To crossfeed or not to crossfeed? That is the question...

[71 dB]

Don't want to be "grammar nazi", but mHz = 0.001 Hz and MHz = 1 000 000 Hz

 

[WoodyLuvr]

@gregorio very informative post; thank you for taking the time to write that as I learned a lot! Cheers.

 

[Zapp_Fan]

Please do not ascribe to me some primitive ideas as if I don't understand that converting the wordlength from 16 bit to 24 bit "recreates" extra 8 bits (or that upsampling "recreates" some frequencies). I may have some misconceptions but not as stupid as that.

Let's talk about the wordlength (bit rate) first.

I do understand that these extra bits will be "padded" with zeros. But only initially, because, as we apply more and more processes to audio, these zeros will be quickly replaced with figures other than zeros. After several processes even 64 bit wordlength will probably be not long enough to accurately represent the full result of all computations without rounding it off.

Do you agree that (from the technical, mathematical point of view - let's forget for now the debate whether we can hear it or not) after 16-bit audio is heavily processed by 32-bit or 64-bit plugins, the 24-bit representation of their final computational result will be more accurate/complete than the 16-bit representation? If so, do you agree that it's better to send out to the DAC the signal in 24-bit format rather than 16-bit?

I have to chime in here because 32/64 bit floating point noise is something we can't afford to start worrying about, whether or not it is technically more accurate. Also, any reasonable VST host will be passing 32 bits between plugins before converting back to the output bitdepth at the end, so whether you change bit depth before the processing stage is irrelevant. Still, probably, using 24-bit output is technically superior to 16-bit output, if any component in your system actually has that kind of noise performance... which is doubtful.

 

[Whazzzup]

well i use minimum setting crossfade on my TT, sometimes no crossfade

 

[WoodyLuvr]

well i use minimum setting crossfade on my TT, sometimes no crossfade

Is crossfade similar to crossfeed?

 

[castleofargh]

here is a hint, they are different words. like crosstalk, or crossfit. ^_^

 

[Whazzzup]

Yes feed

 

[WoodyLuvr]

LOL

 

[Strangelove424]

A crossfade is a kind of audio transition, maybe he got confused.

 

[ironmine]

1. You appear to be confusing two different things here: Anti-alias and anti-imaging filters. To comply with Nyquist, we have to remove all freqs above the Nyquist Point to avoid aliasing. With oversampling our Nyquist Point is far higher, this requires a much simpler, cheaper and less damaging analogue filter and also spreads the dither noise over a wider band, much of which is then discarded when a secondary (digital) decimation filter is applied. This is why pro ADCs oversample into the multiple mHz range, commonly somewhere around 22mHz. However, none of this is applicable in our case because what we're dealing with has already been anti-alias filtered and there is nothing above the 22.05kHz Nyquist Point of our 44.1/16 input file/signal!
1a. What Bob is talking about was absolutely true, for a number of years. I sometimes used to run sessions at 96kHz as many plugins operated audibly better at that rate, many/most soft-synths, compressors, some EQs and various others. There is now (and for quite a few years) no benefit to this as the processing power available to plugin developers has increased significantly and the coding is more sophisticated. Most plugins no longer benefit from a higher sample rate and those few which do (such a non-linear analogue modelling plugins), now simply up/down sample where necessary and can apply better filters when doing so. I no longer run sessions higher than 44.1 or 48kHz for audio quality reasons, I do so only if a client requires delivery of a higher sample rate.

2. In your "Step B", if we were to make that assumption, then what you've asserted might have some merit. However, you cannot make that assumption! Those plugins which do legitimately upsample would do so at x2 (either 88.2 or 96kHz) as any theoretical benefits of filters and any non-linear processes are perfectly addressed by those sample rates and going higher is just a waste of processing and potentially less accurate. Those plugins with a fixed sample rate (such as some convolution reverbs and some soft-synths/samplers for example) tend to operate at 96kHz. I know of no plugins which upsample to 176.4kHz legitimately and by "legitimately" I mean processors which upsample to that rate for any reason other than marketing. The only exception to this would be plugin processors designed to deal with intersample peaks, such as true-peak limiters, although some upsample well beyond 176kHz.

3. Your last paragraph; if we don't upsample to 176 and the plugins do automatically upsample 176 then yes, there would be fewer applications of anti-alias filters. BUT:
A. If we're feeding the plugin a signal with no content above 22.05 and if the processor is not creating any content above that freq, then we're applying a smoother, higher frequency filter where there is no frequency content anyway, to a signal which already has (whatever) filter artefacts from being filtered to 22.05kHz to start with.
B. All plugins do not automatically upsample to 176. In the case of a plugin with a fixed sample rate of say 96kHz then: You upsample to 176, adding a filter in the process. The plugin downsamples, processes and upsamples again, adding two more filters to the process. That's the application of 3 filters where if you'd just fed the plugin 44.1 to start with, there would only have been two filters applied.
C. If the plugin does not upsample (and there's no reason to, in most cases) then: You're upsampling for no benefit, adding an unnecessary processing step, an additional filter and risking lower precision by operating at a higher than optimal sample rate.
D. If we were to upsample to 88.2, we'd be adding a filter. If the plugin is operating at 176.4, then it has to upsample from 88.2 to 176.4 and add another filter, then downsample to 88.2 and add another filter. If we'd just fed the plugin 44.1 to start with there would only be two filters applied, rather than three.

4. Your fear of up/down sampling and the effects of applying filters to this process is unwarranted. The filters in plugins today are far superior to those of 15 or so years ago and are audibly transparent.

5. In the case of a DAC oversampling to say 352kHz, then going from 44.1kHz to 352kHz is theoretically better than going from 44.1 to 176 and then from 176 to 352. It's one less processing step and filter application, the same as 3d above.

None of the above is absolutely set in stone, as of course it all depends on the skill/effort of the plugin developer.

G

You now argue just to win this debate, it's the logic of a Jesuit, you write contradictory things, without hesitation, depending on what you need to say just to prove me wrong. Examples:

From what you say here: "Most plugins no longer benefit from a higher sample rate and those few which do (such a non-linear analogue modelling plugins), now simply up/down sample where necessary and can apply better filters when doing so."... "Your fear of up/down sampling and the effects of applying filters to this process is unwarranted."

By saying so, you invite us to make the conclusion that upsampling and filtering nowadays is implemented at such high quality level so we don't have to be afraid of any quality loss even when this process is repeated many times over and over again. You imply that upsampling / filtering is either harmless or beneficial.

But in the other paragraphs, you say the opposite: "The plugin downsamples, processes and upsamples again, adding two more filters to the process. That's the application of 3 filters where if you'd just fed the plugin 44.1 to start with, there would only have been two filters applied."

So, now, all of sudden, an increased number of downsampling/upsampling & filtering processes is an ugly thing and we need to minimize it. You imply that upsampling / filtering is harmful.

When an upsampling/downsampling happens inside a plugin, it's good and beautiful. When it is done outside a plugin, it's suddenly bad and ugly. How is that?

When you need to derive my logic of thinking of merit, you don't allow me to make an assumption that plugins may upsample to 176, you write: "In your "Step B", if we were to make that assumption, then what you've asserted might have some merit. However, you cannot make that assumption!"

Several sentences later you write: "If the plugin is operating at 176.4, then it has to upsample from 88.2 to 176.4 and add another filter, then downsample to 88.2 and add another filter."

So now, just to prove me wrong, you are making the very same assumption which you wouldn't let me make earlier!

Seeing these logical flaws in your thinking, I do not find your arguments convincing.

I still think that upsampling helps plugins do their job better. Bob Katz and Aleksey Vaneyev (the author of highly respected Voxengo plugins) think the same. To my knowledge (based on the plugins which I have, and whose manuals I've read), if audio is 44 or 48, most plugins will offer x2 oversampling to improve accuracy or will do it even without asking. They may not oversample 88 or 96 to 176 or 192, because 88 and 96 are already good enough, but, most probably, they will oversample 44 or 48 to 88 or 96.

So, if you need to process audio that is 44 or 48, I advise to do at least X2 upsamping of the signal before you feed it to the VST host for serious processing. Better do it yourself, once, in the upsampler (dBpoweramp/SSRC) whose quality and precision have been confirmed to be top-notch, rather than trust plugins do it themselves, probably many times over, in who knows which haphazard way. Remember, that your 44 kHz or 48 kHz signal will be taken in your oversampling DAC anyway to 352 or 384 kHz or even to MHz whether you want it or not. So, if your have to do heavy audio processing and upsampling is anyway inevitable, ask yourself this question: where do you want this processing to happen - BEFORE or AFTER upsampling? For me the choice is obvious. I prefer to do it before. This is what I do - I usually upsample x2 (to 88 or 96) and the result (sound), to my ears, is great (but a bit different from both 44 and 176).

You can find my arguments more convincing or Gregorio's arguments more convincing, I don't care. If that's all you have to say, Gregorio, i am still unconvinced, sorry. But I read your messages with interest and look forward to them. Maybe I am not smart enough to connect all the dots and see the light of truth.

 

[71 dB]

A crossfade is a kind of audio transition, maybe he got confused.

The used of the cross… terms has been a little messy in this thread including myself I'm afraid.

Crosstalk => Unwanted. Happens electronically, audio channels leak to each other.
Crossfeed => Wanted. Happens electronically or acoustically.
Crossfade => One audio clip fades out as another fades in. Unrelated to crossfeed and crosstalk.

 

[gregorio]

You now argue just to win this debate ...

Actually, I'd say that's exactly what YOU appear to be doing! Making up a whole bunch of my supposed implications, taken out of context, to suit your argument. For example:

Several sentences later you write: "If the plugin is operating at 176.4, then it has to upsample from 88.2 to 176.4 and add another filter, then downsample to 88.2 and add another filter." - So now, just to prove me wrong, you are making the very same assumption which you wouldn't let me make earlier!

What assumption have I made? I have stated what would happen given the circumstances YOU described, not that plugins do operate at 176. In fact, later in the post I specifically stated "I know of no plugins which upsample to 176.4kHz legitimately".

you invite us to make the conclusion that upsampling and filtering nowadays is implemented at such high quality level so we don't have to be afraid of any quality loss even when this process is repeated many times over and over again. You imply that upsampling / filtering is either harmless or beneficial.

Nowhere do I imply filtering is beneficial and I'm also NOT implying that up/down sampling is always harmless, just as I did not imply there is no quantisation error with a 32 or 64bit plugin. However, many modern releases will already have had this repeated "many more times over and over again", a consumer on the other hand doesn't! I've done mixes with over 200 plugins, what consumer is ever going to do that? They're going to create a chain of just one or a handful of plugins and NOT repeat the process "many more times over and over again"!

I still think that upsampling helps plugins do their job better. ... To my knowledge (based on the plugins which I have, and whose manuals I've read), if audio is 44 or 48, most plugins will offer x2 oversampling to improve accuracy or will do it even without asking.

You're free to think whatever you want but you haven't explained why adding trailing zeros one step early "improves accuracy" or how doubling the data and bandwidth, where no frequencies exist, "improves accuracy". Until you do, there is no basis for your advice to others!

G

 

[bigshot]

Crosstalk => Unwanted. Happens electronically, audio channels leak to each other.
Crossfeed => Wanted. Happens electronically or acoustically.
Crossfade => One audio clip fades out as another fades in. Unrelated to crossfeed and crosstalk.

You forgot one...

Cross Purposes => The way people discuss things in Sound Science

 

[71 dB]

You forgot one...

Cross Purposes => The way people discuss things in Sound Science

Crossroad, crossdress, crossword,…

 

[jgazal]

You forgot one...

Cross Purposes => The way people discuss things in Sound Science