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Recently found a bug in a piece of a gain control project that I have been working on.   It shouldn't really be a major issue for most people, and only happens when trying to do something special in a dynamic range control project.  This bug can even happen if you already know that this concept can cause troubles -- but will invariably happen over and over again if more than one layer of gain control filtering is needed.   This fact can make the gain control circuit ( or algorithm) need to be more complicated or tricky than otherwise one might think.  SIMPLE straight single attack/decay TC doesn't cause this kind of problem -- but there one must be
careful about aliasing and/or intermod (depending on the realm of where the algorithm runs.)

The effect of this bug is that the gain control will sometimes sound ragged.  There are workarounds for this problem, while still gaining most of the positive effects...  But whenever you see this kind of circuit, or have the equivalent in software -- beware -- you'll have a harder/harsher sound than what you would expect.   Please refer to the attached PDF, but otherwise if you don't like attachments -- basically it is like this:
raw-level-signal ----- moderate attack/slow decay -- additional fast/moderate RC TC --- further on to gain control element.

I am in the midst of fixing a bug right to mitigate this mistake -- but I already knew about this matter.   Hope this might help any future people working on gain control and not aware of this oddity.

3rd Party Plugins - (fb2k) / Re: Biography Discussion
Last post by jazzthieve -
@WilB I wanted to get 5 similar artists tagged instead of 4 so I tried changing the limit attribute here to 5 but I'm still getting only 4 similar artists.

Code: [Select]
var URL = "" + p.lfm + "&method=artist.getSimilar&artist=" + encodeURIComponent(artist) + "&limit=5&autocorrect=1";

Also, from what I gather from the Lastfm API documentation it seems possible to retrieve artist mbid from Can you include writing those to tags in your next release?
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.

Edit: Wasn't one solution to use "Don't reset DSP between tracks" in foobar? Maybe you have set this and never had to wurry.
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).