BLA Revolution EXP vs Lynx Aurora N

No, still in that painful “can’t make my mind up” zone.

I had decided on the BLA, then thought I’m going to need more channels and got hooked on the idea of the A32 Pro.

But I’ve just worked out the patch bay system will cost me £2000 alone!

I’m currently putting my two boys through university and so it’s soaking up my spare cash.

My wife pointed out, if I wait a couple of years my options will widen as I will then be able to afford a wider range of solutions.

So in reference to my current budget perhaps the BLA EXP would be a very good interim solution.

As modding other peoples converters is their schtick - I am easily convinced their own converters must be very good.

I definitely do trust your opinion on gear you’ve tried.

Interesting response from Black Lion in a YouTube comment regarding the reviewers that focus on specs rather than listening:



“Julian Krause's review is good in many places but we believe that much of it must be taken in context. Basically, we're saying that specs don't tell the whole story. Like many things, it's relative.  We personally think that specs matter but it's important to keep context because they're not always universal in what they explain.  This is actually a good opportunity for us to show people that we know what we're doing on both sides of the coin.  We understand specs well and can design with them but also know where they fall short.  We can design for subjectivism just as easily as we can for objectivism.  Especially because we're designing in a field that often times assumes that sound quality is directly proportional to price which we've seen from over a decade of mods is definitely not the case.  This is a lot of the reason why as product designers we personally don't see this as simply engineering design projects.  We see it as our "craft" and because of the fact that we have the ability to design objectively and subjectively to the point where we can actually explain our subjective methods in objective ways, we see this sort of design as going well into the realm of an art form.

We need to think of specs as swimming in "grey area."  Consider that often for some specs to be both highly accurate and universally implementable  we need to limit their scope.  Bandwidth is a good example.  We test between a range of 20Hz to 20KHz which is the AES standard range.  The standard is there to make sure that we're all testing the same way.  The problem is that one can argue that it's not the only way to test.  It's essentially the lowest common denominator. AES is telling us that if you're going to test your device to compare with someone else's measurements, this is how you should do it to ensure an even comparison. The reason bandwidth is an interesting example is because when we measure 20Hz to 20KHz we act like all acoustic energy and all electrical signals cease to exist beyond 20KHz.  It's just simpler to do that.  The fact is that while people argue that yes, the threshold of human hearing is at 20KHz, it's not an instant disappearance of a signal.  Human hearing rolls off somewhat naturally.  That basically means that while the cutoff is at 20KHz, there may still be audible artifacts above that point that just don't hear as loudly.  Not that we suddenly stop hearing them.  They just aren't perceived as loudly.  This range also varies from person to person.  By following the AES standard we have a universal set of guidelines to conduct our tests, but that doesn't mean that there isn't more than what's listed in the AES docs.  It's not a stone tablet of sorts.  Engineering and science is based on admitting that our methods are good but can always be improved.  As in audio, hearing something that doesn't appear in tests doesn't mean that what we're hearing doesn't actually exist.  It just means that current tests don't catch it. If we're not willing to consider the possibility of improvement, then how can we actually improve on something?  

Anyway here are our thoughts on Juilian's review. First his comments on latency are difficult to validate because the user should note that the latency of a device is very dependent on many variables within the operating system that aren't related to the specific interface or drivers.  Using some of the settings he mentions we got at or better than 2ms at 48KHz.  Much of the latency is just a result of Audio Class compatibility.  Consider a setting like 128 sample size at 48KHz.  It's not often understood why the sample size is so important but it's actually fairly simple.  If your data buffer on the OS side is set to 128 samples, that means that the OS will hold 128 samples before it posts to the audio data bus.  This basically helps keep the system organized and better at handling the stream. Consider 128 samples at a system Sample Rate of 48,000 samples per second is about 2.7ms.  That basically means that 2.7ms is the minimum latency you can achieve with a buffer size 128 samples.  This is a simplification because there are usually other OS specific functions and settings that add to this, even if they're not related to the audio.

The other thing we caught was his references to the Dynamic Range and EIN.  Usually Dynamic range is based on the EIN but it's a little unclear how he measures the EIN.  It appears like he's measuring the noise floor of the device using the DAW for the ADC side.  This unfortunately makes the spec unclear.  To measure EIN you should know the specifics of the gain staging on the signal path.  EIN, which stands for Equivalent Input Noise, is basically a method that we use to characterize a device's noise contribution when placed into a bigger system.  We get a generalized noise value and pretend like it's at the input and constant throughout the device.  The problem is that to get an accurate EIN for a device you have to know the specific gain values of the individual stages in the device.  For example if we had a preamp that takes a mic level signal and at the input boosts it by 12dB.  Then it goes to the main gain stage with a range from 6dB of minimum gain to 66dB of max gain.  In this case we'll do it at minimum gain.  Then the output cuts the signal down by 9dB.  If we see at the output that we have a noise floor of -100dBu, then our EIN is -109dBu. Measuring the signal at the output relative to the input to calculate the amount of gain in a device is a common way to simply calculate the EIN is a common way to do it but if you include the conversion stages and measure through the DAW then that value loses it's reference. Consider if you look at the DAW and keep considering the device's output as the amplitude in the DAW then there's not accurate conversion for the output.  The value found this way will always be skewed by the calibration value of the ADC in the system.  If your reference for the output is always when the DAW reads something like say -10dBFS, that doesn't take into account the level that an ADC takes in and equates to -10dBFS.  A device with a +20dBu = 0dBFS calibration will have an EIN that's 8dB higher when compared to a device with the same gain staging but a conversion calibration of say +12dBu = 0dBFS.  Both will show -10dBFS in the DAW but they'll have different EIN values since the second unit doesn't require as loud of a signal to reach the reading in the DAW.  The extra gain that the +20dBu calibrated unit needs to add to reach thee -10dBFS level in the DAW will not be taken into account when calculating the EIN.

The maximum dynamic range is also difficult to calculate without a reference.  Consider that in some systems the limiting factor is the calibration.  For example consider the above system with a +12dBu = 0dBFS calibration.  That means that the loudest you can hit in the analog realm is +12dBu or you'll clip the DAW.  That doesn't matter if the analog section of it's signal path can actually go up to something like 22dBu.  If the analog section can go that high, and has a noise floor of -100dBu, then it's dynamic range is 122dB.  Now if you include the converter, then it's dynamic range drops to 112dB.  Now consider the same analog section in a device with a +20dBu = 0dBFS system.  Then the same device has a dynamic range of 120dB when in reality nothing but the calibration was changed.  The fact is that having a higher headroom in the analog path has many benefits aside from Dynamic Range.  The other issue with manufacturers quoting the dynamic range is that they might post the range of their preamps, but not include the conversion in the gain path.  This would be more helpful to be able to compare external preamps against their own, but it's also somewhat deceiving.

This works sort of the same way for the DAC.  If all tests are done with the DAW showing say a value of -10dBFS, the output levels may actually vary depending on the system calibration.  A device who's calibration is +20dBu may have to cut some of the signal at the output reducing the audible noise floor while a converter that's set to +12dBu may have a higher noise floor since it doesn't have the benefit of cutting the signal in it's analog path.  That's why it's important to know the analog gain staging gain values when quoting EIN this way. It even works the other way.  If a converter chip that's usually made to feed a +12dBu calibration is amplified in the analog path to match a +20dBu calibration, then the noise floor goes up with it.  Measuring only the analog output relative to the DAW meter reading would give this device an 8dB reduction in EIN relative to the DAW's -10dBFS reading.  On the other hand, a +20dBu calibrated converter that wants to perform as a +12dBu calibrated converter,  cuts it's signal at the output.  Relative to the DAW readings of -10dBFS it gets an 8dB boost in it's EIN.

Julian's review is definitely helpful to get some perspective on the many devices out there but much of it sounds more in depth or thorough than it actually is.  While to his credit, he is doing more than just talking about how it looks, we find it interesting that in reviews like this, it's probably the best opportunity to compare the audio quality with examples of these devices, but so few reviewers actually do.  It may be because that seems like the easier route, as specs are easier to get and are often seen as irrefutable evidence.  Reviewers have an audience and the benefit of being independent of the manufacturer.  Too often when we've done samples they're automatically dismissed as having been "fixed" or tailored to our benefit.






We have seen this review and stand by everything we have said about the Revolution 2x2 and the experiences of thousands of people around the world match our own. This interface sounds absolutely incredible. The fact of the matter is,  this review does not and cannot translate to the real world. We also own Audio Precision machines- 2 of them- Just because you own a hammer doesn’t make you a carpenter. We deal with these machines day in and day out.  We have been very honest and open about the fact that our entire design process revolved around making an interface that sounded better than everything else available anywhere near this price range, and that the specs come after. Specs absolutely do not tell the entire story when it comes to how good something sounds. Should we sacrifice how a product sounds so it could look good on a machine or a piece of paper? Thats insane.

It is very easy (like VERY easy) to design a quiet preamp and convertor circuit with low distortion,  in fact, most convertor chips have application notes on how to optimally use their chip for these two things but the fact of the matter is these implementations would make one of most boring and uninspiring sounding interfaces you’ve ever heard.  We have made a career out of fixing that with our mods of other companies interfaces.

In the end one and only one thing is a real measurement of the quality  of an interface, and that is YOUR experience and how well you can make you make music on it.  Everything else is just a matter of record when it comes to specs. THERE IS NO SPEC that will show you on paper how much your ears and brain will enjoy using the interface- imaging, separation of instruments, stereo field, 3 dimensionality and phase relationships between the 2 channels. Any reviewer can rattle off specs but not focusing on the sound of the interface is as bad as a car reviewer who reviews a Ferrari and says it is a bad car since the windows are too small and it is too loud.  Cars and interfaces are very much the same, cars should be driven and interfaces should be listened to… everything else is just  numbers on paper.”

Right on! I think for certain setups the BLA looks great. Every time I look to expand my Aurora(n) I just figured it made more sense to add the analog I/O on the unit itself. Id have to take up space with ADAT to add an external box. But if I could get to the interface of the Lynx without taking up a slot, I would totally get something like the BLA. Enjoy your hybrid approach!