Offline Upsampling

[Jud]

In case you're old enough to remember playing vinyl, obviously I'm too young then I guess those old guys would spend a good 5 to 10 minutes cleaning the disk and stylus professionally before being able to play just 20 minutes of music. In today's world of instant gratification, have we all become to used to having it now?

 

 

I was *much* faster than that at cleaning the disk and stylus! (You'd have to be crazy to do the full bore cleaning every time. I wasn't that insane, just did the full cleaning the first time before playing the record, then used a carbon fiber brush the rest of the time.)

[Jud]

 

Still have, but if now offline upsampling filtering is for the better to you guys I'd like to grab some opportunity just because XXHighEnd can do these things right out of the box

I can do it with Arc Prediction filtering just the same but this is a bit moot because of how all is organized. But if wanted (and you want to wait for it) ...

(but notice there's a 2GB limit for .WAV which (.WAV) is always the result for PCM).

 

 

Peter

 

You know I'd like to see this.

[Jud]

Jud, yes. But I didn't know this thread existed. Edit :

And you know that I know that you know that ...

 

Anyway thanks,

Peter

 

And I know that you know that I know that you... Wait, how many was that?

[Jud]

So I have run into the fact that both WAV and AIFF have a 2GB limit on file size. (A 20 minute classical piece converted to 24/384 ran to 2.7GB. Conversion completed successfully but wasn't recognized by the player.) Other formats not subject to this limitation?

[Jud]

AuI can convert to this format and has knobs to tweak RF64 info in the file; wonder if any player will play it?

[Jud]

I think need test player with RF64. The format has structure like to wav, aiff, dsf as example. But header structure is non full compatible with standard wav and I don't sure what any player capable play it.

Amazing, I clicked the checkbox in AuI to enable BWF, converted to a 24/384 WAV file, and the file played perfectly with Audirvana Plus. I could see from the track loading UI that A+ was treating this single track as two files (one of 16 minutes length, the other the rest of the track) and doing gapless playback between them.

[Jud]

That is amazing. I wonder which other players can do the same?

 

I'll try HQPlayer when I get a moment, unless Miska is already certain it either works or doesn't.

[Jud]

 

Jud,

Interesting info about A+. So as I assume it consider the file as two due internal software structure or really divide to 2 file "on fly». As I right understand, you turn on «Enable BWF» checkbox?

 

Hi Yuri. Yes, I clicked the Enable BWF checkbox and chose conversion to WAV. The behavior of A+ is quite interesting - the file shows as a single track in the playlist. But then the UI indicating loading of the file into memory clearly shows the first 16 minutes loading and playing, then when that's done the last few minutes load into memory and play, but gapless between the two segments so you wouldn't be able to tell this was going on just by listening.

[Jud]

Hi Yuri, this is excellent news! Is there a process for updating registered PROduce-RD version 3.8.5 to 3.8.7?

[Jud]

Would volume control helps me to level volume all my music files? If yes, how do I go about doing it?

 

In AuI Converter there is a "Parameters" setting (look toward the upper left of the application window) that allows you to increase or decrease the volume of the converted file. If you don't already have something that allows you to measure volume, there's an inexpensive app called Audio Tools available for iPhone (perhaps Android too, I'm not sure). I use the SPL graphing function of that app to get an approximate volume at my listening position.

[Jud]

Thanks for the tips! Assuming I get an approximate volume at xxdb, then I would set the volume control at that level and batch convert all my music files?

 

Well, no. The volume at your listening position of - what? The loudest section? An average portion? The softest passage, so you can be sure to be able to hear everything? - a given part of a particular file tells you how loud your system plays that file - we'll call it "File A." Then you play File B, and notice your system plays it 2.5db louder at your listening position than File A. So when converting File B, you set the volume parameter minus 2.5db. This is a one-off process, not batch.

 

If you want everything to automagically play at the same volume without having to fiddle with individual file conversions, you want something called ReplayGain, a proposed standard featured by various software music players. It's something I haven't used myself.

[Jud]

Released AuI ConverteR v.4.1 with advanced DSF meta tags editor. Now AuI ConverteR 48x44 becomes audiophile music manager for control audio files resolution, description DSF and name.

 

DSF meta tag editor - new AuI ConverteR 48x44 v.4.1 | Audiophile Inventory

 

[video=youtube;luG_JQHN-mI]

 

Thanks Yuri. Does this version have filter changes from 4.0?

[Jud]

There is a lot wrong with that article. Not the least of which is it was written at a time when "upsampling" required us of the terrible sounding asynchronous sample rate converters. These days, upsampling and oversampling are essentially the same thing. They have always just been forms of interpolation, it is just that oversampling is always done at multiples of the original rate. With modern s/w or h/w at high rates, the distinction is moot. What matters much more is the quality (parameters chosen) of the digital filter used, the type and quality of dither used (if any), and the ability of the DAC's input to accept the data at a rate that bypasses its own, resource-constrained interpolation filters.

 

Also, the article is wrong to claim that NOS DACs don't produce imaging artifacts down in the audible band. They sure do and they are often quite nasty. I say this as a person who has been using a NOS PCM1704K R2R ladder DAC for many years. But I up/over-sample with s/w to feed it at 352.8 or 384KHz.

 

The quoted portion of the Audioholics article is not great. There is no DAC that is completely "filterless." Believe me, you don't want to hear the unfiltered bitstream. I have, and it sounds like a waterfall of harsh static - a loud, obnoxious "white noise" sound, nothing like music. The digital bitstream, which consists of samples of the music taken 44,100 times per second, *must* be filtered to obtain music. The NOS DACs whose marketing claims no filtering do actually have filters (except the Phasure, which relies on filtering in the PC). For example, the Audio Note kits whose marketing says they have no filtering actually have their circuitry configured so there is a transformer that acts as the analog filter. This primitive filtering, with no upsampling/oversampling/interpolation, results in distortion that some people hear as a "warm" or "exciting" sound. This distortion is so evident that within a couple of years of the origin of the CD player, the industry had settled on a standard of 8x oversampling for DAC chips (44.1 interpolated to 352.8; 48 interpolated to 384) that is still in effect today.

 

There are only a tiny handful of NOS DACs made (Audio Note, Phasure, 47 Labs, Metrum Acoustics, perhaps a very few others). If you don't have one of these, there is a choice of three places to do the 8x interpolation: (a) "on the fly" in the PC; (b) "on the fly" in the DAC; © offline in the PC. If done "on the fly" in the DAC, most DACs accomplish this in three rounds of doubling, 44.1 -> 88.2 -> 176.4 -> 352.8. Also these days, all but this same tiny handful are what are called "sigma-delta" DACs, which "sigma-delta modulate" the input into a DSD-like format at a sample rate anywhere from a few mHz to tens of mHz, and there is the same choice of where to do this part of the conversion process.

 

The choice is also affected by the input rate your DAC allows. If your DAC takes a maximum 96kHz sample rate, then most of the interpolation (from 88.2 to 352.8kHz) and all of the modulation will be done in the DAC, even if you do, for example, interpolate in the PC to 88.2kHz. Or if your DAC has an ESS chip and a max input of 192kHz, then even if you interpolate in the PC to 192kHz, one step of the interpolation (from 192 to 384kHz) and all of the modulation up to 40(!) mHz or so will be done inside the ESS chip. On the other hand, if you have a DAC that allows DSD input, then all of the interpolation and most or all of the modulation can be done in the PC. So how much any external interpolation or modulation will affect the sound is very DAC dependent.

 

The two potential advantages of offline interpolation are these:

 

- You can do it without causing any increased activity or resource use in the PC or DAC while the music is playing; and

 

- The filtering (which must be applied after interpolation) can be more sophisticated, because it can take much more time compared to on the fly interpolation and filtering, which needs to happen quickly enough that you don't notice a pause during playback, or if you do it's brief enough you don't mind it.

[Jud]

Currently testing offline up-sampling to 64 bit float/384k sample rate WAV files (the file sizes are huge!) using r8brain (free) against on-the fly up-sampling using JRiver. It is hard to pick a difference with the tracks I have tried so far.

 

What is the rate you're upsampling to, and what is your DAC?

[Jud]

So, to understand on my side this subject lets assume that the DAC has the clocking as below in the table and the maximum input accepted by USB is 24/96kHz, how to interpret it and how many steps is done for each particular rates up to 96kHz?

 

[ATTACH=CONFIG]16785[/ATTACH]

 

Hi Krzysztof, please remind me what DAC you have. I think the answer in any case will likely be that you would gain fairly little advantage from sample rate conversion in the PC, online or offline, because you will simply substitute one step in the PC for one in the DAC chip. That is -

 

CD rip, no PC conversion, in the DAC you will get: 44.1 -> 88.2 -> 176.4 -> 352.8; then sigma-delta modulation to MHz rates, then conversion to analog.

 

CD rip, PC conversion 44.1 -> 88.2, in the DAC you will get: 88.2 -> 176.4 -> 352.8; then sigma-delta modulation to MHz rates, then conversion to analog.

 

CD rip, PC conversion 44.1 -> 96, in the DAC you will get: 96 -> 192 -> 384; then sigma-delta modulation to MHz rates, then conversion to analog.

 

Because of so many conversions in the DAC even if you do the first one in the PC, I think you will likely hear little difference. That is of course speculation on my part, so if you do the conversion and hear something better out of it, great.

[Jud]

Hi Bill. Many of those are professional software suites, some fairly old. It is also a good idea to know something about what the tests are and their purposes.

[Jud]

copy_of_a:

 

I have seen references to izotope src but i can't seem to find information on it on their website. Is this part of a larger package that is sold by izotope? If so, would you point me to it?

 

rx4.

 

Edit: I keep trying to make those letters upper case and the "helpful" auto spell check keeps changing them back when I save the edit.

[Jud]

 

On my machine (2009 MacPro 8x2.26GHz Nehalem & 24GB RAM) AuI behaves really strange. When batch converting multiple files either AuI crashes or it causes other applications simultaneously running to crash.

 

AuI utilzes 100% of all cores (which is good!) but it doesn't play well together with other softewares also utilizing mulitcore processing. Normally softwares utilizing heavy mulitcore usage should balance each other (at least that's how my other softwares behave ... they "share" the multicores; they do not block all the cores for their own processing).

 

Very strange - have noticed no problems on my mid-2009 MacBook Pro (2.53GHz Core 2 Duo, 8GB RAM).

[Jud]

Understand now c of a, thanks.

[Jud]

Yuri develops his program at a rapid pace, so I think a strategy of wholesale processing of large parts of your collection is bound to end in tears.

 

An alternative you may want to consider is to convert what you plan to listen to, and perhaps get rid of it later so it doesn't take up space. In other words think of the originals as permanent but conversions as temporary.

[Jud]

An important part of the point for me is to provide versions to add to my library with consistent and adequate metadata of albums I have bought in dff and ISO. I wouldn't want to delete them after all the effort to convert and tag them.

 

While I realise that future improvements in Yuri's program may encourage me to redo them at some point and batch copy the metadata from my previous versions using Yate, the provision of true gapless is a basic necessity for including them in my library, or playing them at all, in the first place. So for the moment I am condemned to this Vale of Tears, or in other words, I don't see an alternative to investigating how long it will take to redo them with Yuri's new version.

 

Right, think I'm understanding better now (seems leaping to conclusions is something I'm doing fairly often in this thread).

 

So what you're primarily doing, at least at this point, is converting dff and ISO to dsf with proper metadata?

 

If that's the case, then yep, once there are no more clicks left when converting, it seems to me that the dff to dsf and ISO to dsf conversions would be lossless (as long as you don't change the sample rate), and you could treat the resulting dsf files as originals.

[Jud]

Thanks Yuri, look forward to trying it tonight. Does 3G for "all filters" include minimum phase?

[Jud]

I have a thought but no time - will write later tonight.

[Jud]

I have a thought but no time - will write later tonight.

 

...or, as it happens, the next day.

 

So Yuri, this is my idea, which I think could be fun and educational, but may be a lot of work for you.

 

You could make three builds as public demos, and simply call them A, B, and C. One would be the same as the current AuI; one the intermediate option; and one full band. Don't identify which is which. (Of course it will be easy to tell if anyone wants to measure.)

 

Whoever wants to play along would tell you what DAC they have; what the original and upsampled resolutions were of a couple of tracks they listened to, at least one from 44.1x resolutions and one from 48x; which build they preferred; and how strong the preference was (whether they liked one much better or all were close).

 

That is the general outline. What do you think - interesting, or too much work?

[Jud]

Hi Yuri,

 

I essentially upsample to 24/192 a directory of files which is the results of my CD library ripping with XLD. Most of files are 16/44.1 in AIFF format. Extensions are .aiff on the Mac. From the finder, you can check the icon to see that artwork is not included in the file, as it was in the original.

 

I have a very similar workflow, upsampling mostly to DSF, though I have done AIFF, also very often from 16/44.1 XLD CD rips. I will check to see whether artwork is transferred. I don't bother to look with DSF conversions from PCM, since I don't think artwork is brought over with my SACD rips, so I am accustomed to not seeing artwork with DSF files.

 

I am using 5.4.1. But the metadata tag is not identical to yours. Files are not copied although the box is checked. Is the size limit causing the problem ?

 

I use 5.4.1 as well. I'll check to see what is happening with my files.

 

 

I will PM you to sort out this directory problem.

 

I know I don't have this problem. I've left my settings on default, which puts the files in a subdirectory of the original folder named "ConvertResult."

 

Thanks for the settings suggestions.

 

I use these settings and have been very happy with the results.