The Digital Performer Tips Sheet

Discussion of Digital Performer use, optimization, tips and techniques on MacOS.

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This forum is for most discussion related to the use and optimization of Digital Performer [MacOS] and plug-ins as well as tips and techniques. It is NOT for troubleshooting technical issues, complaints, feature requests, or "Comparative DAW 101."
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bayswater
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Sample Rate Conversion and Anti-Aliasing

Post by bayswater »

It is interesting to go back to this site now and then. The results can change with minor releases of the products tested. Logic was once one of the worst and now is one of the best. DP was not so good a year ago.
2018 Mini i7 32G 10.14.6, DP 11.3, Mixbus 9, Logic 10.5, Scarlett 18i8
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Shooshie
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MagicDave on the LA-2A Leveler:

Post by Shooshie »

Magic Dave on the LA-2A Leveler:

[From Wurliuchi, who gathered these all up for us. Thanks, Wurliuchi!]

Here's what Dave said (pasted in):
magicd wrote:The MW Leveler behaves exactly like a real LA-2A.

"Warm up time" is important to understand. "Cool off time" is also important to understand.

The MW Leveler models a photo emitter and sensor circuit. The way that circuit works is that input volume over time will saturate the photo sensor. The amount of saturation is based on a number of variables, including how much gain reduction you are applying, the transient nature if the input signal, average signal strength, and time. That's the way the LA-2a worked and that's how the MW Leveler works.

The LA-2A is always adjusting to input, so it is always in "warm up" mode when signal is going through. When signal stops going to input on the LA-2A, the photo sensor starts to cool down. This cool down can last as long as a minute or more. Again, how long depends on the characteristics of the signal before signal stropped coming in.

With DP 6.02, we added a feature to make this a little more managable. The added feature was completed just as DP6.02 was ready to ship, so we didn't get a chance to include documentation on the feature.

If you Control click or right click on the MW Leveler meter, you get a sub menu. The sub menu includes the option to save the T4 state. When you select the save option, you are saving the state as it exists at that moment. So, if you play back one bar of signal through the Leveler and save the state at that point, that would not be the same state as if the Leveler had been getting signal for the length of the song. Remember, the Leveler is constantly adjusting to input signal over time.

One thing you don't do with a real LA-2A is non real-time bounce to disk.

Once you have saved a T4 state, that state will be reset in the Leveler under specific conditions. If you bounce to disk, that causes the Leveler to reset at the start of the bounce. That way, if you bounce from a cold start, the Leveler will already be "warmed up" for the bounce.
If you close the Leveler window, that will reset the T4 state.
If you automate a bypass on the Leveler, the T4 state is reset when the plug-in is un bypassed. So for example, let's say you have a vocal track that comes in half way through the song. You can play back however much of the vocal track you want to get the Leveler warmed up. Save the T4 state at that point. Now automate a bypass for the Leveler that unbypasses just before the vocal comes in. The Leveler will be warmed up when the vocal track comes in.

It's important to understand that the MW Leveler is not an imitation of an LA-2A. It is an actual model of the hardware. It does work the same way as the hardware. If it didn't, it would not sound and behave like the original hardware.
In order to eliminate the warm up and cool down times, that would drastically change the way the Leveler reacted and
sounded. Remember, the way the Leveler is reacting in the first bar of music is not necessarily the way it is reacting after 3 minutes of saturation. Even if we instituted some kind of lookahead feature (which would add significant CPU overhead), where would you want to look ahead to? 30 seconds into the track? Two minutes?

The MW Leveler is a musical instrument. It has it's quirks, and they are identical to the original hardware. If you understand how the device works, it is a very powerful musical instrument. The LA-2A is not the famous device that it is for no reason.

The Save State function will give you consistency for bounces and track playback without having to "warm up" the cell every time.

Dave
magicd wrote:
RecordingArts wrote:After having lots of problems with levels jumping, I had inserted some standby automation for the plug in thinking that would solve the problem. (Can't remember where I read it, but I understood that keeping it in StandBy mode would keep it "warmed up"
No, stand-by mode does not keep the Leveler warmed up when the track is not playing.

If you bypass the Leveler (with no saved T4 state), the cell starts to cool off, even if there is signal present. Therefore, if you are comparing dry to processed signal, every time you bypass and then unbypass, the T4 cell is cooling off then warming back up again.

If you put the Leveler into standby mode, signal is still sent to the T4 cell, while the track is playing, although you are hearing the direct input signal at that point. Stand-by allows you toggle the Leveler on and off without it having to warm up every time you turn it back on.

If you have saved a T4 state, bypassing then unbypassing the Leveler resets the T4 to the saved state.

Regardless of whether you have a saved T4 state or not, going in and out of stand-by does not reset the T4 cell.

Dave
magicd wrote:
EMRR wrote:Dave,

Please tell us about the pre-checked 'automatic memory restoration' option.
This is what loads the saved T4 state when you unbypass the Leveler, or Bounce to Disk. If the option is unchecked, the only way a saved state gets reloaded is if you manually select that in the options window.

You also don't want to use a saved state from one model of the T4 with a different model of the T4. In other words, if you save the state with Fast Modern as the chosen model, you won't like the results if you change the model to Slow Vintage and then restore the saved state.

It's useful to know that if you save an MW Leveler preset that includes a saves T4 state, the saved state is part of the preset.

Remember, there is no generic "warmed up" mode. When signal is hitting the T4, the T4 is constantly adjusting it's response.

Let's say that you have a vocal track that starts with the singer screaming loud, but at the end of the song, they drop down to a quiet performance for the last few bars (or maybe the opposite - the singer starts soft and gets loud at the end). In such an example, it would make a big difference where you chose to save the state of the T4.

Dave
magicd wrote:
EMRR wrote:
Thanks. So, if no state has been manually saved, can 'automatic memory restoration' do anything at all?
If no state is saved, there is no memory to restore.
EMRR wrote: Sounds like one would need to dump any saved states included with saved presets, when using them on a different session. Is that correct?
I agree with that. The saved state is specific to the track content and the specific place where you stopped and saved.

Dave
magicd wrote:
RecordingArts wrote: We were working on getting levels for BGVs that only appear in the choruses of the song. When we were setting levels we were playing the chorus over and over, (which I guess saturated the MWL to a certain degree) but when I did a bounce to disk, it seemed I was getting different levels. (Maybe because the saturation was different since there was no signal for a minute or so up until that point)

-Vincent
When you bounce, the state is reset at the start of the bounce. Regardless of how much saturation you do before the bounce, you will get a reset when you do the bounce. That's the biggest reason to be able to save the T4 state.

Dave
magicd wrote:
RecordingArts wrote:It would be nice if the saved T4 state would be saved along with the sequence and could be saved as a normal DP Preset for the MWL..
Where would the automatically saved state come from? One minute into the song? End of the song? If you tried to take an average of the input signal from beginning to end, that would mean the state wouldn't represent any specific part of the track at all.

The best approach is to understand how the Leveler is working, and then make it work for you.

Dave
magicd wrote:
beautypill wrote:I discussed with an engineer in a studio full of outboard equipment.

We were both laughing about it.

Here's the problem with the premise of the plug-in.:

If you want to emulate a real studio processor, the plug-in should start warming up from the moment you turn your computer on.
Nope. Go back and reread my first post.

The "warming up" that you describe has nothing to do with powering up an LA-2A. There is no "warm up" involved at all. We are not talking old MiniMoogs here.

Over time, the T4 cell reacts to it's input. If the T4 cell sits there for three days powered up, but with no signal going through, it is not "warming up" in any way at all.

Once signal hits the T4, the T4 starts to react to that signal. As long as the T4 is receiving signal, it is constantly adjusting it's response. When the signal ceases, that T4 cell starts to "cool down".

At any given point in a track, the T4 will be responding differently, based on all previous input up to that point. The best way to understand this is to think of glancing up and looking at the sun for a quick moment. You look away and see spots until your eye recovers. If you spend longer looking at the sun, your eye takes longer to recover when you look away. That's exactly how the T4 works.

If you changed the characteristics so that there was no memory in the T4 cell, it wouldn't sound like an LA-2A.
Anything that doesn't work the same way as the original T4 can not claim to be a true model of the T4.

One suggestion I saw involved an analysis of the entire audio track that could be used as a reference. That is still making the assumption that you want the Leveler to sound as if it were starting from a cold start on the track. What is more likely is that you will want the leveler started when the cell is already in a reactive state. Therefore, you have to decide what that reactive state will be. Do you want the state the cell was in at the end of the track playback? Do you want the state the cell was in after a bar of playback from the beginning of the track?
This method would also require that the entire track be recalculated every time you changed a parameter in the Leveler or any plug-in in line before the Leveler.

Here is a metaphor:
A guitar has tuning pegs. If the tuning pegs are not set correctly, the guitar is out of tune. Therefore, does that mean that tuning pegs are a bad idea and should be replaced by a system that automatically tunes the strings? Well, that sounds like a nice idea until you examine the question of alternate tunings, the ability to grab a tuner and crank it while playing, etc. In other words, the technology as it exists offers specific capabilities. If you change the technology, you change what that technology can do.

Just because Gibson makes a robot guitar, that does not mean all my other old guitars are unusable now. :mrgreen:

Dynamics processors all have their flavors. The MW Leveler is a model of the LA-2A. If you don't want that flavor, there are other choices. If you do want that flavor, you get it with the MW Leveler.

Dave
magicd wrote:
EMRR wrote: If we accept, for argument's sake, that the model does work like the real thing, then compare people's experiences with the real hardware, we can only assume there's some other intermittent bug that's not flushed out yet. You really don't hear people complaining about hardware LA-2A behavior.


Someday I'll get the pair of hardware LA-2A's I've built fully wired into their case, and do some comparisons. My T4's are a recreation built by someone who's reverse engineered their own version, using correct materials. They may well have different response times than any of the four provided models.
Two notes to add here.

The MW Leveler is not just one model. We found that from year to year the T4 cells differed significantly. My understanding is they were hand made components. The different models in the MW Leveler are very different in their characters. Switching between the models while using the same parameter settings will give very different results. That's not something you can do with the hardware model.

One of the advantages of the LA-2A is that you can use it for drastic gain reduction with very good results. Try 20db of gain reduction with a fixed-attack compressor and see what happens. You won't like it. You can do 20db of reduction with the MW Leveler and get great results. But if the cell is not already in reactive mode when the signal comes in, the Leveler has to get back to a saturated state before it can do that amount of reduction without a volume bump at the start. I suspect that the people who notice a volume bump on the Leveler are using it for extreme gain reduction.

Note two:
It's not just the T4 cell that makes the sound of the LA-2A. The T4 is part of the gain reduction circuit, but it is not the only contributing factor to overall sonic response. For example, take a look at the UAD LA-2A plug-in. See the graphic of the set screw on the front panel? Nice graphic emulation of a parameter that is found on the real thing, but the graphic is just that - a graphic. It doesn't do anything. The MW Leveler, on the other hand, has the actual Response knob, which does do what that parameter does on the hardware unit.

We also used original vintage copper in the wiring of the MW Leveler, but I can't tell you too much more about that...

Dave
magicd wrote:
beautypill wrote:Dave, would you concede that people should be told that Bounce To Disk does not work when using the MW Leveler?

Shouldn't there be a little warning in the Bounce To Disk window that says "If you are using any instances of the MW Leveler, this bounce will not sound like what you have been listening to."?

I mean, isn't this a fair thing to alert the consumer to, since Bounce To Disk is touted as a viable way to print mixes?
I bounce all the time using the Leveler. I just finished a 50 song remaster for Sony that used the Leveler (with no saved states). So no, I don't think people should be told that bounce doesn't work with the Leveler.

Yes, the more documentation the better. As I mentioned earlier, the new save state features in the Leveler didn't make it into the 6.02 update notes because of the timing of the release.

Dave
magicd wrote:
bongo_x wrote: That's an interesting idea, and falls in the category of "I would never think of that because I would never do that". I use an LA2A for 2 things; barely bump it to get maybe 3db of comression/limiting, or smash it all the way to mix in parrallel. I would never use it for heavy steady compression on a vocal for instance, the volume will be all over the place. The middle range is no man's land for me on that unit.

That's making me thing that maybe some people just aren't used to how an LA2A works. If you're putting it on a vocal put it in limit mode and hit it lightly.

bb
Very useful comment. Thank you!

Dave
Whew! Thank you, Dave.
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Beat Detection Engine, "How To"

Post by magicd »

Using the Beat Detection Engine


This is a reprint of an old post:

Let's start at the beginning. The Beat Detection engine in DP is a core technology. From that core technology we can do some interesting things.

DP can look into an audio file and detect what it sees as beats in the audio.
It's possible to turn off beat detection, and there are circumstances under which DP will not analyze for beats unless you tell it to. Beats are visible as faint blue vertical lines when looking at soundbites in the Sequence Editor. If you open the audio in DPs waveform editor and select the Beats tab, the beats are marked with bold blue lines that also show velocity handles.

The first thing to understand about beats is that they are markings within the audio. When DP scans for beats, it has to make decisions. For example, if I hit a fat note on my Ric bass, the oscillations of the decaying note create beats as the string bounces against the frets. DP detects the beats at the oscillation points. Another common example is a partly open hi-hat. As the hi-hat swishes back and forth, DP can detect the "swishes" as individual beats. One thing that DP does very well is to find the beginning of the note as the beat point. The beat is not placed on the peak amplitude of a hit. DP finds the peak amplitude, then looks back to find the beginning of the rise of the hit. This helps notes from being chopped at the beginning.
Because there is interpretation going on here, you may disagree with the decisions that DP makes. There are also times when DP gets it right, but you want to make modifications to the found beats anyway. For example, lets say I've got a grace note or a run of notes, and I want to see that phrase as a single musical event. In cases like these, you will want to be able to edit the beats before you do any other operations. In general, it's a very good idea to be aware of what beats are in the audio before you start to actually use BDE tools.

Workflow:
Make sure DP has found beats in the audio by looking at the audio either in the Sequence Editor or Waveform Editor. If beats are not present, select the audio and choose Find Beats In Selection from the Beats drop down menu (to the far right of the Beats tab).

You can create beats, delete beats, move beats, and mute beats. Remember that a beat is only a marking within the soundfile. Moving, deleting, or changing a beat (or velocity handle) does nothing to the audio. It just informs DP as to the beat locations within the file.

Editing beats is done in the Waveform Editor. Clicking on the beat will audition the audio that is described by the beat. It's possible you may want to manually create individual beats, but it's more likely that you will be deleting, moving, or muting beats. In the work I've done, I mute beats as opposed to deleting them. I've not yet had occasion to create or move beats.

You have several tools available to work with. First, select a range of audio, then go to the Audio menu and choose either Adjust Beat Sensitivity or Adjust Beat Detection. The two commands do very similar things. Both commands give you a window with a threshold slider and Apply button. Adjust Beat Sensitivity works with relative amplitudes of beats (just like Recycle). Adjust Beat Detection works with musical priority probability (!). I generally use Adjust Beat Detection. It is a very smart command and does an excellent job of weeding out false or minor hits. The bottom line is that in order for BDE to do predictable things, the detected beats need to relevant to the musical task. I find simplification to usually be a good thing.

You can also use the Mute tool from the DP Toolbar to mute or unmute individual beats. This works well as clean-up after using Adjust beat Sensitivity or Detection.

Workflow:
Check the found beats to make sure they describe what you want to actually work with. Mute, move, delete or create beats to achieve this.

So now we have beats that make musical sense. What can we do with this information? To start, you can make selections based on beats. There is now a second Edit Grid button in the upper left hand corner of the Sequence Editor. When this is lit, selections will snap to beats. One cool trick here is that if you make a selection in one waveform, then drag so that other waveforms are also selected, all the selected waveforms will snap to the beats within the waveform where you originally clicked. You can not make beat based selections within the sequence Editor ruler. You make the selection within the waveforms directly.

Now that we've made a beat based selection, we can edit with those selections. You can use the Split command to make new soundbites based on beat selections. You can use the scissor tool to cut at beat markings. Clicking and dragging with the scissor tool selected cuts the audio at the beat marks for as far as you drag. Excellent for chopping up dialog or individual notes.

Cutting at beat boundries can also be done as a batch. Select the soundbite or soundbites, and choose Create Soundbites from Beats from the Audio menu. You get a slider and apply button. You also get a pop-up menu that shows all the selected soundbites. You can slice up soundbites based on their individual beats or you can use a single track as a guide and slice all selected audio using a single soundbites as the reference. If you want, you can then quantize those soundbites.

It is possible to time stretch audio in DP. You can use beats to quantize audio hits within soundbites. If you want to quantize beats within phase related tracks, you must first choose a single beat map and copy that to all the tracks you want to quantize. If you don't do that, there could be different beat maps within different tracks and therefore those tracks will not be exactly aligned with each other if you quantize.

Workflow:
Open the Sequence Editor window and display the tracks you want to quantize and keep phase aligned.
Select all.
Choose Copy Beats from the Audio menu. You'll get a small window that allows you to select the master track that all other selected tracks will copy beats from. There's a threshold slider and Apply button. You should have already picked your master beat track (typically snare or kick), and thinned out the found beats to specifically what you want to quantize. Apply the copy. Now you can select all those tracks and quantize beats within soundbites, and you will get the same quantizing for each track.

Quantizing is based on a sequence grid. If the tempo of your audio matches the sequence tempo already, quantizing is a easy. If tempo does not yet match, you must match the sequence tempo to the soundbite before you quantize. There are several ways to do this. DP will attempt to analyze the tempo of the audio based on the beats described. There are many ways to fool DP with this process. For example, what's the difference between 60bpm and 120bpm? What happens when the drummer plays a three over four feel on the hat, but a straight four with the kick? One example I looked at had steady tempo and then a two bar rest, during which the tempo of the overall track changed. Check the sequence tempo by turning on the metronome in DP. If the metronome agrees with the audio, tempo is good. Remember to turn on the Conductor track if there are tempo variations in the audio, otherwise DP will use a single straight tempo as defined by the tempo slider.

In general I've found that with loops and obvious transients in the audio, DP does a good job of detecting tempo. I've also found that you can definitely fool DPs tempo detection a number of ways. In the case of Tempo Changes From Hell (good name for an avant-jazz band), I may ask DP to calculate tempo automatically, but it's more likely I will use the Adjust Beats command from the Project menu>Modify Conductor Track to do this. Here again, audio beats are indispensable. You can snap barlines to audio beats with this command and it makes creating a tempo map quick and accurate. I do this all the time and have yet to find a musical example where I couldn't make an accurate tempo map.

Workflow:
Before you quantize beats or soundbites, make sure the tempo map is accurate and make sure the Sequence Tempo is Copied to Soundbites (Audio menu).

You can also create and apply grooves. This is a MIDI process that can also be applied to audio beats. The groove template is region based, so make sure of the region that is selected when you create and apply the groove. This is where the velocity handles come in. Groove Quantize works with timing, velocity and duration. Grooves can be created and applied from and to MIDI and audio.


Hope that helps!

Dave
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Re:

Post by snakejred »

:shock: WOW NICE


chamelion wrote:I'd like to offer a very cool tip that can help you to streamline level-setting during a mix:

Category: Obscure (at least to me) keyboard controller.
Description: Create the effect of a temporary group with any set of faders, regardless of whether or not the tracks selected belong to existing groups.
Works: During mixdown

OK, you're geting close to a mix, and everything's sounding good. You're fine-tuning your level on the the master fader prior to pulling in a compressor, but you're getting the odd clip light on the master fader. You need to locate the culprit/s, and adjust levels without altering their relative relationships within the mix.

The amazing 'W' trick!:

1. Hide all channels in the mixing console except the ones you want to temporarily group and tweak.

2. Hold down 'W".

3. Change the level of any of the faders, and they'll track as a group until you release the W.

I've tried to find this tip in the manual, but either I've been looking in the wrong places, or it's not there.


Cheers,

Geoff
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razamichoacana
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Re: The Digital Performer Tips Sheet

Post by razamichoacana »

Hello every one I'm not sure if this has ben discused before but im curious if it is possible. I know the W trick to move all fader down together a decibel or so. but what about when you have automation on a few tracks is there a way to lower a decibel on thoise tracks in one operation like the W trick as well with out having to select all the points in the sequence window one track at a time I really hope there is something available I would save so much time

Thanks Lots

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Re: The Digital Performer Tips Sheet

Post by FMiguelez »

.

Yes.
Try snapshots. You can make a time range selection and a to-be-included tracks. Press "t", move the faders as much as you want (they all will move together), and take the snapshot.
Read the corresponding chapter in the manual so you learn how to use all the parameters.

Also, make a shortcut for this, so you work faster.

I know you said you didn't want to select all the points in every track, but if you make a time range selection with the included tracks, you can also use the Change Continuos data command, and lower them all by percentage, amount, etc.

Both actions seem to be time consuming, but with practice you can do each in just a few seconds.
Try both and see which one works better for you. You can accomplish the same thing with either one.
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Re: The Digital Performer Tips Sheet

Post by toodamnhip »

1) Use the "W" command to group the faders
2) Put the faders in "Trim" mode
3) Hold down the faders as the materal plays and it will all be adjusted to the degree to command in Trim mode.
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Re: The Digital Performer Tips Sheet

Post by FMiguelez »

.

That would be my preferred method, but I gave up on it LONG ago.
I always found it unreliable and clumsy (especially with MIDI -and at least on my system with DP 4.61).

If they fixed this in newer versions, that would be very nice, and I'd look forward to try it again, since it's the easiest and least work-flow disturbing.

So I take it it works perfectly for you?
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Re: The Digital Performer Tips Sheet

Post by razamichoacana »

Thanks FMIguel and Toodamnhip I will try ur techniques Im reading more on this subject in the manual

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Re: The Digital Performer Tips Sheet

Post by toodamnhip »

FMiguelez wrote:.

That would be my preferred method, but I gave up on it LONG ago.
I always found it unreliable and clumsy (especially with MIDI -and at least on my system with DP 4.61).

If they fixed this in newer versions, that would be very nice, and I'd look forward to try it again, since it's the easiest and least work-flow disturbing.

So I take it it works perfectly for you?
Yes though MIDI is a bit more problematic but does work...
I love trim mixing and have used this style on real mix boards and in DP alike...I like the knowledge of how many db I am adding or subtracting
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Re: The Digital Performer Tips Sheet

Post by zed »

Here's something simple which may already be obvious to many, but which some others may have overlooked or not be aware of.

The point is this... when you install a new version of DP, or when you get a new monitor or reconfigure your existing monitors, you may find that when you open up your older DP projects the DP counter, mixing board, clipping and editor windows, CPU meter, etc., are no longer exactly where you want them. They are often misaligned, and in the case of DP6, opening up projects which were saved in DP5 result in a consolidated window that is bigger than it was before. In my case it was resulting in the scroll bar at the bottom of my window being buried below the edge of my screen. Pressing the green button at the top left of any window will resize it to deal with that issue.

But the real TIP which I want to share is that once you have resized and moved windows around to have things looking like they used to be, go to Windows > Windows Sets > Capture Window Set… and name a new window set. It will capture everything in the exact position you want it, and the next time you open up another project created previously, just select that window set and it will instantly resize and reset everything to your liking.

I know that this is just basic DP functionality... but since I neglected to take advantage of this for so long, I would bet that numerous others may be overlooking the ease with which you can deal with that above mentioned problem. And I just love that DP makes this so easy!

EDIT > I just wanted to add that window sets remember which VIs and plugins are opened or closed, so it is also a quick way to automatically close all your open plugins, for example.
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zed
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Re: The Digital Performer Tips Sheet

Post by zed »

Oh my my!! I am tickled silly right now. I just learned about Startup Clippings. I can't believe I have used DP for so many years and never realized how this works!

If you create startup clipping window in a project, it is saved with that particular project, and the special thing that it does is that it will OPEN any file that is in that window. Therefore, if you have a text file, for example, which logs where you are in a project or has notes about what you want to do next, you can drag that file into the startup clipping window and then whenever you open that project, this text file will open automatically. And EVEN BETTER, it opens up and presents itself to you before DP has finished loading so you have notes that you can read while you are waiting for the project to load.

To create your own, go here:
Project > Clippings > New Startup Clipping Window

When you have finished your project session, simply update and save the text file with any notes you will want to remember for the next time, and you will be reminded automatically.

And of course this will work with any other file type (from any application on your system) that you want to drag into this clipping window.

Nice, nice, nice!!
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Shooshie
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Re: The Digital Performer Tips Sheet

Post by Shooshie »

I'm going to have to explore that a bit more to see how it might interface with idea organizer apps like Yojimbo. I've always used an organizer of some sort to keep records on files I've been working on, especially for other people, where someone else might need to know all the tricks I'm using if they were to pick up where I left off. But I do it for my own projects, too. I prefer a spiral notebook sitting by my keyboard, but that's hard to keep with the file, and hard to know what notes I may have made specifically about that file. So, it's better to keep records in an idea organizer or text file. With Startup Clippings, it sounds like I could keep those files in such a way that they always remain connected with a particular file, which would make it a lot more consistent when needing to know how I programmed a particular patch for a sound, or what versions of which mixes are the "right" ones, and so forth.

Good idea, and thanks for bringing it to our attention. The manual kind of glosses over that; I don't think I'd have gotten the idea for startup clippings from scanning the manual's chapter on Clippings.
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daniel.sneed
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Re: The Digital Performer Tips Sheet

Post by daniel.sneed »

zed wrote:Oh my my!! [...] I just learned about Startup Clippings. I can't believe I have used DP for so many years and never realized how this works! [...]Nice, nice, nice!!
I could not have lived that long (!) without that :
It's my DP-Reason-Rewire time saver : Startup clippings open my Reason projects, as soon as my DP projects are up an running. Much awesome !
dAn Shakin' all over! :unicorn:
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crduval
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Re: The Digital Performer Tips Sheet

Post by crduval »

Hi all;

Here's a tip on moving plug-ins, its probably in the manual, but I didn't see it.

In the mixing board view, you can re-arrange and/or move plugins without changing settings by command clicking them - the cursor turns into a hand tool. This is great for experimenting with effect order or inserting additional plugs into an effects chain without having to save settings as presets. You can also move plugins from one track to another. I didn't check to see if you could option command click to move a copy (I will try this - just thought of it as I was typing - that would be handy for doubled tracks...)

Hope that helps someone out!
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