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Cabbers producing music

Started by popcorn, October 30, 2018, 12:03:56 PM

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popcorn

I'm a bedroom musician, like a few people here. I used to have loads of stuff on my Soundcloud but I recently took everything off it that I'm not 100% proud of. So one song basically.

I tend to stick my questions about Ableton and the like on Reddit, but I'd much rather have them answered by lovely bald men then anonymous Redditors, so I'm starting this thread. Let's fill it with arguments about midi and USB audio interfaces.

Here's my brilliant, exciting opening question. When you're recording to tape (which I've never done), you can speed up and slow down the tape, which affects pitch. Is this possible to digitally? I mean, I know it's kinda possible, because you can do it, but presumably it's not the same because of the way the sound is chopped up into loads of little bits. It gets artefacty and weird very fast. Can someone who understand com puter explain this better to me?

#1
It can be done quite easily in Audacity (and any DAW I would imagine)

Select (click and drag) the area you'd like to slow down. You can also go "Edit>Select>All" if you want to select everything. Click "Effect>Change Tempo". Then drag the slider to the left however much you want to slow the track down (you can also speed it up by dragging right) and click "OK"

The sound quality won't be affected just because you're in the digital domain, technically it should be better than tape. It's pitch shifting that is generally worse for adding artifacts.

jake thunder

You need a tool that'll slow down / speed up audio without preserving pitch. Some DAWs may have a built in tool to do this. My DAW doesn't, so I have to export the file and use Adobe Audition to change the speed.

Audacity will do the same thing and is free:
https://www.audacityteam.org/

I assume there's some quantisation but I can't hear any artifacts or reduction in quality.

Seems like the best way to do it would be to change the sample rate of a soundcard when recording / playing back, but not sure if that's possible.

I'm not bald btw so could be talking shit.



the

Quote from: popcorn on October 30, 2018, 12:03:56 PMWhen you're recording to tape (which I've never done), you can speed up and slow down the tape, which affects pitch. Is this possible to digitally? I mean, I know it's kinda possible, because you can do it, but presumably it's not the same because of the way the sound is chopped up into loads of little bits. It gets artefacty and weird very fast.

What you're talking about is changing the playback speed of the sample (which also affects pitch).

The easiest way to achieve this effect in Ableton would be:

1) Load your sample into a Sampler/Simpler and use the pitch bend while the sample plays. (Assuming you have a MIDI/USB keyboard with a pitch bend wheel on it. If not, you can draw a pitch bend curve in on the Automation lane.)

Or:

2) If using the sample as a clip instead, change the Warp setting to Re-Pitch. When the clip plays, it will play back at the speed dictated by the song tempo. Changing the tempo will now change the playback speed and pitch (rather than stretching it, which the other Warp settings will do).

Funny how changing the playback speed used to be the norm and timestretching the exceptional, now it's the other way around. :)

Edit: I'm talking about samples within a composition here. If you're trying to change the pitch of a whole tune, yes, you'd be better off doing it in an audio editor such as Audacity.

popcorn

Right, I phrased my question badly.

This isn't actually something I want to do. I know how to do it in Ableton.

I'm just wondering about the technical process behind it. Is there a fundamental reason why altering the speed and pitch of a digital recording will always produce different results to doing it with tape? The reason I ask is that I saw an interview with a producer once who said it was something computers will always be bad at compared to physical tape, and I'd like to know more.

Captain Z

Edit: Already edited/answered above.

Quote from: popcorn on October 30, 2018, 12:43:14 PM
Right, I phrased my question badly.

This isn't actually something I want to do. I know how to do it in Ableton.

I'm just wondering about the technical process behind it. Is there a fundamental reason why altering the speed and pitch of a digital recording will always produce different results to doing it with tape? The reason I ask is that I saw an interview with a producer once who said it was something computers will always be bad at compared to physical tape, and I'd like to know more.

I'm really not a technical head, but I would have thought that a tape would give a lesser quality when slowed down because BITD anything that ran slower (VHS/compact cassette etc) when you had the choice was always lower quality.

Buzby would be the best to answer this I'm sure.

Having said that it is all subjective because in some cases this lesser quality, be it digital or analogue is desirable.

PlanktonSideburns

Is it that digital audio is just capturing a little chunk of audio x amount of times a second, a bit like film, so if you slow it down you start to notice the gaps between the chunks?

Where as analogue signal records the whole thing, somehow , I dunno

the

Someone more knowledgeable than me will cover this I'm sure, but what the producer is getting at is that, when you slow something digital down, you're playing it back at less samples per second. Basically you're playing it back at a lower resolution.

How systems deal with playback or rendering of audio at a lower resolution than it was recorded it depends on how that system deals with dithering, which is a mathematical process. The artifacts may be more or less audible depending on how it's calculated.

An analogue recording will also degrade when played back at a lower speed, but in a different way. Analogue playback is a physical action converted into a electrical signals, as opposed to a stream of zeroes and ones that need to be mathematically calculated and processed in an error-free way to achieve the resultant audio.

NoSleep

Quote from: popcorn on October 30, 2018, 12:03:56 PM
Here's my brilliant, exciting opening question. When you're recording to tape (which I've never done), you can speed up and slow down the tape, which affects pitch. Is this possible to digitally? I mean, I know it's kinda possible, because you can do it, but presumably it's not the same because of the way the sound is chopped up into loads of little bits. It gets artefacty and weird very fast. Can someone who understand com puter explain this better to me?

As you say, the problems with changing the pitch & speed of audio in digital are noticeable when you start to move far away from your starting point, but if you want to half speed or double speed then most DAW will allow a change of sample rate, so you could select to record the audio at an alternative sample rate then play it back at your normal rate.

For changing the speed/pitch to something nearer the original rate I can think of two methods offhand:

1) Using Logic, create a new instrument in EXS24 (the built in sample playback instrument) and then you can tune your sample + or - 2 octaves in increments of a cent. I was also going to suggest using the Time/Pitch Machine but I don't think that has survived into Pro X in that form.

1a) The above will work for Kontakt (or any other soft sampler app), too, I imagine.

2) Using REAPER, drag your audio into a new arrange page, then simply adjust the Playback Rate (default = 1) either with the slider or enter a new numerical value. Playback Rate is found just below the main arrange window to the right of the tempo and time signature (just called "Rate"). You can then render (bounce) this for use in whatever DAW you otherwise use.

NoSleep

Quote from: Better Midlands on October 30, 2018, 12:51:16 PM
I'm really not a technical head, but I would have thought that a tape would give a lesser quality when slowed down because BITD anything that ran slower (VHS/compact cassette etc) when you had the choice was always lower quality.

The difference from digital being that process of slowing it down is analogue so the only deterioration is the natural loss or gain of higher and lower frequencies dependent on the limitations of the tape deck - WYSIWYG. In digital you are more limited in speeding and slowing the pitch by your audio interface's capacity to do so (most are designed to run at set speeds 22.05k, 32k, 44.1k, 48k, 88.2k, 96k etc), so the audio has to be reconstructed to simulate any speed change between those set values (which can sound less convincing than using a tape machine as you move further out from the original speed).

Of course, some tape machines don't have a varispeed controller, so you still might be limited to half/double speed.

One disadvantage of digital over analogue is that you'll be up against the problem of aliasing. Essentially the problem stems from the idea that a digital waveform is not a perfect representation of the original analogue sound, but is built from samples taken at a constant rate (44.1kHz or whatever) which approximates the sound. This sample rate is normally fixed, regardless of how you manipulate it later.

Imagine if you want to reproduce the effect of playing a tape at 1.25x speed, but digitally.  You can't change the digital sample rate, so instead what you do is you step through the original digital wavetable at 1.25x speed and build a new wavetable from the points that you hit.  This can never be as precise as just playing the original analogue data at 1.25x speed because:

  • The original digital source is itself lossy and can't provide a precise value for the wave at a given point when the point you hit is between two samples.
  • The new wavetable is sampled at the same rate as the original, and so many of its values have to be approximated by some kind of best guess, given the two neighbouring samples.

In practice, signal degradation in analogue processing probably actually leads to a worse result than digital manipulation, but you can sometimes still hear the effects of aliasing in very high frequency sounds like cymbals which have been manipulated, particularly if the software uses naive algorithms to approximate the value of the wave between samples.. The problem is normally compounded if you manipulate an already manipulated piece of sound.

popcorn

Right, I think that question has been pretty thoroughly answered now. Cheers everyone. Consider yourselves all karma'd.

Here's my next question. I have a synth bassline I'm really fond of. It's really simple but I think it could be a really intriguing little syncopated groove if done right. Problem is, I can't do it right. Three separate, shit attempts:

https://www.dropbox.com/s/ega466oevl0df9c/illusion11.mp3?dl=0

https://www.dropbox.com/s/4qppscdl3yblkit/illusion%20shibuya4.mp3?dl=0

https://www.dropbox.com/s/4cppxjufwwm2iun/illusion%20synth%20start%20again.mp3?dl=0

These each sound bland and boring to me. I've spent hours fooling around with soft synths trying to get the bassline to sound warm and rich and complex, like the synth bass in this Ultraista track, but nothing I come up with hits the spot.

NoSleep

Try adding a buzzy top end sound over what you've done. I think that's what they've done in the example: bass line with a lead line following it to give definition.

jake thunder

There are two oscillators on that Ultraista bass synth or two different synth parts - one octave apart.

Try this VST - I'm sure there are some presets that sound similar:
http://www.best-free-vst.com/download.php?p=Minimogue-Luxus


popcorn

Aye, I have the Arturia Minimoog VST. I'm shit at using it though apparently. Nothing I make with it sounds remotely as good. I've tried using multiple octaves a few times...

the

#16
That Ultraista bass is quite a throaty synth bass that's had its filter opened up, but then had a low pass filter put on top of it. That's why you can hear it rattling but there's not much going on at the top end.

So I'd suggest finding a synth bass that sounds throaty with the filter open, then take the tops off with a LP filter. Tweak both filters a bit (and the resonance) and you'll probably find it.

You've nearly got it in the third example, it's just that's not throaty enough and is more like it's humming. Do it more talented.

popcorn

Quote from: the on October 30, 2018, 06:11:02 PM
That Ultraista bass is quite a throaty synth bass that's had its filter opened up, but then had a low pass filter put on top of it. That's why you can hear it rattling but there's not much going on at the top end.

So I'd suggest finding a synth bass that sounds throaty with the filter open, then take the tops off with a LP filter. Tweak both filters a bit (and the resonance) and you'll probably find it.

You've nearly got it in the third example, it's just that's not throaty enough and is more like it's humming.

Hang on a minute. Does this mean that applying a low-pass filter to a synth creates a different effect to applying the low-pass filter in the synth itself? I thought they were both just cutting high frequencies and there'd be no difference.

NoSleep

It can be the same and it can be different. The filters in a synth may be controllable via "keyboard follow" so will open up a little as you play up (or down) the scale instead of the notes disappearing as they reach the ceiling (or floor). An external filter will simply filter everything above or below what is set.

the

#19
Quote from: popcorn on October 30, 2018, 06:17:38 PMHang on a minute. Does this mean that applying a low-pass filter to a synth creates a different effect to applying the low-pass filter in the synth itself? I thought they were both just cutting high frequencies and there'd be no difference.

It can be markedly different aye, depending on how laryngeal the synth's filter is in the context of your patch. Have a fiddle with your little knob.

In this case, the later LP filter would just be acting like an EQ rolling off the top end, while the noise underneath is still shouting.

purlieu

This is all rather intriguing. I've never yet, in my life, managed to come up with a bass sound I'm remotely happy with. I consider it some sort of bizarre alchemy.

Quote from: purlieu on October 30, 2018, 07:28:02 PM
This is all rather intriguing. I've never yet, in my life, managed to come up with a bass sound I'm remotely happy with. I consider it some sort of bizarre alchemy.

I've always found basslines/bass sounds the hardest part of making music, regardless of music style - especially if I want it to be a dominant part of the tune.

popcorn

Here is my next "how did they done that?" question if anyone wants to have a crack.

It's about the main arpeggiated guitar riff in Map of the Problematique by Muse. (It's the sound that opens the song.)

I know the theory behind this sound - there's a sequencer that's pitching the guitar to different octaves. I can recreate a version of it using Ableton and Guitar Rig. I believe you could also do it with a pitch pedal with a midi input. What I'm curious about is this quote from the producer, Rich Costey:

QuoteThat song was originally all done on keyboards and I really wanted to hear it on guitar, but it was impossible - there was no way to play the keyboard part on the guitar. We spent about two days back-engineering what the keyboard part was on guitar. So it is actually a guitar that is going through three different modular synths that are opening up at different times. Two of the synths are routed into different pitch shifters - one is an octave up, the other is an octave down. Then we chose what octave we wanted to hear based on which synth we wanted to open up at which time. We had like an ARP 2600, some other things and a little spring reverb that was sort of playing the high octave. It was all done with hardware and the guitar was split into three: One went into the ARP 2600, Korg MS-20 and and EMS Synthi AKS.

What I can't make sense of is how synthesisers were used to process the guitar signal. I've barely used actual hardware synths so I don't know what he's getting at. Does anyone know?

NoSleep

#23
You can input audio into some synths and use the filters, envelopes and gates in the synth to affect the sound. You can even use the input audio to control the filters at the same time, so, for example, a filter will open up the louder/harder you play/sing. Quite useful. There's a load of that happening on my track on HA HA TAPE II where I stick a guitar through an old Roland SH09 which I'm sequencing at the same time, so you get a gated rhythmic pulse from me just playing long drawn-out chords (most noticeable from about the 5:20 mark as everything else drops out).

popcorn

Man, I've been messing around with soft synths for years and I had no idea you could do that. The noises in your track from 5:20 sounds like pure synth to me, I would never have guessed it was from a guitar input.

So in the case of the Muse track, they're using the synths as pitch-shifters? I don't really understand how you'd do that... filters make sense, pitch-shifting I don't get.

NoSleep

Quote from: popcorn on December 09, 2018, 12:18:48 PM
Man, I've been messing around with soft synths for years and I had no idea you could do that. The noises in your track from 5:20 sounds like pure synth to me, I would never have guessed it was from a guitar input.

There's some pure synth bleeps and squelches going on over the the less prominent rhythmic pulse of the guitar chord.

NoSleep

Quote from: popcorn on December 09, 2018, 12:18:48 PM
So in the case of the Muse track, they're using the synths as pitch-shifters? I don't really understand how you'd do that... filters make sense, pitch-shifting I don't get.

No, they're pitchshifting (some of) the results to copy the original keyboard arpeggio.

popcorn

Quote from: NoSleep on December 09, 2018, 12:22:09 PM
There's some pure synth bleeps and squelches going on over the the less prominent rhythmic pulse of the guitar chord.

In your track or in the Muse track? edit: OK I see, ignore this. thanks.

popcorn

Quote from: NoSleep on December 09, 2018, 12:24:38 PM
No, they're pitchshifting (some of) the results to copy the original keyboard arpeggio.

OK, thanks... but I'm still a bit confused about the process here.

So they program a sequence to modulate the guitar signal rhythmically, pitching the octave up and down at different points. That makes 100% sense. But are they repitching it using the synths? And if not, I'm not sure what they used the synths to do. To affect the filter sound? Add distortion? etc.

NoSleep

Quote from: popcorn on December 09, 2018, 12:26:49 PM
OK, thanks... but I'm still a bit confused about the process here.

So they program a sequence to modulate the guitar signal rhythmically, pitching the octave up and down at different points. That makes 100% sense. But are they repitching it using the synths? And if not, I'm not sure what they used the synths to do. To affect the filter sound? Add distortion? etc.

The synths will be used for filtering and gating. Then some of that is pitched up or down an octave using a pitchshifter or octivider. They say they split it up into three sections, one at the original pitch, one up an octave and one an octave below (or maybe two octaves up, dunno, not heard it). Putting the same guitar through each sequenced section would result in three separate tracks sounding like one complete take.