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Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Recent Posts
2
WavPack / Re: WavPack 5.6.4 Has Multithreading Option
Last post by shadowking -
Thanks for the explanation David.  Fortunately this sample responds well to -hx6 or  better -hhx6 .  Except DNS
 and even -s0 doesn't work too good .  I agree -s1 is something that always stood out for me . Not really bad but a bit
too correlated to the signal . -s.5 is more natural and 'spread out' not as much as -s0 but enough to go undetected and even then still sounds good.
3
Support - (fb2k) / Can't update
Last post by Retrowave -
I have ver. 2.1 32bit installed and want to update to the latest.
I download the .exe from the website, double click it and... nothing happens. I tried running it is admin mod but... same.
Am I doing something wrong?  :-\
4
Listening Tests / Re: Great killer sample, easy to ABX on most codecs
Last post by shadowking -
@synclagz you are right (i proposed similar bitrate and settings) but I ended up on an alternative path for now.
I did a long abx session last night and after hearing the increased security that the heavy settings bring I went that way.
I wrote a similar post in the past about maximising quality without blowing bitrate. It works similar here too.
Higher -x and  -h levels = decreased motivation vs totally 100% transparent.   That became my new goal.

I arrived at two settings and its a quality 1st approach 'while keeping bitrate constrained ' to save space . Some speed is sacrificed if needed for both encoding and decoding (esp for encoding).  I don't see it as lowering quality to keep bitrate down (hopefully I succeeded to an extent in the goal)- but keep quality high and bitrate low as possible while :

a) decreasing likelyhood of finding problems
b) even if found its impact will hopefully be limited and harder to confirm for multiple listeners.
c) decrease chances of feeling need for higher bitrates & less efficiency. I want to save half or more lossless bitrate.

So the settings
1-   A behemoth heavy duty   -b384hhx6s.5  used with or without correction files.
2-   A fast encoding method with practically tiny impact on quality  -b384hhs.5c

The alternative -b metric is 4.35

I guess this could be adopted to the -h mode with -b400hx5-6s.5 or maybe a bit higher.
6
3rd Party Plugins - (fb2k) / Re: Playlist-Tools-SMP
Last post by paregistrase -
Nothing has changed, except the menu name xd I mean, file being in use is either a Foobar thing or your OS or your HDD being too slow. And it may happen randomly.

I can adjust the timeouts between tools to give more time between them, but that's totally random anyway. I could allow the user to set the timeouts though, in case the standard ones don't work.

I think that could be the lyrics, I change to write at 1/3 and the file in use errors disappear.
9
3rd Party Plugins - (fb2k) / Re: Playlist-Tools-SMP
Last post by regor -
Nothing has changed, except the menu name xd I mean, file being in use is either a Foobar thing or your OS or your HDD being too slow. And it may happen randomly.

I can adjust the timeouts between tools to give more time between them, but that's totally random anyway. I could allow the user to set the timeouts though, in case the standard ones don't work.
10
MP3 - Tech / Re: Clipping in MP3
Last post by fooball -
I'm sorry, you used a lot of words that are difficult for me to put together...
Let's see whether I can explain...

In the digital domain, the waveform is just samples.  Reconstructing the waveform into continuous analogue involves estimating what the signal was which resulted in those samples.  The estimate will be perfect if there were no frequency components in the original waveform greater than half the sampling frequency.

If an input with 0dB peaks is sampled, the only digital samples at 0dB (ie full scale) will be those where the sample point happens to coincide with the peak of the input waveform.  Mostly, the samples will be either side of the peak, and therefore at digital values slightly lower than maximum.  If the digital waveform has no samples at maximum, and is then scaled so that the peak sample is at maximum, reconstructing the analogue waveform will require the waveform to go above 0dB.

This is not a problem in the digital data, and it is not a problem in the DAC, but it might be a problem if there is no headroom in the output filter to accommodate an analogue peak slightly over 0dB.

Hence all this vagueness about digital peak and analogue peak.  Digital music players are working in the digital domain where the peak sample value is not necessarily coincident with the peak analogue voltage.  But, except for highly contrived waveforms, there is hardly any difference and any slight clipping as a result would be barely audible (if at all).