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.
Clipping from resampling is generally inaudible. If you use guard against clipping and convert tracks separately, it can change loudness difference between tracks in album.
If transition between source tracks is gapless, you can get audible clicks between tracks after resampling.
Example. 2 source files 96/24 with gapless transition - https://www.dropbox.com/s/hvwc9mx4h77afur/96-24%20gapless.zip?dl=0
Converted to 44/16 with sox (sox.exe" %%A -b 16 "%~dp0%%~nA--.wav" rate -v 44100 dither) - https://www.dropbox.com/s/l0l0ynkyb1ga1nh/44-16%20click.zip?dl=0
Between 44/16 files there is clearly audible click. Solution for this can be merge files before conversion, then split after conversion.
Last post by LigH -
So it might have been a mistake of e.g. accidently copying from an extracted QTFiles directory into an existing QTFiles64 ... OK, surely a helpful analysis.
P.S.: Apparently the used makeportable batch was too old (in 2014, it could only handle 32-bit installers; the 2015 update supports iTunes 64-bit correctly).
Last post by nu774 -
But another user reported in the German doom9/Gleitz board that he gets "ERROR: 193: CoreAudioToolbox.dll" when running "qaac64 --check" with the "makeportable" results of different more recent iTunes64 installers; only an older version from 2015 seems to work for him. Could you explain what this error is supposed to mean?193 is ERROR_BAD_EXE_FORMAT.
This error happens when qaac64.exe can find only 32bit CoreAudio DLLs.
If the user has correctly copied QTfiles64 and still see that error, then I don't know.
Scientific Discussion / Re: Have a working 'expander' based on DolbyA (not same design) -- works well.Last post by jsdyson -
The processor code (the da-avx or da-*** program) (the 4 band one) basically (in the actual program) does 32bit floating or 16bit signed integer wav file I/O -- 44.1k, 48k, 96k or 192k. The only reason for the flac/mp3 file conversions is for convienience -- there is also a script that allows playing the music directly in realtime. MY USE OF MP3 ON SIMPLE DEMOS IS DUE TO LIMITED SPACE. I do use the best reasonable conversion params for the MP3. Flac 96k/24bit available by arrangements if you really need. 24bit/96k or floating-point/96k are my native formats for doing work.
So, nothing severely distorting is going on. MP3 on external demo stuff is out of necessity -- but the conversions are usually well above 200k. For convenience, I supplied flac file and mp3 file conversion scripts. If the makemp3 script is missing, then that is my fault -- I have a whole series of scripts that do conversion IN/OUT of the pseudo DolbyA (4 band) processor. I'll check to make sure that the makemp3 script exists -- it is up to you to make sure that you have sox (a very common audio tool) and lame (one of the better mp3 converters.) THE ONLY REASON FOR MP3 over OTHER LOSSY SCHEMES IS IT BEING COMMON. I prefer opus, and usually use it at 258k or 384k. Flac is essentially perfect, and is what I use for high quality archives (usually 96k/24bit.)
Using the program in its most basic mode -- think of this:
1) You have a piece of over-compressed music -- youtube stuff is often a very good example. However, a lot of commercial stuff is also over compressed (refer to the originals on my repository -- directly from the source that I got them modulo a moderate/high bitrate mp3 conversion. Only reason for using mp3 on demos is a lack of space. I can provide people .flac 24bit 96k to anyone who asks.
2) You want something a little less 'obnoxious' sounding, but mostly just want to simple tool without lots of tweaks (the tool has one tweak that isn't terribly sensitive.
HERE IS WHAT YOU CAN DO on LINUX (or if you have a good WINDOWS BINARY):
LInux or some windows configs:
da-avx --thresh=-3.0 <infile.wav >outfile.wav
Windows confs that the above doesnt work:
da-avx --thresh=-3.0 --inf=infile.wav --outf=outfile.wav
The infile.wav would be the ugly inputfile. The outfile.wav would be the new output file.
The output file will have a little less excess ambiance and often have significantly less tape/room noise (on older stuff.)
Notes about the da-*** programs:
* The expander is very gentle -- not much information is really lost unless it is truncated away by using 16bit I/O and ineffective dither.
* Theoretically, the 4 band expander can be undone, but I haven't had to do that, so I didn't bother writing an undo program -- more useful things to do.
* Any of my external use of mp3 is because of limited external storage. I have almost unlimited storage locally, so mostly use 24bit flac at 96k.
MP3s at 48k produced from pristine sources are good enough to see the improvement based upon the recovery/correction software or the simple 4band expander. I understand that mp3 isn't perfect -- I only use it for external compatibility purposes, and normally use the 'extreme' mode preset of lame when producing mp3s. Otherwise, for internal archives where I don't want to use flac, I use opus at the highest reasonable rate. I don't use .wav for archive because it essentially wastes O(half) of the space.
On the archive site I have only provided several unchanged source files -- there are also 'resurrected' versions of those files to be able to hear the difference. If you listen to those without a comparison -- you'll be able to tell that something VERY GOOD is happening in the processing. Almost all decisions are based upon space limitiations. If you want 'my best' copies of something that you are interested in so as to 'better' evaluate -- just let me know
I have some copies that are so good -- they are dolbyA master tape copies. I will not/should not release because I happened into them many years ago when I was working on another thing all together and forget the history.
So -- let me know what will help you, and I'll try to do so.
Last post by knik -
v1.3.1 Beta 1
Unfortunately, nothing new but fixes a few things and updates the docs to be more consistent. It's a beta because it has a few internal changes and I might not have caught everything.
My own "status bar.txt" sample in the "complete" folder was broken by this update so anyone else using it will need to re-import.
Look at this! The new version is even faster!
Last post by dekatch -
ill attach another picture hopefully it gets shown as preview and not just as a link.
would be sweet to get some input on that on how to probably achieve this. because those custom skins are to bloated imo and i really want to black fb2k out