Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 12:17:32 GMT -6
Sooooo….what happens when you run off two mixes….one clocked internally…the other clocked externally…..and then go listen to each on a consumer play back medium? Cheers Wiz You’re not hearing my whole chain because the trinnov isn’t printed. maybe you could print the trinnov with everything, da trinnov and ad clocked to a different master clocks and then play it back? Maybe one device doesn’t have a good pll arrangement in it when it’s slaved to another clock.
|
|
|
Post by Johnkenn on Jan 12, 2024 12:26:21 GMT -6
Anyway, the other side of the story is that a lot of people actually prefer a little jitter on the clock as it smoothes out harmonic content and softens harshness." I missed this when it was posted. This could be the answer to everything. Maybe the internal apollo clock is the technically better clock in this situation - I did mention that I thought it might be a little wider image wise - and the Burl, while technically worse - might be “softening harshness” which makes me prefer it?
|
|
|
Post by Johnkenn on Jan 12, 2024 12:28:48 GMT -6
Both things are true in this gig: You have to trust your ears. You also have to be aware that you can fool yourself (like when you're making genius EQ moves in bypass). I CAN say that every time I’ve done the bypasses EQ moves, the only way o figured it out was that I wasn’t hearing the moves I thought I was making…so we ultimately suss it out with our ears.
|
|
|
Post by svart on Jan 12, 2024 12:28:56 GMT -6
What irks me is I'm always up for a random funny post or gear post but myself, svart or Dan's posts which have some real experience or insights into actually how all of this this works is generally left high and dry. No post likes to be seen, I'm not expecting anyone to have Svart's sort of knowledge but how are people supposed to be AE's if you can't at least process a different technical point of view amongst or against "ears are king" (which are easily fooled) crap? Manufacturers create this stuff for you. Do you beleive it's all ignorance? Don't get me wrong, the differing opinions even on a scientific level, lapses in knowlege etc. all has its part to play. I've learned stuff from Chris (aka Svart) despite doing this for decades and I've met some tool sheds of engineers in my time. Before anyone gets annoyed I ain't saying you need to be a technical expert to be a mixing or mastering "engineer" however if you can't marry the technical with sound to a certain extent then god help you.. Edit: If you don't understand what I mean, retro preamp sounds awesome 20 likes. Foundation of how everything works and can be applied to actual production, zero likes. I mean, seriously?
I’ll be sure to like more of Svart’s posts…I wasn’t aware we were counting likes. No one is questioning whether what you guys are telling us is scientifically true…what I AM asking is why I’m not hearing what you guys are telling me I hear. The problem with dogma is that when you (just referring generally here) tell me what you’re hearing, I go, “that’s weird, I don’t hear that at all.” When I tell you what I hear, you say, “that’s impossible. You’re fundamentally incorrect.” This is why we get into yelling matches over this stuff. Just trust that I’m hearing what I’m telling you. I think if you were here, you’d hear the same thing. I appreciate Svart considering other possibilities and putting those out there. This is not about not believing in science, it’s about, “why am I hearing it this way when it’s supposed to not be this way.” Yes, I'm not saying anyone isn't hearing *something*, I'm just musing about what might even be the cause of hearing something different because there's soooo much that can affect it in a lot of different ways that might even end up with the same results. Without direct measurements, it's really all just talk though. I do find it interesting to roll around in my head some. I'd love to end up having a chance to determine what might be the cause, but I'd need to use a bunch of lab equipment and do modifications to interfaces to get direct measurements.. All stuff that's a whole lot easier said than done. Personally, I've just come to the conclusion that "good enough" is indeed good enough and that I'd rather have features that I need than worry about maximum performance, in my own system of course.
|
|
|
Post by notneeson on Jan 12, 2024 12:50:13 GMT -6
Both things are true in this gig: You have to trust your ears. You also have to be aware that you can fool yourself (like when you're making genius EQ moves in bypass). I CAN say that every time I’ve done the bypasses EQ moves, the only way o figured it out was that I wasn’t hearing the moves I thought I was making…so we ultimately suss it out with our ears. Yep, it’s fleeting. I’m just saying it’s worth challenging assumptions. Which is exactly what you’re doing.
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 13:05:54 GMT -6
I’ll be sure to like more of Svart’s posts…I wasn’t aware we were counting likes. No one is questioning whether what you guys are telling us is scientifically true…what I AM asking is why I’m not hearing what you guys are telling me I hear. The problem with dogma is that when you (just referring generally here) tell me what you’re hearing, I go, “that’s weird, I don’t hear that at all.” When I tell you what I hear, you say, “that’s impossible. You’re fundamentally incorrect.” This is why we get into yelling matches over this stuff. Just trust that I’m hearing what I’m telling you. I think if you were here, you’d hear the same thing. I appreciate Svart considering other possibilities and putting those out there. This is not about not believing in science, it’s about, “why am I hearing it this way when it’s supposed to not be this way.” Oh we're not really, it was a mere observation and then of course Svart's post about clocking in another thread was probably one of the most liked ..
Yeah, I mean audio is fundamentally quite complex and it can be rather confusing. Then you have to consider all possibilities to get the theory lined up with practice, I'm still going with the assumption that we just don't like things all that clean. I haven't done a study on it but I see it come up more often then not, although IMO it's not a bad situation to be in when things are so good technically nowaday's that we have to cover it in sugar. At least we have the option..
I certainly trust the opinions of those on here, it's just about finding out what's going on and if it's anything to be concerned about. I don't rip open and end to end test a new converter every time I get one, who's got time for that crap?
P.S Avid Carbons just arrived..
|
|
|
Post by seawell on Jan 12, 2024 13:16:39 GMT -6
I’ll be sure to like more of Svart’s posts…I wasn’t aware we were counting likes. No one is questioning whether what you guys are telling us is scientifically true…what I AM asking is why I’m not hearing what you guys are telling me I hear. The problem with dogma is that when you (just referring generally here) tell me what you’re hearing, I go, “that’s weird, I don’t hear that at all.” When I tell you what I hear, you say, “that’s impossible. You’re fundamentally incorrect.” This is why we get into yelling matches over this stuff. Just trust that I’m hearing what I’m telling you. I think if you were here, you’d hear the same thing. I appreciate Svart considering other possibilities and putting those out there. This is not about not believing in science, it’s about, “why am I hearing it this way when it’s supposed to not be this way.” Oh we're not really, it was a mere observation and then of course Svart's post about clocking in another thread was probably one of the most liked ..
Yeah, I mean audio is fundamentally quite complex and it can be rather confusing. Then you have to consider all possibilities to get the theory lined up with practice, I'm still going with the assumption that we just don't like things all that clean. I haven't done a study on it but I see it come up more often then not, although IMO it's not a bad situation to be in when things are so good technically nowaday's that we have to cover it in sugar. At least we have the option..
I certainly trust the opinions of those on here, it's just about finding out what's going on and if it's anything to be concerned about. I don't rip open and end to end test a new converter every time I get one, who's got time for that crap?
P.S Avid Carbons just arrived..
"I'm still going with the assumption that we just don't like things all that clean" - I agree with this 100%! If you get the chance at some point, I'd be very curious to hear a few things printed through the Carbon with internal clocking vs external.
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 13:31:07 GMT -6
Those converters were older and the asynchronous sample rate conversion technique to strip the received stream of jitter, reclock everything to it's internal clock, and feed the da chip an optimal sample rate that is a not a multiple of 44.1 or 48 didn't exist for audio converters then. Older multichannel converters and still most modern multichannel converters are quite poor unless you pay thousands of dollars and many of the good ones still like the Dangerous Convert 8 and Lynx Aurora N still let the end user defeat the internal clocking scheme. Crazy PLL arrangements were previously almost essential for high performance from many optical connections like Toslink and MADI like Lynx's "Synchroclock" and RME's "Steadyclock" but still asynchronous operation is necessary for high performance from Dante converters forced to use the Dante clock. The ESS Sabre chips allow this technique even in cheap consumer hifi products but of course audio quality and internal clock quality wildly varies.
The current Benchmark Crane Song, Weiss, and Lavry converters do not even have a clock input. They do not even want to give you the idea that you can mess with it. The Apogee Symphony rack mount case does but the converter chips in all Apogee products down to the lowly Groove are all asynchronous. Maybe that's just so other equipment can be spoofed into believing that it is the master clock and that the Symphony is slaving to it. This is all reflective of the new way of doing digital which is to have every device be it's own master clock and operate them (and every process) at the optimal sample rate. The only difference is feeding them single sample rates will have the anti-imaging filter applied at the end of the audible band. This is all a natural extension of oversampling noise shaping filters, which allowed Philips to get 16-bit performance out of physically 14-bit resistor ladder DACs in the 80s which eventually led to delta-sigma modulation based converters. Even going back to the 90s, Weiss hardware was multirate like the best modern plugins as an improvement over FIR smoothed algorithms to behave almost identically when fed different sample rates.
benchmarkmedia.com/blogs/application_notes/13127453-asynchronous-upsampling-to-110-khz
I mentioned in another post I've tested this with modern Apogee and Antelope as well, I just didn't have them for that particular test. So with this new way of doing digital, what does one with a bunch of outboard inserts do? Have multiple interfaces with none as the master, all internally clocked and hope for the best? The only example I didn't include in that test was all converters clocked internally because there was audible distortion and playback kept stopping so I couldn't print it that way. Their clocks are probably not as stupidly overbuilt as the Crane Song and heat can always be an issue in older multi-channel Apogee before they started using the lower powered ESS ICs in the multichannel and desktop versions. The Symphony I fan and the effin Quartet. Still everything Apogee since the ESS based Duet/Quarter/Symphony I/O module uses the ESS chips in asynchronous mode so you cannot really physically influence the clocking of the digital to analog conversion. You can probably add a few low level spurs of jitter but nothing really drastic.
All outboard inserts are pinged and latency compensated. The DAW and interface have no way of knowing that they are running at a different sample rates, the same as plugins with oversampling but external analog and digital hardware cannot report their latency and you're still hoping that they do not have a lot of phase shift and use linear phase anti-aliasing filters so that all of the signal comes in at the same time if paralleled unlike Softube, U-he, and Overloud plugins targeted at guitarists and keyboardists or cheap converters with minimum or mixed phase filters like Alesis, Focusrite, and RME Firefaces.
All modern DAWs not Pro Tools native have ping functions. That's the last real selling point for HDX because AAX DSP development is dead and the current hybrid engine admits the whole system is a glorified UAD Apollo with mostly worse plugins. The last competitive with native things that came out in AAX DSP were the clean Pro Audio DSP DSM, Sonnox Oxford Dynamic EQ, and the distorted Brainworx Bx true peak limiter. HDX does so automatically but assumes all of your converters have the same latency as the Avid branded converters and the third party PTHD addons just spoof that.
I haven't downloaded your test yet but that won't work with older digital hardware that needs to synch to a master clock. A lot of newer hardware has no synch and no need to because it will ignore it anyway like the Crane Song Solaris, Instellar, and Quantum units. Other units can sync the AD only. Again if you are operating AES into and output those converters from an interface, it is just pinged like a hardware insert.
Now with AVID rebranded interfaces like the DAD made MTRX studio, you cannot compensate the latency of the ADAT interface you hook up into it unless you enter a delay manually by hand because AVID wants you to get HDX to have working hardware inserts and wants you to buy an AVID branded converter and will insult you on their forums for asking about their gimped by design software: duc.avid.com/showthread.php?t=419112trail of breadcrumbs my ass if you want to use a high end stereo converter or something cool like a dangerous converter ad+ with the hammond transformers, a lavry ad with the soft saturation, or you have the stereo burl bombers instead of the mothership that will spoof the avid latency over digilink. Logic has had a hardware ping and automatic hardware insert latency compensation post ping (unlike for plugins lol) since I started using DAWs and so do Reaper and Cubase. Of course this can change with buffer so you should reping if you change the buffer.
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 13:43:08 GMT -6
And to clarify, I mean the Apogee ESS based DA converters, the A to D I am less sure of but the new Crane Song Instellar only has word clock outs, not word clock ins suggesting it is asynchronous and cannot be synched itself.
|
|
|
Post by seawell on Jan 12, 2024 14:00:38 GMT -6
And to clarify, I mean the Apogee ESS based DA converters, the A to D I am less sure of but the new Crane Song Instellar only has word clock outs, not word clock ins suggesting it is asynchronous and cannot be synched itself. Let’s say for example you want 32 channels of I/O in your studio for tracking and having some hardware inserts during the mix. If you have two 16 channel latest generation Lynx, would you have them both set to internal clock or set one as the master for the other?
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 14:21:09 GMT -6
And to clarify, I mean the Apogee ESS based DA converters, the A to D I am less sure of but the new Crane Song Instellar only has word clock outs, not word clock ins suggesting it is asynchronous and cannot be synched itself. Let’s say for example you want 32 channels of I/O in your studio for tracking and having some hardware inserts during the mix. If you have two 16 channel latest generation Lynx, would you have them both set to internal clock or set one as the master for the other? You set one as the master clock, the other as the slave clock, and turn on syncrolock on the Aurora that's the slave clock.
|
|
|
Post by seawell on Jan 12, 2024 14:42:12 GMT -6
Let’s say for example you want 32 channels of I/O in your studio for tracking and having some hardware inserts during the mix. If you have two 16 channel latest generation Lynx, would you have them both set to internal clock or set one as the master for the other? You set one as the master clock, the other as the slave clock, and turn on syncrolock on the Aurora that's the slave clock. Right, so sometimes you can’t stick to internal clocking on everything depending on your setup. That’s why I don’t get what the gripe was with what John was doing. If you have multiple converters in your set up someone is going to have to be the master or you’re going to have performance issues. Doesn’t matter if it’s newer or older converters. The only argument for internal clock always being superior that makes sense to me is that if you have one converter, of course I don’t think it would be worth syncing that one converter to an external master clock.
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 15:33:58 GMT -6
You set one as the master clock, the other as the slave clock, and turn on syncrolock on the Aurora that's the slave clock. Right, so sometimes you can’t stick to internal clocking on everything depending on your setup. That’s why I don’t get what the gripe was with what John was doing. If you have multiple converters in your set up someone is going to have to be the master or you’re going to have performance issues. Doesn’t matter if it’s newer or older converters. The only argument for internal clock always being superior that makes sense to me is that if you have one converter, of course I don’t think it would be worth syncing that one converter to an external master clock. I didn't post any gripe with it. I don't believe any of the converters he is using are asynchronous so he has to experiment with what makes the optimal master clock. I don't think you're understanding what I'm saying. These new digital to analog converters that use asynchronous sample rate conversion cannot clock to anything else. They cannot physically do it. There is no point for them to do it or any point to in general with a good internal clock that doesn't not use asynchronous sample rate conversion.
All these need is the samples in order, which almost anything without a damaged cable or fried i/o chip can send them. No matter what jitter is added by the interface in-between the samples and the converter. The only difference sending different sample rates to them makes is that your dac apply a filter down lower when it upsamples them.
The converter receives those samples in order and they reclock those to whatever frequency they want with their clock in order to feed the converter chip the optimal sample rate for the converter chip. So it takes the reclocked and resampled pcm and feeds that to the converter chip, which is always based on delta-sigma modulation, which reclocks that into the mhz at a lower bit depth, noise shapes away the insane amount of dither through delta-sigma modulation, and then converts the current to voltage, sends it to the line amplifiers, and out of the converter.
There is neither any need for digital to analog converter that uses asynchronous sample rate conversion to obey any external clock nor can it.
Now an analog to digital converter, bad clocking or extra jitter, will make it sample more erroneously and is the absolute last thing in the whole wide world you want and it cannot be removed by a digital to analog converter after the fact. If you need to sample 1234 and your Fuckusrite samples 1.2..3...4, the samples are always stored as 1234 in the computer despite being sampled over 1.2..3...4 in time. It will be lower fidelity to say the least.
|
|
|
Post by wiz on Jan 12, 2024 16:06:50 GMT -6
Sooooo….what happens when you run off two mixes….one clocked internally…the other clocked externally…..and then go listen to each on a consumer play back medium? Cheers Wiz You’re not hearing my whole chain because the trinnov isn’t printed. I get that……and when I re read my post it sounded a bit dickish……..not intended What I am getting at is…….if there is mid push…..less width…..etc…..using Apollo……and not…using another clock….which one translates? Thats the right one. cheers Wiz
|
|
|
Post by ab101 on Jan 12, 2024 16:08:24 GMT -6
I just want to express my appreciation for all the experts and others seeking expert advice that have posted on this thread as well as the dithering thread.
I try to like a lot of posts here. I like everyone here. This is a special place.
A wise person learns from every person. (Pirkei Avot Ch. 4:1 - an ancient proverb!)
I am grateful!
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 16:10:31 GMT -6
You’re not hearing my whole chain because the trinnov isn’t printed. I get that……and when I re read my post it sounded a bit dickish……..not intended What I am getting at is…….if there is mid push…..using Apollo……and not…using another clock….which one translates? Thats the right one. cheers Wiz what? For a da converter? No way because if it’s mid pushed, you will scoop mids to make it normal and then when some guy plays back your mix on a normal converter, it will sound mid scooped. I do think the Apollos sound mid pushed though and feel that many of the UAD color effects are voiced around that in addition to whatever they are attempting to model.
|
|
|
Post by wiz on Jan 12, 2024 16:14:14 GMT -6
I get that……and when I re read my post it sounded a bit dickish……..not intended What I am getting at is…….if there is mid push…..using Apollo……and not…using another clock….which one translates? Thats the right one. cheers Wiz what? For a da converter? No way because if it’s mid pushed, you will scoop mids to make it normal and then when some guy plays back your mix on a normal converter, it will sound mid scooped. I do think the Apollos sound mid pushed though and feel that many of the UAD color effects are voiced around that in addition to whatever they are attempting to model. Hey Dan thats exactly my point…perhaps I didn’t phrase it well…about adjusting to what you hear. I also think you are right about Apollo sound and it’s plugins and also Luna. using the entire chain as I am…I find they all play really well together sonically…far more than when say I was using Logic,Apollo,UAD plugins. Ths system works sonically cohesively cheers Wiz
|
|
|
Post by seawell on Jan 12, 2024 16:22:03 GMT -6
Right, so sometimes you can’t stick to internal clocking on everything depending on your setup. That’s why I don’t get what the gripe was with what John was doing. If you have multiple converters in your set up someone is going to have to be the master or you’re going to have performance issues. Doesn’t matter if it’s newer or older converters. The only argument for internal clock always being superior that makes sense to me is that if you have one converter, of course I don’t think it would be worth syncing that one converter to an external master clock. I didn't post any gripe with it. I don't believe any of the converters he is using are asynchronous so he has to experiment with what makes the optimal master clock. I don't think you're understanding what I'm saying. These new digital to analog converters that use asynchronous sample rate conversion cannot clock to anything else. They cannot physically do it. There is no point for them to do it or any point to in general with a good internal clock that doesn't not use asynchronous sample rate conversion.
All these need is the samples in order, which almost anything without a damaged cable or fried i/o chip can send them. No matter what jitter is added by the interface in-between the samples and the converter. The only difference sending different sample rates to them makes is that your dac apply a filter down lower when it upsamples them.
The converter receives those samples in order and they reclock those to whatever frequency they want with their clock in order to feed the converter chip the optimal sample rate for the converter chip. So it takes the reclocked and resampled pcm and feeds that to the converter chip, which is always based on delta-sigma modulation, which reclocks that into the mhz at a lower bit depth, noise shapes away the insane amount of dither through delta-sigma modulation, and then converts the current to voltage, sends it to the line amplifiers, and out of the converter.
There is neither any need for digital to analog converter that uses asynchronous sample rate conversion to obey any external clock nor can it.
Now an analog to digital converter, bad clocking or extra jitter, will make it sample more erroneously and is the absolute last thing in the whole wide world you want and it cannot be removed by a digital to analog converter after the fact. If you need to sample 1234 and your Fuckusrite samples 1.2..3...4, the samples are always stored as 1234 in the computer despite being sampled over 1.2..3...4 in time. It will be lower fidelity to say the least.
Right, it's not that I don't understand what you're saying it's just that I didn't see anyone mention here that they have an asynchronous converter so I missed how we got off on that. If John had one, of course that would need to be the master. Anyway, we've gotten so off track he had to start a new thread to get an answer so I'll chill 🤣
|
|
|
Post by Johnkenn on Jan 12, 2024 16:29:33 GMT -6
You’re not hearing my whole chain because the trinnov isn’t printed. I get that……and when I re read my post it sounded a bit dickish……..not intended What I am getting at is…….if there is mid push…..less width…..etc…..using Apollo……and not…using another clock….which one translates? Thats the right one. cheers Wiz Not sure width is really something that would make a difference in how I mix, though...it just sounds better. If we're talking about the Apollo clock seemingly being more mid pushed than the Burl, then yeah one would make you mix a little differently...but that's what the Trinnov is supposed to do - make things flat. So if all things are equal frequency wise, I guess the key would be to pick the one that gives the most pleasant representation of the frequency ranges...which is the Burl. If we're talking about the difference in using either the Burl or the Revolution (to bring in the topic from the other thread), I'm not totally sure which one I prefer. But I do know that the max output of the Revolution 6x6 just isn't hot enough for my liking. Add into that, you have to reduce the output of the Trinnov so it doesn't clip with any adjustments it's making.
|
|
|
Post by chessparov on Jan 12, 2024 18:29:37 GMT -6
Before anyone gets annoyed I ain't saying you need to be a technical expert to be a mixing or mastering "engineer" however if you can't marry the technical with sound to a certain extent then god help you.. When I woke Rick up from his nap, to ask about all this... He Clocked me all right. Really good. Now I'm relying on Dan to guide me... internal Scarlett clock? Or UA Volt? Can't eat or sleep. Definitely no Rueben sandwiches. Chris
|
|
|
Post by seawell on Jan 12, 2024 19:08:54 GMT -6
I'm not talking about the design holistically, neither am I talking about subjectivity in an overall sense. ADC has one job.. Some people prefer the old Lexicon units and they are just signal degraders really, I get it and IME or IMO technically correct can sound pretty boring sometimes. However we're talking about clocking actually increasing the performance of an easily measurable concept which is signal degredation. I have never seen anything that would point to an increase in performance via external clocking in a modern PLL system, you want to mash things up a bit by using an external clock then fine. Although again, I'd rather have the choice and I'm not the only one saying this. I'll quote Svart again and he's not selling converters anymore so there's no "narrative" to be found. There's either two things going on here, the converter has been cheaped out / not very well designed or people might like a bit of jitter in their coffee? ------------------------- "Coming from someone who designed their own converters and designs clocking solutions for RF that are 100x better jitter and phase noise than audio requires.. External clocks for audio are almost never worth the cost. The act of using interconnects, cables, buffering, phase locking, re-clocking through a DLL/PLL to reach usable sampling frequencies, etc., All conspire to reduce clock performance to levels below all but the worst internal clock. Most modern designs use multi-MHz DPLL-based oscillators then divide the frequency down 256-512 times to reach converter oversampling frequencies, or further down to direct word sampling frequencies. This divides deterministic oscillator jitter by the same amounts, leading to internal system clock jitter to be extremely low, generally almost as low as to be considered negligible in general sampling work. Power supply and system noise on the I2S data bus are a greater issue since careful PCB layout is needed for optimal performance, but rarely understood by those who've only done audio work. My converter clock was in the sub-picosecond jitter range, about 600 femptoseconds. To give a bit of comparison, ADAT standard required clocks with less than 1 nanosecond of jitter at 48KHz.. quite a large difference! Anyway, the other side of the story is that a lot of people actually prefer a little jitter on the clock as it smoothes out harmonic content and softens harshness." The Lexicons were better than most modern reverbs because since they couldn't sound realistic at all and most modern reverbs don't really either, they had to have their algorithms designed holistically to sound like a good special effect with a wide variety of material. The frequency modulation and chorusing makes voices and guitars sound angelic. The random hall is huge and surreal. Unlike most modern flexible "clean" or "utilitarian" itb reverbs you can make sound AWFUL with most settings deviating from the carefully selected presets. The Alesis Midiverb II is iconic and didn't even have settings, it had 99 built in patches Good luck making something like Valhalla Shimmer, a Strymon Big Sky, or Liquidsonics Tai Chi sound as iconic, utilitarian, and adhering to the signal like that or the iconic Quadraverb Taj Mahal setting. Keith Barr pretty much wrote the best primitive Schroeder reverb possible, the distortion made it sound smaller and adher to noisy signals better, and with it, shoegaze was invented. My not even top of the line lexicons(PCM70, 80, 91) eat any plug in for lunch, it's wild.
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 20:38:09 GMT -6
The Lexicons were better than most modern reverbs because since they couldn't sound realistic at all and most modern reverbs don't really either, they had to have their algorithms designed holistically to sound like a good special effect with a wide variety of material. The frequency modulation and chorusing makes voices and guitars sound angelic. The random hall is huge and surreal. Unlike most modern flexible "clean" or "utilitarian" itb reverbs you can make sound AWFUL with most settings deviating from the carefully selected presets. The Alesis Midiverb II is iconic and didn't even have settings, it had 99 built in patches Good luck making something like Valhalla Shimmer, a Strymon Big Sky, or Liquidsonics Tai Chi sound as iconic, utilitarian, and adhering to the signal like that or the iconic Quadraverb Taj Mahal setting. Keith Barr pretty much wrote the best primitive Schroeder reverb possible, the distortion made it sound smaller and adher to noisy signals better, and with it, shoegaze was invented. My not even top of the line lexicons(PCM70, 80, 91) eat any plug in for lunch, it's wild. i mostly end up using Megaverb now that the Alesis is gone. Sometimes SP2016 that’s ported from the Princeton 2016 from like 10 years ago. I can’t get behind any of the EMT ports 100%. Relab I can only stand sonsig. I’m going to try the Tsar more.
|
|
|
Post by seawell on Jan 12, 2024 21:33:17 GMT -6
My not even top of the line lexicons(PCM70, 80, 91) eat any plug in for lunch, it's wild. i mostly end up using Megaverb now that the Alesis is gone. Sometimes SP2016 that’s ported from the Princeton 2016 from like 10 years ago. I can’t get behind any of the EMT ports 100%. Relab I can only stand sonsig. I’m going to try the Tsar more. I tried to do a shootout between Acustica Silver when their PCM 70 library came out against my hardware but it was so different I just abandoned it. Didn't sound anything like the hardware to the point where it was laughable. Have you tried any of the liquid sonics stuff? I actually like Reverberate 3 with some libraries I've picked up over the years. Really digging in and tweaking the options there got me closer to the hardware than anything else I tried which surprised me since it's an IR library based plug in. Man that reminds me...who remembers when Waves IR-1 was like $1,100 when it first came out? Times have thankfully changed in plugin pricing!
|
|
Deleted
Deleted Member
Posts: 0
|
Post by Deleted on Jan 12, 2024 22:00:58 GMT -6
i mostly end up using Megaverb now that the Alesis is gone. Sometimes SP2016 that’s ported from the Princeton 2016 from like 10 years ago. I can’t get behind any of the EMT ports 100%. Relab I can only stand sonsig. I’m going to try the Tsar more. I tried to do a shootout between Acustica Silver when their PCM 70 library came out against my hardware but it was so different I just abandoned it. Didn't sound anything like the hardware to the point where it was laughable. Have you tried any of the liquid sonics stuff? I actually like Reverberate 3 with some libraries I've picked up over the years. Really digging in and tweaking the options there got me closer to the hardware than anything else I tried which surprised me since it's an IR library based plug in. Man that reminds me...who remembers when Waves IR-1 was like $1,100 when it first came out? Times have thankfully changed in plugin pricing! i tried the relab multiple times and could never get close to dark 80s movie adr. Sp2016 just does it instantly but doesn’t have the otherworldly modulated tails. I liked Cinematic Rooms but it’s boring and SP2016 is better than it. Tai Chi too tweaky to get what I want and it’s “yellow” and modern. You can tweak it for hours to try to get what you’re thinking of, which is like an Alesis patch lol I remember altiverb being a ton of money in 2007. Waves, only the Renaissance stuff, MV2, and like S1 hold up. Renaissance stuff is still cool and worthy but like I just paid 400 for the Weiss complete and don’t regret it at all. I paid like maybe 270 each or so for the Oxford Dynamics and Reverb? I needed a reverb with no cpu use a couple years ago that didn’t suck and it sounded good. 500 for the Massenburg stuff and I use it all the time but don’t like the bugs at all.
|
|
|
Post by seawell on Jan 12, 2024 22:13:35 GMT -6
When I woke Rick up from his nap, to ask about all this... He Clocked me all right. Really good. Now I'm relying on Dan to guide me... internal Scarlett clock? Or UA Volt? Can't eat or sleep. Definitely no Rueben sandwiches. Chris I've heard Rick recommends that you vibrate at a high enough frequency that your converters sync to you 🤣
|
|