDD and Mass lossless conversion
Last post by wcs13 -
Some news report : my batch conversion is finally taking place as we speak !

As told previously, I'm using foobar's built-in converter to copy/convert my 37.000 lossless files to FLAC 1.3.2 in a new HDD, including copying all non-FLAC files to the destination folders (*.* trick). Preserving of course all tags, ReplayGain, etc.. I ran two previous tests on a small number of files that seemed to work, so Fingers Crossed now.

Once the conversion is over, I will use the latest version of the binary comparator to see if everything's been copied as expected.

Now I just have a question about it (for @Case since he suggested it ;) ) : the binary comparator manual says you need to create a single playlist containing all the "old" files, immediately followed by all the "new" files. Then the binary comparator magically "cuts" that playlist in two perfect halves and compares all the "old" files to the "new" files. Does this mean that I need to create a single playlist with 37.000 + 37.000 = 74.000 files ?  :o  Is that really the way to go ? I was expecting something a bit more intuitive, like a dialog asking us to enter a "left directory" and a "right directory" in order to compare their contents, lol
Attached is the corrected binary version of the 'mild 4 band expander' project (esp good at undoing mild multi-band side-chain compressors. (as I had promised).   I have talked about the sources elsewhere (forgot what I have said here -- been chatting at a lot of places.)   One place for the source code is on:

Please contact me here (or email -- if preferred):

Be good, safe, and enjoy.

If transition between source tracks is gapless, you can get audible clicks between tracks after resampling.
Interesting.  I've never noticed this issue using the "Resampler (Sox)" component in foobar2000.  Typically, I'm converting FLAC 24/96 or 24/192 to 16/44.
-L (linear phase response) is used by default.
-v (very high quality) seems like overkill for 16 bit, -h (also a default) should be enough:

QualityBandwidthRej dBTypical use
-hhigh95%12516-bit mastering (use with dither)
-vvery high95%17524-bit mastering

Also, regarding dither, by default it uses TPDF noise, it needs -s option to use noise shaping. I can't say which is better or if I would notice the difference between the two (or even if I would notice a lack of dither :-)), certainly not with vinyl rips.
Audio Hardware / Re: AK70 MkII - Real World Improvement?
Last post by iphoneman -
I thought that getting loud enough was not the way these things are analyzed.

It might be loud enough, but if the amp of a unit is being driven to its max, then perhaps some distortion is creeping in.
Audio Hardware / Re: AK70 MkII - Real World Improvement?
Last post by eric.w -
The main things to look at are:
- do your headphones get loud enough?
- is there an audible noise floor? (this can be a problem with sensitive in-ear headphones)
- is there a high output impedance? (which can mess up the headphone frequency response)

The AK70 specs say it has a 2 ohm output impedance. That should be fine with 16 ohm (and up) headphones.
Aside from that, if the max volume and noise floor are satisfactory for you, I don't think there will be anything to improve on.

(P.S. - And should I be using a portable amp with the HD600?)
The only reason to is if they don't get loud enough. I use mine out of a macbook pro and it's just loud enough for me (I put the volume at 100% volume very rarely, for quiet / classical music).
That's not possible with Default UI AFAIK, but it can be done using Columns UI (not 100% sure).

Btw, this thread belongs in the foobar2000 forum not HA's off-topic. There you will receive better advice.
@Viniman, why are you still using the old v0.9.1 foobar2000 ?  :o
If you upgrade your foobar you'll be able to use Playback Statistic version 3 (was released on 2011-07-13). There's no lag time on version 3. Version 3 introduced new data pinning scheme :  "Playback statistics are now pinned to a combination of artist + album + disc number + track number + track title information, contrary to pre-3.0 versions which would pin data to file paths".
I don't use SoX, but I agree.   Your settings are fine (for both the digital and digitized vinyl).

all this so I can free up space and still have good quality for archival.
WAV files?   FLAC will cut your file size almost in half (losslessly) and tags (metadata) is better-supported.

It's very unlikely that you'll hear any very-slight and very-short duration clipping caused by re-sampling, but still, it's good practice to guard against clipping. Other processes, such as EQ or mixing are more-likely to boost the peaks/levels and that's where you have to be careful about clipping.   The vinyl may not even be 0dB normalized so there may be some headroom in those files, and they probably aren't digitally-limited so if they are normalized there should be fewer 0dB peaks to begin with.

Dither is generally also good practice when downsampling.   But, at 16-bits or better it's not audible under any normal/reasonable conditions so practically speaking, it shouldn't make any difference.    And since dither is noise, it's not necessary with (already noisy) digitized vinyl.   Theoretically, you wouldn't want to add any noise to vinyl.  But, the dither is very low level compared to the existing vinyl noise so it won't do any harm.