CAKEWALK SONAR 7 - NOW WITH CONCLUSIONS!

[BuleriaChk]

I may be repeating myself, but I installed the following updates for multicore and AT2 no longer crashes Live 6, although it still crashes SONAR. These are updates not covered in SP2 or any of the automatic updates. The first two actually solved the problem -live 6 ran for 24 hours without crashing.

(Sonar crashed in the first 2 minutes, as usual.. Nevertheless, I do think the issue is IK instability, not Sonar, since both Live and Sonar run without issue with any other plugin I've tried to date....

Hope this helps - for anyone with multicores, the updates may be worth it on general principles, but read first and backup your system before you try... YMMV....

Best Regards,

Chuck

http://forum.cakewalk.com/tm.asp?m=1174423
KB896256
KB936357
kb2005243

http://www.cakewalk.com/support/kb/kb2005243.asp
http://www.microsoft.com/downloads/details.aspx?FamilyID=c2ab5a48-8240-4934-bbd8-34fb8a0fce3b&DisplayLang=en

Hi, Craig, Thanks much for your response; I've tried everything! (And it crashes in Live 6 also). Tried it again (loading as a sfx, not a soft synth) and it crashed again.) I am on an Intel Dual Core E6700, Asus MB, etc. and everything else runs flawlessly. I am using ASIO, though come to think of it. I'll try WDM and see what happens..... But it does run smoothly within Revalver, which certainly makes the VST implementation suspect, IMO.... Chuck

 

[pbognar]

Sort of off topic, but I had to pull the trigger and upgrade from Music Creator 3 to Sonar Home Studio 6 XL for $79 (October special - too hard to pass up - Cakewalk, you got GOT me!!!).

 

Looks like an incredible value, and I assume that some of the Sonar 7 or 8 features will trickle down to Home Studio in the future, besides, SE and PE are probably overkill for me at this point.

[Anderton]

Still cranking on the AES videos...but getting back to Sonar.

Meanwhile, version 7.0.1 has been released with bug fixes and such, in particular, some Step Sequencer weirdnesses have been addressed. Here's the official word from Cakewalk:

Cakewalk Releases Free SONAR 7.0.1 Update

Boston, MA (October 16, 2007) - Cakewalk, the world's leading developer of powerful and easy to use products for music creation and recording, today announces availability of a free update for SONAR 7 Producer Edition and SONAR 7 Studio Edition. The 7.0.1 updates for both Windows XP and native Vista (32-bit and x64) versions of SONAR 7 Producer Edition and SONAR 7 Studio Edition are available as free downloads for registered SONAR 7 customers at http://www.cakewalk.com/Support/SONAR/default.asp.

The SONAR 7.0.1 Update addresses various performance and stability issues across the entire application, along with specific Step Sequencer functionality/usability enhancements, and numerous plug-ins and component optimizations. In addition, the update addresses several issues reported by customers after SONAR 7's initial release. A comprehensive list of what's included in the SONAR 7.0.1 update can be found at: http://www.cakewalk.com/support/kb/kb20071015.asp.

Many of the new and exciting features found in SONAR 7 are a direct result of active dialog with our users. Furthermore, Cakewalk's delivery of timely product updates is an extension of Cakewalk's commitment in providing the very best in product support for our customers.

Availability

The SONAR 7.0.1 Update is available as a free download from www.cakewalk.com for all registered SONAR 7 Producer Edition and Studio Edition customers.

[Anderton]

Boost 11 is a loudness maximizer (as in, "it goes up to 11" - the famous quote from the movie Spinal Tap, which was also the inspiration for Digidesign's "Eleven" guitar amp plug-in). Loudness maximizer is basically a fancy term for a limiter, and like all limiters, it can give a volume lift to a sound but if overused, produces the kind of nasty effect that seems to afflict most recordings these days: A squashed, flattened sound with no dynamics and audible dynamics.

 

Referring to the first image, Boost 11 has the same basic control set as similar plug-ins, including the freebie (and actually, extremely good) George Yohng's W1 limiter; but the Boost 11's metering is a big plus here.

 

The way the Boost 11 works is you boost the input (using the Boost control, natch) for the desired degree of squashing -- I mean, limiting. The input meter shows what's coming in, the output meter shows what's coming out, and the Reduction meter in the center shows the amount of reduction being applied.

 

Of even more use is the scrolling display of the inputs and output waveforms. Note the red peaks in the display on the left (although unfortunately, they're hard to see because of the image compression algorithm -- they're much more obvious in real life): These are the peaks that get limited in order to raise the overall average level, thus producing a louder sound. If you only see a few red peaks, then you're probably not doing too much violence to the signal. But if you see a lot of red, back off the boost.

 

The slider to the left of the waveform in display duplicates the Boost control, and the slider to the right of the waveform display duplicates the Output control. The Output control sets a "ceiling," which is set here to 0.1dB -- in other words, the output will be clamped so that nothing goes above 0.1dB.

 

The second image shows a piece of music without Boost11 (middle strip), with Boost11 (lower strip), and with too much Boost 11 (upper strip). All of them have the same peak level (0.1dB), but it's clear that the musical excerpt with Boost11 is louder signal than the one without Boost11. As for the one with too much Boost11, note the "flattening" of the signal between 25-30 seconds. As you'll hear in the next post, this produces an audibly unpleasant sound.

[Anderton]

Well, the proof is in the listening

Attached are three audio examples: A file without boost, one with a judicious amount of Boost 11 (about 3dB), and the third with what I consider to be way too much boost (about 6dB). I think the one with a little boost sounds pretty good; it doesn't affect the signal much or make it sound "squeezed," but definitely gives a hotter level. The one with too much isn't terrible, per se, but you can hear it has a hard time handling the big brass swells toward the end. However, the electric guitar peak on the first note gets reduced quite a lot, and you don't really hear that as unnatural.

[Anderton]

These days, you need some kind of limiter/level maximizer, or the squashing crowd isn't happy. Boost 11 is pretty efficient; while the typical use for this type of processor is on buses, I personally find it most useful on individual tracks to tame something like a resonant filter-induced peak from a synth, as well as to make narration really stand out with vocals. Boost 11 doesn't have any "miracle technology" that other maximizers don't have; like the rest of them, push it too hard, and it won't be able to cope (not unlike people, I guess). You'll end up with a gritty, distorted sound on the louder peaks.

But really, if you're pushing it THAT hard, you're abusing loudness maximization. In fact, regarding a wish list, I'd like it if the metering and controls were a bit more finely-tuned. For example, the Reduction meter goes up to -24. Whoa! That's a lot. I'd rather have a meter that went to -10dB or so, and any more reduction would trigger a message that says "Learn how to use this freakin' thing, okay?" Well, maybe not something that obvious. Still, I can't help but feel that some people will see the Reduction meter barely moving, and figure they aren't hitting the thing hard enough. Ditto the Output and Boost controls: I'd give up range for resolution. and overall, I'd rather see a bigger scrolling waveform display and smaller knobs.

Admittedly, this is getting pretty picky. Fact is, Boost 11 does what it's supposed to do, and does it well. I guess it's just that in a world where there's so much overcompression, I'd like to see Boost 11 sort of gently lead the user into not pushing audio too hard.

Oh, one other VERY cool thing: The Boost 11 is not specifically tied to Sonar 7, and you can load it into any VST host you'd like. I've used it with Wavelab and Sound Forge, for example, and it gives a very good account of itself. The Linear Phase signal processors play well with others, too; we'll get into those next.

[Anderton]

I mentioned earlier that you can't cut/paste VST automation data between tracks, but that copying MIDI controller data is now much easier. Both of these statements are true, but I should also add that non-VST automation data -- mix levels, panning, send levels, and the like -- can be cut, copied, and pasted easily among tracks.

[Montanna1053]

I need some help, maybe you or someone in this forum can help. I am new to the forum thing so please pardon my ignorance on posting and replying and format.

Here is an email I sent to Tech support at Cakewalk.com:

I followed your instructions and removed the VST adapter, restarted the computer, went to Sonar 7_OPTIONS/GLOBAL/VST PLUGINS and deleted everything in the directory and put in C:/Program Files/Cakewalk/Vstplugins.

You told me to click on the SCAN button, but it is still greyed out and not selectable. I did check the boxes to scan on startup, rescan failed plugins and scan exixting plugins. No change.

My Sample content folder is located in the Windows directory in a hidden file, along with the folders from Sonar 3, 5 and 6. I thought when I did the uninstall those folders might have been uninstalled also. Could that be the problem? Sonar 3 and 5 folder seems to be empty excepty for 1 or 2 files, but Sonar 6 folder has all the files in it still. I don't have the exact path yet, but I will send that to you when I get home and boot up my laptop.

I also remember deleting a file or entry in the initialization file by accident. I was hoping that uninstalling Sonar 7 and running a registry cleaner to clean out orphan files might correct that. I have tried to uninstall and reinstall Sonar 7 about 4 times now. I noticed that someone else had an entry in a forum somewhere sayting they had the same problem that I have.

No one has responded to this message yet as of 5:27 PM Arizona time.

I anxiously await your reply and possible solution to my problem.

I hope you can help.

Thank you

[Anderton]

As I don't have that problem, I can't find a solution. Obviously Sonar isn't finding the VST directory, but I don't know why. Have you asked this question over at the Cakewalk forums?

[Anderton]

Sonar is inching ever-closer to being a mastering program (probably for some people, it's already there) and part of what makes that possible are two new processors: The LP64 EQ and LP64 Multiband (compressor). We'll cover the EQ first.

 

First off, these are quite different compared to the Sonitus track EQs. They are hungrier for CPU; I inserted 16 of the LP64s in a track, and the CPU power spiked as high as 33%, with fairly wide fluctuations. With 16 Sonitus EQs, the CPU power never went above 1%.

 

They also have long look-ahead buffers, which allow for lowering gain just before a peak occurs; there's about 20ms of delay through the plug-in. Because of this, both processors are slow to respond to edits you make. With the LP64, if you alter a parameter with the graphical interface, the output will mute while the parameter is moving (the muting function seems to be new with 7.01; previously, it did noisy glitching). This makes the LP64 unsuitable for automation, so you'll likely not use it as a track EQ, unless it's to make a permanent change to a track. Really, the LP effects are designed to be bus effects, particularly if you want to spruce up your master output.

 

To compare the LP64 and Sonitus EQ, I copied a four-bar clip of music, opened an effects bin in each clip, then inserted the LP64 in the first clip and the Sonitus EQ in the second clip. Both clips were looped so one would play right after the other, and both EQs were set to the same numeric settings; check out the attached picture.

 

My first surprise was that the two EQs affected the sound very similarly in terms of frequencies and gain. However, their "personalities" are audibly different, which I think is apparent even in the attached MP3 file. The Sonitus has a certain quality in the upper midrange that adds a bit of an edge, whereas the LP64 is more neutral in how it affects the sound.

 

Interestingly, a thorough audition of the LP64 didn't make me want to get rid of the Sonitus EQ, which has a distinctive sound that can be used to great advantage on tracks. But when you want the EQ to be as neutral as possible, and leave your mix pretty much alone except for whatever changes you program in, the LP64 gets the nod. I wish it could be automated, but I can cope . . . I can break a file down into objects, and insert an LP64 in each one, with stepped changes.

 

How useful is the LP64 to you if you're upgrading? I'd rate the LP64 as a very useful tool to have in your bag of tricks, even if it gets plugged in only on the master bus. It has a natural, honest sound that lends itself very well to bus EQ.

[ogeva]

I think the step sequencer is still buggy, despite the fixes on 7.0.1. I was hoping it would be better with the patch, and in some ways it is.
But still - sometimes when I copy a clip and paste it elsewhere, unlink it from the original and update some of the beats, openning it again shows the original pattern and not the correct one. You hear the correct one, but since the original is displayed it's impossible to edit it.
Also, when copying step sequencer clips to a different project, they are copied as step sequencer clips, but openning them shows no data. The work just the same, but uneditable.
I wish they would have a facility to convert a step sequencer clip back to regular midi, it would solve all of these issues.
Oren

[Anderton]

I wish they would have a facility to convert a step sequencer clip back to regular midi, it would solve all of these issues. Oren

 

 

That's easy to do: Just right-click on the Step Sequencer clip and select "Bounce to Clip(s)," or select it and go Edit > Bounce to Clip(s). The Step Sequence will be a MIDI clip, and if you double-click on it, it will open to the Piano Roll view.

[ogeva]

Thanks! I don't know how I missed that one.
It's an easy fix to the step sequencer quirks. Just bounce to clip and make it a step sequencer and it's the correct step sequencer clip

[Anderton]

Now let’s look at the LP64 Multiband Compressor; check out the attached image. For those not familiar with multiband compressors, they’re extremely useful tools for dynamics control as they split the signal into multiple bands, then allow compressing the bands individually. Thus, what’s happening in, say, the bass range doesn’t affect the midrange or treble, and vice-versa. As a result, there’s less potential for pumping and breathing, and more precise control.

The LP64 Multiband offers five bands of compression. Each band has buttons to select the band, solo it (very useful, so you can hear exactly what a band does to the sound), and band enable. You can also adjust the center frequency and amplitude of each band via a graphic user interface, and there’s a line of knobs along the bottom for Attack, Release, Threshold, Ratio, and Gain for the selected band. However, these also can be controlled by an “All” button, so if you want, for example, the same attack on all bands, click All and adjust the Attack – it will affect every band. You can also enable “Program Dependent Release” for the Release parameter, which chooses a release time automatically, based on the program material entering the compressor. Finally, there are input and output meters, as well as input and output level controls.

I presume the filters that split the signal into the five bands are based on the ones used in the LP64 EQ, as they exhibit the same “gapping” problem when you edit them in real time. And again, the LP64 Multiband likes sucking on the CPU. In a project with four Rapture LEs inserted, without the LP64 Multiband the CPU meter would peak at 42%, spending most of its time in the 25-40% range. With the LP64 Multiband inserted, peaks hit at 55% and the usual consumption was around 35-50%. So like the LP64 EQ, you’d mostly want to use this as a bus processor; or if you did use it on a track, bounce it so you could free up CPU power. (I did crash Sonar with it once by wildly varying one of the band’s frequencies, but wasn’t able to duplicate this so it may have been some combination of other factors.)

Okay, so how easy is it to use? Basically, it’s as easy or difficult to tweak as any other multiband – there’s a lot of interaction among the various stages, and you need to be careful to keep things under control. My modus operandi for tweaking a multiband compressor is to first decide how to deal with the high and low ends, which are covered by shelving filters (the three midrange bands use bandpass filters). I like to add a decent amount of compression to the low bass range, but not touch the lower midrange; that’s why it’s good to be able to solo individual bands.

Let me also add that you should not try to use a multiband compressor to fix problems with the mix! Get the mix as close to perfect as possible, and use compression only to fix problems that can’t be addressed through mixing alone.

The attached audio example shows a loop with and without multiband compression (using the settings shown in the screen shot). I felt the high end needed a lift, but EQ alone didn’t do it; I wanted the high end to have some more punch as well. The low end had a nice tone, but was a bit too dynamic – I wanted a “flatter” sound. Lowering the bass just via EQ caused the sound to lose some strength, so I decided to compress it; that way, when I lowered its level a bit, there was still a nice, strong sustain.

As you can see from the screen shot, the bass peak was moved quite a bit to the right to “grab” the bass range up to about 200Hz. Similarly, the high end was moved down to about 6kHz. The high-mid range I wanted to compress ended up being centered at around 3.10kHz. The other two bands were not enabled.

You’ll note that the bass band level came down by a few dB, while the high-mid and high bands were brought up. In the audio example, the first four bars don’t use the LP 64 multiband, while the last four bars do. You’ll note the processed section’s high end has a lot more presence, and the bass is tamed and brought into a better perspective with the rest of the parts.

As with the LP64 EQ, the LP64 Multiband is not keyed specifically to Sonar and works with other hosts…which is nice, because it does have a very transparent, open, desirable sound that avoids the “claustrophobic” effect some compressors can impart. Put the two LP processors in your stereo bus, and with some judicious tweaking, you can really improve the sound without adding undesirable coloration. Actually, make that “without adding coloration,” period. I would go so far as to say that with these two processors, Cakewalk has brought its audio processors to a new level, one inhabited mostly by “designer plug-in” companies. Good show – even taking into account the glitching that can occur when making edits. I can live with that, given the exceptional sound quality.

[Anderton]

Whenever doing a Pro Review, I try to be working on a "real-world" project in parallel using the product being reviewed. Not only does this serve as a test bed to check out various aspects of the program, but it lets me have a more "global" view than just looking at the individual parts.

 

In this case, the real-world project is a sample library of chill-type loops -- sort of Kraftwerk meets Eno, with a slight veneer of New Age. I must say that Sonar has been a piece of cake (sorry, couldn't resist) to work with. Crashes have been few and far between, and come to think of it, have occurred only when I've been really pushing the program to test something out, not in the course of creating the loops.

 

While working on the project, I've been tweaking the colors. I'm trying for a highly readable color scheme that's still not too violent on the eyes, making a little tweak from time to time, and living with it for a while. Once I have it nailed down, I'll post it here.

 

At this point I'm still mostly recording and tracking, but here are some of the new features that I've really found particularly useful:

 

The new MIDI toolsets. This is effing great for editing parts, and I can edit much more efficiently now. I'm not using the magnifier that much in piano roll view, but use it quite a bit when doing in-track MIDI editing. This usually happens after I've recorded most or all of the tracks for the loop, and want to get a more global view of the MIDI data in all the tracks.

 

Select controllers within note duration. This is a life-saver. I'm recording these tracks as spontaneously as possible to keep a good "feel," which means my timing isn't always spot on. Being able to have the controller data from my live performance motions "travel" with the clip is great.

 

Other MIDI tweaks. These are the little things - note glue, note split, multiple controller lanes, etc. that once assimilated into your workflow, make life a lot easier.

 

Dim Solo mode. It's very cool to be able to focus on a track, but in context. And a lot easier than adjusting a zillion level controls...

 

As I get further away from working on this project in the MIDI domain and moving more into audio (saving out loops, editing them, acidizing, etc.) I'm sure I'll find some other favorite goodies.

[Anderton]

Sonar now has an External Insert plug-in, like Cubase. The purpose of this is to let you use external hardware with Sonar that looks and acts like a software plug-in (although of course, you can only instantiate one instance of a hardware plug-in if there's only one piece of hardware). Actually, Sonar could always do this, but it was relatively complicated: You had to send a track to a bus terminating in an unused audio interface output, patch that to the effect input, patch the effect output to an unused audio interface input, record that to a track, then shift the track in time as needed to compensate for the delays caused by going out to an external device, then coming back in again. And you had to mute the original track, too. The plug-in concept is a lot easier.

The attached image shows the GUI for the "External Insert" plug-in. You specify the external audio interface output in the send section (you can also choose to send the left, right, or mono signal), and where the effect is coming back in the Return section. In this case, I was using the Dock 1 ports from the E-Mu 1820m interface. You also have level controls for the sends and returns, as well as the option to invert the phase of the return signal.

The part in the middle is the Delay block, which compensates automatically for delays through the system and effect: Click on the upper delay button, and Sonar "pings" the effects loop, and sets a delay compensation time.

To test this out, I loaded a drum loop into one track, and the drum loop processed through an external effect (DigiTech RP250, set to bypass) into another track, then listened. Without delay compensation (the default setting), there was audible slapback echo. Clicking on the delay button gave 79 ms of delay, and the slapback echo effect disappeared. I then threw the second track out of phase, to see if the timing was precise enough that it would cancel. It was close, but not exact; I used the manual offset option to add 860 samples (about 20 ms) and they cancelled. That's pretty close, and in most cases, 20 ms wouldn't be a big deal.

The implementation on this is painless. The send and receive block out any options that can't be used, but you can send two tracks into the same effect; however, you can't use the same outputs, although you could bring the right output back into one track, and the left into another. But that's kind of a hassle...I think you're better off just sticking to one hardware effect per insert.

One more thing: The send and return gains, phase invert option, left mute, right mute, and mono are all automatable parameters. Overall, I'd say for those who want to use external processors with Sonar, this feature makes it very easy.

[garrigus]

advantage on tracks. But when you want the EQ to be as neutral as possible, and leave your mix pretty much alone except for whatever changes you program in, the LP64 gets the nod. I wish it could be automated, but I can cope . . . I can break a file down into objects, and insert an LP64 in each one, with stepped changes.

 

 

Craig,

 

You should be able to automate it with a track envelope...

 

Scott

 

--

Scott R. Garrigus - Author of Cakewalk, Sound Forge 6, 7/8 and SONAR 1, 2, 3, 4, 5 Power books.

** Sonar 7 Power - Coming this month! ***

http://www.garrigus.com/

 

Publisher of DigiFreq. Win a free Absynth 3 or Kontakt 2 DVD Tutorial and learn cool music technology tips and techniques by getting a FREE subscription to DigiFreq... over 20,000 readers can't be wrong! Go to:

http://www.digifreq.com/digifreq/

[Anderton]

Yes, it's physically possible to automate it, but you get the same gapping problems as if you adjust it manually, making it more or less useless for automation unless you can make your changes during passages where no audio is happening.

[andyw_nz]

I've been using Sonar 6 Pro for a while now, and I'm really happy with most of it. The one thing that lets it down for me is the stave notation view - I would really like to be able to create a decent lead sheet for songs. Features like 'repeats', multiple lyric lines and articulations etc. would be incredibly helpful. I've been using Finale for my lead sheets, but it isn't my favourite bit of software!

 

I've even considered changing to Cubase because it is a feature that is so important to me, and from what I can tell, Cubase does it a whole lot better!

 

Does Sonar 7 have any improvements in this area or will I need to continue battling with Finale to get my lead sheets?

[strav100]

I was wondering how the new features in Sonar 7 would allow better integration with the Creamware Scope Professional card? Ableton Live 7 looks like it will have support for external effect/instruments . how does this feature in Sonar 7 shape up?

[Anderton]

Go back a few posts and you'll see I covered the external effects thing. Any questions on what I wrote? As to external instruments, I always figured hey, give it a MIDI out, and record the audio...

 

As to SCOPE, what kind of integration are you looking for? It already does a good job as far as I'm concerned...I'm a big fan of the SCOPE system.

[Anderton]

A sidechain effect, at least in terms of software, is an effect with a standard audio input along with a second input that allows audio other than the primary audio input to control some aspect of the effect's sound.

One classic example is a noise gate with a "key" input. This allows an audio signal separate from the one feeding the input of the noise gate to open and close the gate. For example, the output of a kick drum could provide the sidechain signal for a bass line going through the noise gate. This would "lock" the bass to the kick, making it sound like the world's tightest rhythm section.

Another classic example is compression. For example, suppose you're recording a singer/songwriter and you put rhythm guitar through a compressor. By feeding the vocal into the compressor's sidechain input, compression will occur if the vocal, not the guitar, exceeds the compression threshold. Thus, the guitar will be compressed when the singer is singing, but otherwise returns to its uncompressed state if there are no vocals.

You can think of vocoders as having a sidechain input-the input for the modulator. Sonar doesn't include a vocoder per se (although you can use the Pentagon soft synth as a vocoder, as described in the Pro Review for Sonar 6), I just mention it for sake of example.

Yet another use for sidechaining is to use a processed version of the effect's input signal to control the effect. Again using compression with a vocal as an example, you can do de-essing by feeding a copy of the input vocal with a massive treble boost into the sidechain input. The trebly parts will kick in the compression, while the sections without a lot of treble will remain uncompressed.

[Anderton]

Sonar 7 includes three modified versions of previously-available effects that do chaining: Sonitus:Gate, Sonitus: Compressor, and Vintage Channel VC-64. Of these, the VC-64 definitely has the most complex implementation, so we'll deal with that last and start off with gating.

When you insert one of these effects into the effects bin, the sidechain input becomes available as a destination for a track or bus. Referring to the attached image, the Sonitus:gate is inserted in a track called Pad_HugeBrass+Strings, which plays a continuous, sustained pad. The track Kit_ClassicElectro (shown in the Inspector) has a drum part that goes to the main out, but also has a send to a bus called Bus > Gate (circled in red for clarity). The bus output's destination is the Sonitus:Gate side input (circled in orange for clarity).

So the signal flow is that both audio tracks feed the main output, but the drum track also feeds a bus, which provides the control signal for the Gate's sidechain input. As a result, the drum track "chops" the pad track, making it sound more rhythmic-as you'll hear if you listen to the audio example, which includes the drum sound and the "chopped" pad sound.

Note that you can also just route an audio track out to the sidechain, but in that case, it won't be going to the main out. However, this implies you can come up with audio tracks designed solely to control sidechain inputs that never appear in the main output.

[johndefiore]

Hi Craig,

Going back to the external insert plug-in, have you been able to get good results with the plug on a bus? Do you get consistent pings every time? Have you tried it with several other plugs in the same effects bin? These seem to be problematic areas for many people. I know Cakewalk has promised an improvement in stability for a future update, but in the meantime it's useful to know who is having problems with what. The following thread on the CW forum describes some of the issues:

External Insert Thread

Thanks,

John

[Anderton]

I have yet to try this with a bus, I've just done tracks so far. But I'll check it out as soon as I finish up the section on sidechaining.

 

As to the "inconsistent ping" issue, usually it hits a number and stays there. If I keep pinging, I can get it to change but if I ping a few more times, it settles down to a consistent number. It doesn't nail the delay perfectly (see my original post above, I needed to tweak for best results) but 20ms seems close enough for most parts. I don't know if using the E-Mu interface has anything to do with the overall stability I'm experiencing; maybe it's a factor.

 

So while I wouldn't say the external insert thing is flawless, I certainly haven't even come close to experiencing some of the issues others have. But I'll try the bus insert and see if I can break that