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3rd Party Plugins - (fb2k) / Re: foo_enhanced_playcount - Record all song plays and scrobbles (JScript)
Last post by MordredKLB -
This plugin has so much potential!! Unfortunately in my opinion it's too unstable for using it yet. But I'm hoping for further improvements both on the part of the api and on developing the app.
What do you mean by "unstable?" Is it causing crashes? Is something not working correctly? I appreciate the feedback, but you haven't provided me any information on what can be improved.
3rd Party Plugins - (fb2k) / Re: My components
Last post by arch21 -
I just checked, your foo_unpack_jma, foo_unpack_lha, and foo_unpack_unix, they don't register associations.

Thanks for all the answer. Yeah, I also sometimes check your git repo, it has been fixed. Thank you :)
General Audio / Re: Removing Spaces or Gaps between Tracks
Last post by j7n -
Start with a player that works correctly with gapless files, such as Foobar2000. Convert all necessary files to WAV format. Then work on them in any editor you choose. After you're done, compress the resulting files using an encoder. I do it all time time working with older editors such as Sound Forge and CoolEdit that predate gapless standard.

Not sure if it is the same Wave Editor, but the one by Abyss Media, works by temporarily decoding all input files (including WAV or MP3) to a 16-bit file in the temp directory using the BASS library. Files still have the exact duration at this stage. Upon saving as MP3 directly from the program, a Xing header isn't written, and gaplessness is lost. Saving to WAV is fine. It isn't a particularly good editor, because the accuracy is limited to 16-bit, an applying an action to a small section causes the entire file to be rewritten to disk. It seems that most processes can't be applied to a selection.

You could also remove gaps from existing MP3 files without re-encoding and losing quality using the Utilities > Edit MP3 gapless playback information in Foobar. If the option doesn't show, enable it in Preferences > Display > Context menu. This is quite a cumbersome, lengthy process. You need to use a visual audio editor with sample count on the time ruler as reference. Decode all files to wav, see how many nearly silent samples there are at the start, enter that number for all files. Decode them again. See how long each file is, copy and paste the exact duration in the second box. You may need to cut off some samples at the end, and try splicing with the next file. (Yes, it takes a hell lot of time.) The amount of silence that can be removed this way is limited to about 1/10th of a second, maybe less.
CD Hardware/Software / Re: EAC: Defeat audio cache - Yes vs No
Last post by greynol -
It's explained pretty clearly in either of the wiki articles linked in the discussion.  I don't think I can do any better since I authored both of them.  To be honest, I've moved on from discussing ripping.  This topic was low hanging fruit, though korth had already addressed it.  He is one of the resident gurus on ripping.

Your rips will go faster.  Accuracy will not be affected.
3rd Party Plugins - (fb2k) / Re: My components
Last post by kode54 -
My archive readers register associations for all supported formats.

And I already answered this in email, I think. Secret Sauce creates temporary files the same as the plugin it lifts its library from, to facilitate multiple instances in the same process. Without it, you only get one instance globally, 16 channels. And I thought I fixed that MUNT issue already.
CD Hardware/Software / Re: EAC: Defeat audio cache - Yes vs No
Last post by pinkyy -
I read the guide multiple times, if I understood what it meant I wouldn't be asking this question :)
The ripping engine has not changed since long before 2005.

Be happy you have a drive that doesn't require cache flushing.
Thank you! Can you please tell me what does cache flushing mean?
Will a rip be more accurate if a drive doesn't have cache flushing?
Audio Hardware / Re: How to monitor the output band to external DAC
Last post by DVDdoug -
That's not correct. A 24 bit file has 8 additional low-order bits, not high-order.
Yes and no...

If you look at the raw numbers, no...   You can't add more lower-order bits to a 16-bit integer because they would be to the right of the decimal point.    

If the DAC only supports 16 bits then the upper 16 bits are played
If everything is scaled properly, yes...  The maximum is 0dBFS in both cases and with 24-bits the additional resolution is on the low-end.

So my point was...  I think ASIO and WASAPI exclusive will  alter the data to scale the bit-depth properly to match the hardware, even if they won't alter the sample-rate.

The "regular" Windows drivers will alter everything...   I can play that 24-bit 384kHz file with WMP on my cheap motherboard soundchip.
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