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Post by RealNoob on May 27, 2022 19:32:25 GMT -6
It's not just Rob experiencing frustration with latency on this. According to Julian Krause's review, the latency on this is thing is surprisingly bad. Not out to bash anyone. Julian thinks it is odd given the quality components they've chosen. I had seen this review but as others have pointed out, he must have had a bad unit as mine and other's units are noise free. At this point, latency is a non-issue. I almost didn't try it because of his review. I am glad I did. The conversion is better than my Orion Studio with 129db DR, mastering quality outs, as they claim. I regret contributing to the negative press, although I have been careful to only post here about it, especially if due to my own machine. I don't care to bash anyone or anything either. All is well with my Revolution 2X2
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Post by kcatthedog on May 28, 2022 4:33:55 GMT -6
BLA had an excellent rep before its sale a while back. I have read about inconsistencies since the sale but frankly questioned the accuracy of those comments.
Any new adopter can possibly get some problems. Ironically, the mass produced manufactured product may have better QC, but, of course, the smaller shop , smaller run builder can have excellent QC too, dependent on the expertise of its builders.
For example, I have heard anecdotally, I think, no complaints about our friends at Audioscape, all its gear built by hand, must be the morning oj!
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Post by Deleted on May 28, 2022 5:04:14 GMT -6
I had seen this review but as others have pointed out, he must have had a bad unit as mine and other's units are noise free. At this point, latency is a non-issue. I almost didn't try it because of his review. I am glad I did. The conversion is better than my Orion Studio with 129db DR, mastering quality outs, as they claim. Yeah, I know spec's don't mean everything but they do give some insight.. What part in total makes this better than an MOTU Ultralite MK5 for example? I quote: "All internal gain-staging has been optimized for the very lowest signal-to-noise ratio." Well it hasn't really has it? Why would some of the specs be so poor if it's optimized for a low SNR? I was intrigued by this so I looked up the audio science review: www.audiosciencereview.com/forum/index.php?threads/black-lion-revolution-2x2-review-audio-interface.22141/They say lazy filter implementation, lack of headroom (IMD /I) and that THD sweep is?!! So, please let me know why these converters are "better" than an Orion exactly. Line Outputs Output Impedance = 95Ω (balanced) Max Output Level = 12dBu Frequency Response = 20Hz – 20KHz +/- 0.25dB THD+N = .002% (@ +10dBu) Dynamic Range = 106dB** Instrument Input Input Impedance = 1MΩ (unbalanced) Gain Range = 55dB Frequency Response = 20Hz – 20KHz +/- 0.25dB THD + N = .00114% (@ 0dBu) Dynamic Range = 103dB (A-Weighted) Crosstalk = < 100dB
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Post by raddistribution on May 28, 2022 7:18:47 GMT -6
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.
Regardless, we are thrilled with the success of the Revolution 2x2 and the support from thousands of users around the world means everything to us. We are a very small company and we are glad that we were able to bring something different to the market that we all at Black Lion love. As always we will do our best to listen to our customers while we work to bring out new offerings.
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Post by RealNoob on May 30, 2022 18:39:29 GMT -6
I had seen this review but as others have pointed out, he must have had a bad unit as mine and other's units are noise free. At this point, latency is a non-issue. I almost didn't try it because of his review. I am glad I did. The conversion is better than my Orion Studio with 129db DR, mastering quality outs, as they claim. Yeah, I know spec's don't mean everything but they do give some insight.. What part in total makes this better than an MOTU Ultralite MK5 for example? I quote: "All internal gain-staging has been optimized for the very lowest signal-to-noise ratio." Well it hasn't really has it? Why would some of the specs be so poor if it's optimized for a low SNR? I was intrigued by this so I looked up the audio science review: www.audiosciencereview.com/forum/index.php?threads/black-lion-revolution-2x2-review-audio-interface.22141/They say lazy filter implementation, lack of headroom (IMD /I) and that THD sweep is?!! So, please let me know why these converters are "better" than an Orion exactly. Line Outputs Output Impedance = 95Ω (balanced) Max Output Level = 12dBu Frequency Response = 20Hz – 20KHz +/- 0.25dB THD+N = .002% (@ +10dBu) Dynamic Range = 106dB** Instrument Input Input Impedance = 1MΩ (unbalanced) Gain Range = 55dB Frequency Response = 20Hz – 20KHz +/- 0.25dB THD + N = .00114% (@ 0dBu) Dynamic Range = 103dB (A-Weighted) Crosstalk = < 100dB To people around me, I am a tech-head. However, with you guys, I am still learning. I cannot comment on components and their effect equalling success in comparisons. That said, to me, the Revolution sounds more open and natural. My opinion is not from extensive flipping and ABing, it is from having used the Origin Studio since 2017 and then plugging in the Revolution on the same speakers. I use a Switch Witch presently with 3 sets of speakers and a sub. It has served really well for that with the Orion volume knob close in the rack. However, with the BLA on the video monitor shelf, its harder to reach. I am looking to grab a monitor controller like the Audient or Drawmer to not only be able to switch speakers but also sources.
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Post by Deleted on May 30, 2022 19:51:49 GMT -6
To people around me, I am a tech-head. However, with you guys, I am still learning. I cannot comment on components and their effect equalling success in comparisons. That said, to me, the Revolution sounds more open and natural. My opinion is not from extensive flipping and ABing, it is from having used the Origin Studio since 2017 and then plugging in the Revolution on the same speakers. We're pretty laid back around these parts and it's a great place to learn with so many experienced EE's, mixing engineers, tech's, designers etc. glad to have you on board. Simply put I was just curious, for example the Steinberg MRX816 didn't score all that well in terms of performance metrics but a lot people loved the sound of that interface. Appreciate your opinion..
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Post by urkirka on May 31, 2022 15:20:28 GMT -6
Ugh, the Babyface pro FS makes my HD 600 sound like Beyerdynamics. An unbalanced treble-forward sound with a shrill, metallic timbre in the top, just like the beyers. Good and punchy bass though. What is it with germans and that caustic treble anyway.. The latency/stability/build/feature set is pretty much perfect otherwise. Soundwise a real bummer for the amount of $ it cost.
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Post by RealNoob on Jun 1, 2022 22:10:50 GMT -6
To people around me, I am a tech-head. However, with you guys, I am still learning. I cannot comment on components and their effect equalling success in comparisons. That said, to me, the Revolution sounds more open and natural. My opinion is not from extensive flipping and ABing, it is from having used the Origin Studio since 2017 and then plugging in the Revolution on the same speakers. We're pretty laid back around these parts and it's a great place to learn with so many experienced EE's, mixing engineers, tech's, designers etc. glad to have you on board. Simply put I was just curious, for example the Steinberg MRX816 didn't score all that well in terms of performance metrics but a lot people loved the sound of that interface. Appreciate your opinion.. Interesting. I had an MR816 before the OStudio. While I had it, I thought it sounded great. Someone did a comparison against an Orpheus and folks had a hard time discerning which was which. A guy on the purple site even reported producing commercially available live albums, recorded with 3 MR816s. Clips always sounded great. The OS sounded a bit more open on top and focused on the bottom and the MR had an unusual focus in the high mids where the OS seemed more natural. Maybe just my unit, who knows. Seems like the manufacturers in the Pro-Sumer interface market continue to raise the bar.
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