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STEINBERG MR816csx Interface and CC121 Controller - now with conclusions


Anderton

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Craig :
I would have thought that this unit would be designed in such a way as to make tracking as conventional as possible.
You set-up a mic.
You get yer' level.
Slap on a reverb/talent enhancer.
Record.
Playback....uh-oh....no effects ?
Uh....O.K. Let me instantiate another plug-in for playback. Now, let's see, was that a plate or a hall ? What was the decay time on that ? Uh? What about pre-delay.
Talent :
Hmmm....why does it sound different on playback ? Wow, that punch in was wierd.

Now, i may be exageerating a bit here, but, hmmm...you would have thought that it would have been thoroughly thought through ?

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Craig/All....

Thank you for posting my email.

Regarding the need to "switch" the FX (vis he control panel) from tracking to mixing.

Should you find that perfect setting that the talent and you as the engineer liked during tracking, but didn't want to print the FX, you can save the setting as a preset (eg. John's Vox Verb.) The preset will show in the preset list (as well as Media Bay) to recall when mixing or any time in the future.

The beauty of the MR series is found in the tight integration between hardware and software, not to mention the sonic qualities. The MR series allows you to monitor with the onboard FX, at near zero latency, and the FX control panels are integrated directly into the software - not via a secondary application or window. As well, you can control the FX settings directly from the unit.

If I am correct, the reason why the FX cannot run simultaneously during input (tracking) and ouput (mixing) is because of bandwidth limitations.

Should any one have additional questions, please feel free to PM me at any time. I will check threads as often as possible.

Thanks.

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I will need to check with the engineers about the technical specifics, but I do believe it's a combination of what the FW bandwidth can support and what was needed to accomplish what the engineers desired.

Also, Craig is correct that when using the MR-Series FX, say the Rev X Reverb for a mix-down, the mix-down takes place in real-time.

Thanks.

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If I am correct, the reason why the FX cannot run simultaneously during input (tracking) and ouput (mixing) is because of bandwidth limitations.


Should any one have additional questions, please feel free to PM me at any time. I will check threads as often as possible.


Thanks.

 

First of all, if anyone has any questions, unless they are not of general interest (in which case a PM would be more appropriate), please post them here so that we all can learn the answer. What gives Pro Reviews their credibility is the fact that we're not just limiting the review to my opinion, but opening it up to questions and comments from everyone -- users, potential buyers, manufacturers, etc. That way this becomes much more than "just" a review, and we all benefit from each other's knowledge.

 

Now, let me address the "bandwidth limitation" thing. I'm sure someone from Steinberg will correct me if I'm wrong.

 

The object of having a hardware/software combination is twofold. One, when you're limited by hardware, you can use the software and when you're limited by software, you have the hardware to fall back on. Two, there are instances where the two can be integrated to produce something neither can do by itself - like that wonderful Quick Connect button :)

 

As soon as you use hardware processing, you gain the following advantages:

 

* Virtually no load on the host computer.

* Extremely low latency compared to having to monitor through a software processor.

* Algorithms that can throw a ton of machine cycles at the processing because they're dedicated to that process, unlike the CPU in a computer.

 

And you gain the following disadvantages:

 

* Can't instantiate more than one instance. Think about it: If you had a rack reverb unit, you could use it while tracking or use it on mixdown, but not both. Hardware DSP within an interface works the same way.

* Can't do non-real-time bounces with most DSP. This is discussed in the Duende Pro Review, and on the second page, Shandor posted an ingenious workaround for software that doesn't provide for real-time processing, like Wavelab.

 

Another possible disadvantage is that presets aren't stored with a project the way they are with software plug-ins, because the hardware is external. However, the whole Steinberg/Yamaha Studio Connections initiative is designed to remedy this, and we'll find out if the MR816csx takes advantage of this protocol.

 

So to summarize, I don't think the "use it on playback or use it while recording" issue relates to FireWire bandwidth as much as it does to "silicon bandwidth" - a dedicated DSP chip can't miraculously become two chips.

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Oh and by the way, welcome aboard Brian! I hope you'll find, as most manufacturers have, that Pro Reviews are actually a lot of fun. It's also like having the world's best focus group at your disposal :)

 

True story: One company wanted to do a Pro Review quite some time after the product had been released. I advised them against it, saying they should probably wait until the next version comes out. They said "We really don't care about the review, we just want to see what people don't like and their wish lists so we can incorporate those comments in the next version."

 

A couple companies at NAMM told me specifically that several of the features in their new updates were a direct result of user comments on these forums. Now that's pretty darn cool!

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And you gain the following disadvantages:


* Can't instantiate more than one instance. Think about it: If you had a rack reverb unit, you could use it while tracking or use it on mixdown, but not both. Hardware DSP within an interface works the same way...


So to summarize, I don't think the "use it on playback or use it while recording" issue relates to FireWire bandwidth as much as it does to "silicon bandwidth" - a dedicated DSP chip can't miraculously become two chips.

 

 

 

Yes, but, it might have been possible, within the unit's software itself, pre-Cubase, to route the REVX as a SEND/RETURN AUX that could be accessible on monitor and playback ala the DSPFactory. I don't know. I'm just throwing this idea out there for you guys to mull over.

 

This may throw a wrench into the terrific integration with Cubase, maybe, I don't know, but, in this case it would have worked more like a console AUX SEND/RETURN rather than an either/or situation/limitation as it stands now.

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I noticed there's a new version of MR Tools, so while it's downloading, I figured I'd mention some other points of interest.

There's an input pad switch that operates very much like the phantom power switch: When you hold down the pad button and hit a Quick Connect button, that channel now has a pad inserted; and when you hold down the pad button, you can see which channels have pads because their Quick Connect buttons light up.

In terms of sample rates, you can do 44.1, 48, 88.2, and 96kHz - no 192kHz. Is that a deal-breaker for anyone? I thought not...

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Well it took me a while...but I figured out how to monitor the MR816csx's effects while recording, and apply them on playback. Let's look at using them as VST effects.

 

The key is to configure the MR Control panel so that on the MR816csx tab, under "Settings," you select "External FX." This is what allows the Channel Strip compressor/EQ/"Sweet Spot" morpher, as well as the reverb, to act as a plug-in within Cubase. When set as a VST plug-in, the effects apparently cannot be used while recording (which requires selecting Direct Monitoring under Devices > Device Setup; if selected for recording, these won't work as VST effects).

 

You also have to decide in advance how you want to allocate the hardware. The choices are 8 mono channel strip instances, 4 stereo channel strip instances, 6 mono channel strip instances and one instance of the REV-X (which you probably will want to use as a Send effect, as you have only that one instance), or 3 stereo channels with an instance of the REV-X.

 

Everything worked pretty much as it was supposed to, although I ran into situations where when I called up a Channel Strip or REV-X, I saw "not available" written in the effect GUI. This appears if you try to instantiate more instances that are allocated, but it sure seemed I didn't go over the limit. I tried disabling, removing channels just in case they had instances, and the like; no luck.

 

Finally, I quit Cubase, turned off the interface, turned it back on again, booted up Cubase, and all was well. This is the type of thing that could be pilot error, or it might have just been a glitch. Who knows? It's working now.

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Check out the attached image, which shows the Channel Strip plug-in.

The compressor is toward the upper left, with Attack, Release, Ratio, and Knee options (soft, medium, hard). There's also a compression graph and gain reduction meter (the reduction is about -3dB here). Below it are the side chain filter controls for the compressor, and the Drive control, which gives the signal some extra "oomph."

Toward the right is the 3-band parametric EQ - low and high shelf, and parametric mid.

The big knob on the bottom is the morph knob, which changes a variety of parameters as you turn it. Let me see if I can come up with a little movie here of how moving it changes the parameters...I'll post this, then come back in a few minutes after the video is rendered.

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If a picture is worth a thousand words, then with a frame rate of 30 frames per second, these videos are equal to 28,770 words. (The two videos are identical, except that one is rendered as WMV, and the other as MOV.)

 

As you watch the knob move, note that EQ, compression, and even sidechain settings vary in response. I wouldn't think of the knob as something that adds more or less of a particular effect; instead, it's more like a rotary switch that selects about a half dozen or so presets, but you can morph between these presets to get the "in between" sounds. It's not particularly predictable; I think the best way to use the Sweet Spot Morphing Effect is simply to play your source material, and turn the knob slowly until you find something that sounds right.

 

I'll do some audio examples tomorrow, I've spent quite a few hours on this tonight, and it's about time to take out the trash and then go to sleep!

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Hi Craig,

Thanks for doing this pro-review -- I bought the alesis io26 based on your previous review, and I like it a lot, but I'm thinking about upgrading to the mr816 since I'm a cubase user.

One thing that I'm very curious about is if you can use the MR816 with external preamps. I know it's supposed to have nice onboard pre's, but I would like the ability to bypass them on occasion since I'll have some nice "mojo" pres from seventh circle audio. I've read the user manual, and haven't been able to find a conclusive answer as of yet.

I know it's early in the review, but if you have the opportunity to test this functionality, it would be greatly appreciated! :thu:

Thanks!

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Hi Craig,


Thanks for doing this pro-review -- I bought the alesis io26 based on your previous review, and I like it a lot, but I'm thinking about upgrading to the mr816 since I'm a cubase user.


One thing that I'm very curious about is if you can use the MR816 with external preamps. I know it's supposed to have nice onboard pre's, but I would like the ability to bypass them on occasion since I'll have some nice "mojo" pres from seventh circle audio. I've read the user manual, and haven't been able to find a conclusive answer as of yet.


I know it's early in the review, but if you have the opportunity to test this functionality, it would be greatly appreciated!
:thu:

Thanks!



The eight input connects are Neutrik combo connectors (a good sign right there - they're quality components), so you can plug in an XLR or a TRS line input. So, if you wanted to use an external preamp, you would simply feed your mic into the pre, and plug its line out into the MR816csx's line in.

I don't have a block diagram or schematic, so I can't tell if the line ins bypass the input preamps entirely, or just pad them down. I would assume they bypass the preamps because the input impedance is 10k as opposed to the preamp's 3.5k, and besides, the build quality is such that I suspect they went for the better performance obtained by bypassing the pres (maybe someone from Steinberg could comment?). Maximum input level the line ins can handle is +14dbu.

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cool, thanks for the quick reply.

In posing this question on a different forum, someone responded:

"Those pres and A/D are derived from [Yamaha's] premium digital consoles. On the consoles, the pres auto switch out of the circuit at a predetermined gain setting. I think the pre switches out below -10, you can audibly hear it switch on the PM-1D, 5D, DM2000, etc... I would hope they would use a similar method on the 816 if the TRS jacks don't already."

Hopefully the above is true for the MR816 -- but I guess I'll be skeptical until I hear it from the horse's mouth.

Otherwise, I assume the two insert points could be used (as on the io26) and would completely bypass the preamps -- but it would be nice to have more than two inputs that could bypass the pre's.

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A few people have written me with comments along the lines of "well this sounds like a cool box, too bad I don't use Cubase." They didn't even bother posting the question, because they figured it was irrelevant to the thread. Wrong! The MR816csx acts like a standard ASIO interface, providing all the I/O. While you don't get the Quick Connect option for tracks, that's hardly surprising.

What is surprising is that you can use the internal hardware effects with many hosts! I tried this with Sonar, and it worked fine. Apparently, the effects look to Sonar like external effects. I inserted an external FX plug-in into a track, and sure enough, the effect showed up in the drop-down list of possible sends. As shown in the first attached image I selected FX1, the channel strip with Sweet Spot morphing; the MR816csx routing was three stereo channel strips and one REV-X reverb. Had I chosen FX4, the signal would have gone through the reverb.

Anyway, the return drop-down menu also showed the effect out, so I selected that as well. I "pinged" the external effect's delay parameter to do the proper path delay compensation, and was good to go.

Now check out the second attached image. This shows the setup with the REV-X installed as a second plug-in, following the channel strip. (Of course, as you're allowed only one instance of the REV-X, I highly recommend placing it in an Aux Send so you can use it with multiple channels; however, this was more of an "I wonder if this will work" test rather than actual music-making.)

To edit these, you need to call up the MR816csx's editor (in the lower left). Clicking on the E button (consistent with the Cubase GUI) brings up the effect's GUI, where you can tweak as much as you want. If the GUI isn't called up, you can still access two parameters per effect (in the case of reverb, that's reverb amount and algorithm).

Pretty cool! I though the effects might be "keyed" to Cubase in some way, but they're not. Next up, I'm going to see about using the effects while recording.

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Yes, the last post was only five days ago; I've spent most of this week getting the materials together for the next issue of the Harmony Central Confidential newsletter (you DO subscribe, don't you?) which ships next Wednesday, and also, recording audio examples of the "sweet spot" morphing for posting (hopefully this weekend).

 

Pro Reviews develop over time for a couple reasons. First, it allows others the chance to chime and ask questions. Second, it allows the reviewer to get more comfortable with the gear, use it under different circumstances, etc. Third, sometimes I have a very full travel schedule and when I do, it's impossible to keep up with a hardware Pro Review. I can do software only as long as I can load it on my laptop, but unfortunately the MR816 doesn't fit in my carry on!

 

The next two weeks will be dedicated almost exclusively to Pro Reviews as I'll be home until the Frankfurt Messe,.

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Just to add a little something:

Recently had an MR816 CSX for testing and with the newest drivers I could go down to 64 samples buffersize for a small 44.1 kHz Cubase 4.51 project on an 2.4 GHz Intel Mac. I measured 7.5 ms latency from input to output, including all converters and firewire-protocol - a really good result.

I then proceeded to measure the latency of the interface´s internal signal processing at 44.1 kHz and got 1.8 ms from input to output. A figure that´s right up there with the best of them.

Personally I would have immediately bought two Steinberg-MR-interfaces for my studio if they had
a) an internal powersupply and even more important to me:
b) gates in addition to compression and eq on the internal channel strip

As it stands I´ll have to look at the new Lexicon firewire interface or get the Metric Halo one (classy but expensive...).

Looking forward to the rest of the review!

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Just to add a little something:


Recently had an MR816 CSX for testing and with the newest drivers I could go down to 64 samples buffersize for a small 44.1 kHz Cubase 4.51 project on an 2.4 GHz Intel Mac. I measured 7.5 ms latency from input to output, including all converters and firewire-protocol - a really good result.


I then proceeded to measure the latency of the interface´s internal signal processing at 44.1 kHz and got 1.8 ms from input to output. A figure that´s right up there with the best of them.


Personally I would have immediately bought two Steinberg-MR-interfaces for my studio if they had

a) an internal powersupply and even more important to me:

b) gates in addition to compression and eq on the internal channel strip


As it stands I´ll have to look at the new Lexicon firewire interface or get the Metric Halo one (classy but expensive...).


Looking forward to the rest of the review!

 

 

Those latency figures are very impressive indeed. I'll check for new drivers and see how low I can go.

 

Question: Why is the internal power supply important to you? I understand a general dislike of wall warts, but this is a "line lump" that doesn't take up more than one outlet on a barrier strip.

 

As to the gate, it does seem like a curious omission given that many dynamics processors these days include gates or expanders and you might want to gate before the compression. However, I can't think of a host that doesn't include a gate as a plug-in. That's not the same as having it before the compressor, but it's a workaround.

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Those latency figures are very impressive indeed. I'll check for new drivers and see how low I can go.


Question: Why is the internal power supply important to you? I understand a general dislike of wall warts, but this is a "line lump" that doesn't take up more than one outlet on a barrier strip.


As to the gate, it does seem like a curious omission given that many dynamics processors these days include gates or expanders and you might want to gate before the compression. However, I can't think of a host that doesn't include a gate as a plug-in. That's not the same as having it before the compressor, but it's a workaround.



Well, regarding the wallwart:
I do a lot of recording outside of the studio - and want my rack as small and portable as possible, but with everything prewired and tightly secured. Open the lids, connect firewire and one (!) electric outlet - done (except for the audio connections of course).
A wallwart takes up a lot of extra space - I´d either have to make the rack 2 units higher to fit the supplies for the two MRs I´d like to use or make the rack unbelievably deep to mount them.
As for the gate: I want to use the channel strip for latency-free monitoring for the musicians. Sometimes I record (heavy) bands that require gating on kick and possibly toms to get feel and sound right even during tracking. Thats not possible with the built in gates of any native DAW - even the 7.5 ms throughput of the MRs is too much for a good drummer. Direct monitoring (or Protools HD...) is currently the ony option in this regard.

My 2 cents, YMMV :)

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even the 7.5 ms throughput of the MRs is too much for a good drummer.

 

 

Hi,

 

I think 7.5 ms throughput is too much for any time aware musicians. I know that I possibly could not hear the difference but when I change latency in my DAW I can feel the difference and that is not good.

 

Cheers,

 

Mats N

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Okay, Doctor, I see where you're coming from. Agreed that for your traveling needs, the line lump is less convenient. As to the latency...that's always an interesting topic. As a guitar player, 7.5ms doesn't bother me - I just put on headphones, and I have about the same delay as if I was seven feet away from my amp. Keyboards don't bother me either, because older digital keyboards had scanning delays that got me conditioned to having a slight delay anyway...so the VST instruments just take me back to playing a 1989 keyboard :)

 

Drums, though: As I noted in my review of Native Instruments' Maschine, you really have to have a low-latency system for real-time playing, or it'll bug the hell out of you to hit a pad and hear a delay, even if it's slight.

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