Does Bitwig Cause Distortion When Automating Parameters Like REAPER Does?

[limitlesssss]

Apologies, I'm asking this question here since I currently don't have access to Bitwig. I'd like to find out if Bitwig also behaves like REAPER when automating parameters. In REAPER even automating volume causes distortion. I hope this is not the case with Bitwig. Here's the context:

In REAPER:
• Generate a 997 Hz sine wave
• Insert REAPER's own Frequency Spectrum Analyzer Meter on the master
• Change the analyzer's floor to -180dBFS
• Hit play
• You should see a pure sine wave with no distortion

Now (while the sine wave is still playing) simply boost or cut the track's volume by moving the track's own fader, and look at the analyzer again while boosting/cutting, and you'll see* so much distortion happening.

* Now I understand seeing distortion in an analyzer does not necessarily mean that it's audible, but still I'm curious as to why this is happening. And if I had a choice between automation without any distortion and automation that caused distortion, I'd choose the former every time, regardless of the audibility of said distortion.

This also happens if you use third-party gain plug-ins such as Kilohearts Gain, or any other plug-in that has an output gain functionality. I tested many plug-ins from different companies such as Kilohearts, Voxengo, ToneBoosters, TDR, FabFilter, DMG, and they all behaved the same way.

But if you use REAPER's own FADE functionality at the end of your sine wave track, you'll see that your analyzer shows very smooth attenuation without any distortion whatsoever.

Now I admit, I though maybe it had to do with the windowing functionality of the analyzer, so I changed the windowing from Blackman-harris to other options, and when Rectangular was chosen, the result of the volume boosting/cutting had less distortion depending on which company's plug-ins I was testing.

But the question remains, why is it that the FADE feature of REAPER introduces NO distortion at all, regardless of what windowing you choose in the analyzer (even when Blackman-harris is used), but simply moving a volume fader or a gain plug-in introduces so much distortion when automated?

The bigger question for me is:

Is automation inherently flawed in digital? Or is there a DAW out there whose automation behaves the same way REAPER's fade in/out functionality does? In other words: Is there a DAW out there that does not introduce any distortion at all when automating its parameters?

I would also love to know how Bitwig fares in this regard, since its whole claim to fame is its ability to automate/LFO everything.

[limitlesssss]

Also these are Fabien and David Rick's responses on GS: I did what Fabien suggested but the block size does not go smaller than 4 samples in REAPER. Even in Audio Devices when buffer and samples are set to 1, this problem still remains.

I would love to know how Bitwig behaves in this regard. Please enlighten me. Thank you.

This is normal behaviour. Not too many people are aware of it.
Every host handles this differently. You'll also likely experience difference between plugin formats, various host settings, and whether this is offline rendered or not.

Audio plugins (much like your DAW as a whole) are processed in a block-wise manner. They don't eat sample by sample, but take a block of samples at once. That's exactly what the audio buffer size defines. This packaging allows them to be aligned much more efficiently in memory, and thus calculated much faster, e.g. in realtime. All automations and control parameter update in between every audio buffer, i.e. whenever the output buffer is written. Not on every sample. Some hosts even do this for their own audio processing, i.e. their own gain and EQ automations.

Plugin devs can partly smooth all external control over and beyond the audio buffer size to avoid all-too obvious distortion (when the host decides to really just update the control params once per audio block), but the problem is largely outside their control. Usually more about the host dev's considerations on CPU performance.

This is the historical default, but some plugin formats and hosts adopted more advanced methods. FL for example uses a dynamic internal audio buffer, which occasionally switches down to 1 sample during automation (to make it smooth whenever necessary).

Do you see the same happening when rendering in reaper? Check Reaper's rendering preferences for something along "Block size to use when rendering" (try setting it to 1)

Ah. You're seeing Amplitude Modulation products from the gain automation. The sideband structure is determined by the Fourier Transform of the modulation signal. Some DAW programs only do gain changes at block edges ('cause it's easy to program). That means you'll get stair steps in the gain control signal. Square waves have lots of harmonic components, which sprays crap to either side of the carrier (audio signal) as shown in the picture you linked. More sophisticated DAW's do sample-by-sample gain interpolation, which lowers the modulation bandwidth. Of course you could still write a step change into the automation data, but some DAW's will slew-rate limit that edge to reduce its bandwidth.

It turns out that this same effect can be observed in automated consoles. I remember reading an interview with a console designer (probably from Neve or SSL) who explained that he'd had to add a "de-zooting" circuit to keep that from happening. He didn't say what that circuit was, but I suspect it was some kind of LPF after the gain control DACs

[BobDog]

I have to ask, who are “Fabien” and “David Rick”?

[] Peter:H []

Sorry to break it to you, but Amplitude Modulation does add distortion artefacts.
Even aggressive compressors can do it
A 80 Hz sine has a wavelength of 12.5 milliseconds.
If your compressor attack is below 12.5ms, it will distort that sine cycle. if you combine it with low release you can distort it really aggressive.
Why should this not happen if the amplitude is manipulated by automation? So it starts with the nature of AM, therefor Bitwig will give you distortion artefacts in certain situations as well.

[limitlesssss]

I have to ask, who are “Fabien” and “David Rick”?

David Rick is a very knowledgeable gentleman who shares his knowledge on GS forum. Fabien Schivre is Tokyo Dawn Records (TDR)'s lead developer.

[limitlesssss]

Sorry to break it to you, but Amplitude Modulation does add distortion artefacts.
Even aggressive compressors can do it
A 80 Hz sine has a wavelength of 12.5 milliseconds.
If your compressor attack is below 12.5ms, it will distort that sine cycle. if you combine it with low release you can distort it really aggressive.
Why should this not happen if the amplitude is manipulated by automation? So it starts with the nature of AM, therefor Bitwig will give you distortion artefacts in certain situations as well.

Okay, I generated a 0.01 Hz sine wave that takes around 1 min 40 seconds to complete only one cycle. I added a fade-out, that was shorter than half its cycle, and as you see, it still does not introduce any distortion. If we consider fades some form of amplitude modulation, how is it then that an amplitude modulation (in this case: fade-out) that is shorter than one cycle of a waveform, still does not introduce any distortion?

[Trancit]

Okay, I generated a 0.01 Hz sine wave that takes around 1 min 40 seconds to complete only one cycle. I added a fade-out, that was shorter than half its cycle, and as you see, it still does not introduce any distortion. If we consider fades some form of amplitude modulation, how is it then that an amplitude modulation (in this case: fade-out) that is shorter than one cycle of a waveform, still does not introduce any distortion?

My guess would be:
Everytime you do something "live" with the volume on a signal while the sequencers is playing, it does distort the waveform...
Every form of signal manipulation over time is some kind of "Waveshaping" ... there is imho no other way... other than smoothing it out over a longer period of time which would go really into the noticeable "laggy" realm

A fade on the clip though is probably precalculated as it´s a permanent setting on the clip so it´s not happening in realtime = no "waveshaping" = clean to read for an analyzer

And now we are coming to the main part:
1. is there any way to make this "distortion" hearable?
2. Ist it even there or just an artifact of how analyzers work with there analysis windows?
3. If it´s there, is it probably less "distortion" but more some kind of "Jitter" of the moved parameter not at least because of how automation in any DAW works (not per sample but per buffer... could be another reason why it´s not happening on the (precalculated) fade out)?

I think the problem is perhaps a mixed bag within those and probably not worth to break your head upon...
If you can´t hear it... and I bet you don´t... why worrying??
There are for sure much more and much intense negative side effects of audio manipulation happening all the time than a tiny bit of "waveshaping" far below the hearable range...

[Benedict]

Is there any distortion in the audio signal when you render and open in a decent player - one would assume that even Audacity will do this fine and has that advantage of visualizers?

The reason I ask is that maybe there is never distortion in the audio per se (past what is already advised as adding a level curve is actually AM). If not then I think you are taking the eyes over ears approach and the signal is fine but the readout can't keep up and shows visual distortion.

[limitlesssss]

My guess would be:
Everytime you do something "live" with the volume on a signal while the sequencers is playing, it does distort the waveform...
Every form of signal manipulation over time is some kind of "Waveshaping" ... there is imho no other way... other than smoothing it out over a longer period of time which would go really into the noticeable "laggy" realm

A fade on the clip though is probably precalculated as it´s a permanent setting on the clip so it´s not happening in realtime = no "waveshaping" = clean to read for an analyzer

Here's the same sine wave where its volume is being automated. You can see the distortion. Volume automation can also be considered “pre-calculated” like fades, but it still introduces distortion.

And now we are coming to the main part:
1. is there any way to make this "distortion" hearable?
2. Ist it even there or just an artifact of how analyzers work with there analysis windows?
3. If it´s there, is it probably less "distortion" but more some kind of "Jitter" of the moved parameter not at least because of how automation in any DAW works (not per sample but per buffer... could be another reason why it´s not happening on the (precalculated) fade out)?

I think the problem is perhaps a mixed bag within those and probably not worth to break your head upon...
If you can´t hear it... and I bet you don´t... why worrying??
There are for sure much more and much intense negative side effects of audio manipulation happening all the time than a tiny bit of "waveshaping" far below the hearable range...

These are great questions. To address your first question, I would quote myself from the GS thread:

Now I understand seeing distortion in an analyzer does not necessarily mean that it's audible, but still I'm curious as to why this is happening. And if I had a choice between automation without any distortion and automation that caused distortion, I'd choose the former every time, regardless of the audibility of said distortion.

I like your second question very much. Probably someone with access to Matlab can answer that question.

[limitlesssss]

Is there any distortion in the audio signal when you render and open in a decent player - one would assume that even Audacity will do this fine and has that advantage of visualizers?

The reason I ask is that maybe there is never distortion in the audio per se (past what is already advised as adding a level curve is actually AM).

I sure hope you're right and this is just the inability of analyzers to keep up. I think someone with access to Matlab and the knowhow can answer these questions better.

If not then I think you are taking the eyes over ears approach and the signal is fine but the readout can't keep up and shows visual distortion.

I certainly am taking the “eyes over ears” approach with this one. As I said in the previous post, I'm not claiming that I can hear the distortion. But, if I could avoid it (however inaudible it might be), I'd certainly choose that route. I'd certainly want to avoid this distortion if possible, even though it probably is inaudible.

[Benedict]

I certainly am taking the “eyes over ears” approach with this one. As I said in the previous post, I'm not claiming that I can hear the distortion. But, if I could avoid it (however inaudible it might be), I'd certainly choose that route. I'd certainly want to avoid this distortion if possible, even though it probably is inaudible.

That right there is avoidance/obsession and leads nowhere good. Music is not on that road.

People spend stupid money on a Neve because it distorts. All recorded audio is distorted in some way from being recorded. Even having that piano poker poking the piano is distorting the sound of the piano with his big head and shiny shoes.

If were to waste time, this would be full of not-perfect-ness. But who gives a flying bananpeel as it is compelling. Perhaps in great part because of all those things.


Recorded music is never about a perfect reproduction (that is marketing), the recording and mixing engineers are there to build the illusion that this performance is happening. This is barely sticky taped together but it feels like a great thing as it happens. All lies. Lovely lies distorting reality so that I am happ-ee when I hear it. Even when I think about it.

[dayjob]

oh boy. you're gonna love the mp3 format.

[BobDog]

The first thing to do is test your test equipment.

[pdxindy]

The first thing to do is test your test equipment.

Or just keep making music

[BobDog]

Well there is that as well

I was just trying to say if you are measuring something make sure you understand the range and accuracy of your measurement tools.

If you are a musician then I don't understand why you would be looking at this at all!