Re-pitching flex buffer live recording

No matter what I do, I can’t seem to make this happen. I use track 1 as the source, track 2 recording from track 1 and playing it back.

It all works, but as soon as I try to pitch down, the playback stops.

I already tried:

  • Microtiming the record trig one step right (it just hits me: Should that actually be the playback trig I did this to?)
  • No microtiming
  • No time stretch
  • sat all memory settings to the same as in the video

He even pitches it down without microtiming anything.

Any ideas what I can try next?

Should be left (-1/384).

Or right for playback trig (+1/384).

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I have to reopen this thread, even though I know it’s one of the most frequently asked questions.

How can I pitch down a live-resambling of ex. track 1…? When I set the microtiming of the play trig to +1/384 and pitch the track down by 2 or more, it still stops playing.

Pitching down, should mean the play head is slower than the recording, no? Not sure I understand, why it stops playing it almost immediately then… Doesn’t it seem like max marco does exactly this in the video I posted above here.

Also, any tips, tricks or fun on live-resembling is more than welcome. Just beginning to explore this part of the OT.

Also; Merlin guide doesnt mention the need to place the play trig on +1/384:

But when I have the REC and PLAY trig on same step, it stops playing in the first iteration of playing the pattern, and only begins again when the PLAY+REC trig is triggered again, after the pattern restarts. The second time, and every time after this, it all plays perfectly fine, with no stopping. Can anybody tell me if what merlin writes is even possible?

just a thought: what’s the amp envelope release set to on your track 2 that stops playing?

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On the track with the play trig that triggers the record buffer, it looks like this.

What I want, is in the first 2 minutes of max Marcos video. Pitching down doesn’t work for me, like it does for him. Is it because I’m resampling internal source, and he is using an external source? Or did some firmware change this functionality?

Just tried, Src3 recording (T1) with pitch / rate down should work with +1/384 microtiming and
default settings.

Merlin’s approach is not my reference.
Without +1/384 microtiming, a SRC3 recording is played 1 pattern cycle after, if its trig position corresponds to its rec trig.

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@sezare56 Thanks. That also works here - but without microtiming, it doesn’t work.

And if you look in the video I originally posted, Max Marco pitches it down without microtiming either of the play or rec trig in the first 1 minute 47 seconds. After this, he plays with microtiming. How come that doesn’t work for me?

My OT can’t even play back live recording buffer witht no microtiming, without audio dropping out in the first pattern iteration. (After the first iteration, audio plays fine without microtiming - but still pitching is not possible.)

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Because he’s using external recording, like you noticed. Different rules apply external vs. SRC3. MM explains the differences in the video

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Hmm, do you happen to have a timestamp for that explaination? I can’t find it, and to me it seems that using src3, he can still pitch down the playback of the live recording buffer without microtiming. I can’t.

My setup:

  • t1 playing a saved sample
  • t2 recording main to record buffer 2 with REC trig on step 1, no microtiming
  • t2 playing record buffer 2 on step 1, no microtiming
  • mixer (headphones) set to play cue, and t2 cued and muted from main, to avoid phasing.

Whenever I pitch down, no matter how little, I just get a click-sound in this case.
Time stretch is set to off, rate set to pitch.

Any ideas, or does you OT behave the same way?

I know this is a bit complicated to explain, but I’m not sure how to show all this in a video.

Not possible if you want rec trig and flex trig at the same position.
Where in the video?

Why Main and not T1 or CUE?

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@sezare56 Look from 4:27 to 5:27. Do I misunderstand what he does? Looks like he pitches it down, after removing all microtiming.

Does it make a difference if I choose main or t1, when there is nothing else playing than t1?

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Also work with MAIN, as you send T2 to CUE, but T1 as SRC3 seems safer to me.

Can’t check the video now, isn’t he using different start points?

No. In this part of the video he is pitching up. For an rec trig with an SRC3 recording source this works (after the buffer got already filled once), because it behaves like a delayed rec trig when recording only from ABCD sources. You can think of this fact as the rec and play head aren’t perfectly aligned anymore as soon as an SRC source gets involved.

The text overlays at 2:29 and 4:48 also implies this fact.

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Thanks. This would calm my quest for understanding a little. :slight_smile:

But can you explain what you wrote in another way? I really would like to get this. Are you saying there is a difference between recording from a/b/c/d and SRC, when I’m live recording and playing it back?

Yes, there is a difference.

When a SRC3 source is involved, a small delay/latency gets introduced.

I guess this delay is due to internal processing, because the SRC3 source needs to produce some output first, before it can be copied over to the rec buffer.

You can think of this delay/latency as like the rec head is behind the play head when you set rec and play trigs on the same step. And since the rec head is behind the play head you either hear silence (when the buffer was initially empty) or you hear the old content.

When an SRC3 source is involved, you need microtiming to compensate for this delay/latency (+1/386 on the play trig, for example).

When only A/B/C/D sources are involved there is no delay, so there is nothing you need to compensate for (you can set rec and play trig to the same position).

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Thanks to everyone for being patient. I get it now :nerd_face::joy:

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Isn’t a bit funny that it can delay compensate external audio source but not its internal one?
I never understood why that happens really.

There is no need to delay compensate external audio sources, because essentially it just takes the values from the A/D converters and copies them into the rec buffer (beside summing when multiple external sources are involved there is no processing involved).

With SRC3 sources it’s a completely different matter. Before a single value on the output of an internal track can appear much processing needs be done. Fetch the value from the raw sample source, apply envelopes, LFOs, apply FX etc.pp. and this chain of processing steps requires a fair bit more time than summing and copying.

Actually I’m not sure what happens when you record from both external and SRC3 sources into a single rec buffer. It would make sense that in this case the external audio gets also delayed a bit so it is in sync with the SRC3 source. But that’s just a guess.

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Good guess. You need to add microtiming.

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