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Post by svart on Jun 28, 2022 14:06:00 GMT -6
Zero difference. His null test nulled the audio completely. Only thing left was the power supply noise from the analog summing unit. Remarkable really. compared to a totally ITB mix? Yep. The skinny is that he stemmed a bunch of tracks so that he could either use the analog summing, or sum them digitally, to get a stereo mix. Once he did it both ways, he was able to run the null test between them. The only thing left is power supply hum/buzz. You can't hear a single trace of the audio.
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Post by svart on Jun 28, 2022 13:32:23 GMT -6
What are the cliff’s notes? On the beach Zero difference. His null test nulled the audio completely. Only thing left was the power supply noise from the analog summing unit. Remarkable really.
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Post by svart on Jun 28, 2022 11:17:23 GMT -6
As many of us already knew.
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Post by svart on Jun 28, 2022 8:15:17 GMT -6
I'm going to own up to having bought into some of those online mixing classes like URM/Nail the Mix. I figure what the hell, it's great to see others work, especially if I like the tunes they worked on. I gotta say, I learned a lot for sure. Biggest thing was being able to hear the raw tracks and the progression of how they change as the mix is built. Tracks were a lot less "exciting" and a lot more "even" than ones I track, which explains why I always have trouble getting a mix together with my tracks without tons of carving and why I have a lot easier time working with other's supplied tracks. Overall everything was much more mid-centric and it took a while to get used to it.
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Post by svart on Jun 27, 2022 7:12:17 GMT -6
Wow. That's utterly ridiculously overkill.. I mean, it's going to be 120dB in there.. Impossible to hear the fidelity.
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Post by svart on Jun 25, 2022 19:35:57 GMT -6
Here’s an issue I run into a good bit. I track a lot in the same studio, so when I get done tracking I bring it home and have a template I pull the stuff into…because it’s the same drum setup, same acoustics etc. So when I send them the first rough the day-of, they’re not hearing a board mix. Which usually sounds pretty shitty. I started doing that because one or two newbie guys didn’t understand what a rough board mix was. So - most of my time is spent tweaking…maybe different comp here on the drum bus, levels, effects…but they’re hearing something that is going to be halfish done already. Maybe I need to start sending shitty board mixes so they can hear the difference. That won't work. They always expect it to sound great. If it doesn't, they'll immediately lose faith. I know from experience.
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Post by svart on Jun 25, 2022 17:11:41 GMT -6
Consider this.. I listen back to mixes that I did years ago and think "WTF" about how scooped they are and realize that I mix with a lot more mids now, to what I think is very bright and midrangey.
And I recently compared a pro mix to one of my own very, very meticulously.
The pro mix was even more mid pushed than mine but I had never noticed it before.
I then pushed the mids in my mix to the same levels and it sounded weird, but I kept listening and it became "normal" to me. It now sounds fine when I listen.
I believe it's a case of getting used to our own sound. It's "normal" to us. It fits our ear response and our desires.
But others will have different mixes and styles they've come to recognize as "normal".
It's more likely they just have a different sound they hear in their head than you do rather than your skills falling short.
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Post by svart on Jun 25, 2022 8:37:06 GMT -6
OK, I researched prices more. Just trying to ask a fair market value so both buyer and seller are happy. I was able to find an AML ex1081 for $380 used on Reverb, so I'll beat that and offer them for $280 each. Assembled, tested, free shipping, no import duties, etc. I hope that makes it more attractive. Next, the Hairball Lola Preamp with dual 990 op-amps and the stepped input control plus variable output control (not stepped). I found one, although it doesn't state if it has stepped input (which is extra $) for $540 on Reverb. So I will reduce the asking price to $495 each. Again, free shipping, just cover any transaction fees if applicable. Last, the rare TBDD recreation of the Roland Dimension D for a 500 series rack. I will lower that to $900 also with free shipping. I hope these seem an even better deal for those interested. I will update prices in the original post. I'll pm about the AML ez1081s.
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Post by svart on Jun 24, 2022 8:19:07 GMT -6
MOTU M4 for cheap. MOTU 828ES for mid/high $.
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Post by svart on Jun 23, 2022 15:44:02 GMT -6
The waveguide on the tweeter is horizontally wider. Less reflections from above or below. I also think with a treated room, reflections from a distance will be much less of a problem than a narrow sweet spot. I much prefer a wider sweet spot that I can move around in within reason. I get that the wave guide is more narrow vertically than it is horizontally, but doesn't the wave guide still narrow the horizontal field to one degree or another, at least when compared to a monitor with no wave guide? I too prefer, if possible, a wider sweet spot that I can move around in, which is why I was concerned that the Neumann would actually have a narrower sweet spot than I was used to. But it sounds like you're saying that the KH310 has a wider sweet spot than normal? I'm confused. Edit: I went and read your review. I do see that you're saying these have a wide sweet spot. This still confuses me, as I would think any wave guide at all would narrow the field to one degree or another. In any case, it's good to know that a narrow sweet spot doesn't seem to be a concern. Waveguides widen the dispersion by increasing coupling of the diaphragm with the forward air volume as well as the reduction in baffle diffraction. This smooths the frequency response to reduce beaming in addition. I guess an analogy is blowing through a straw. The air front is very focused, with a gentle dispersion vs. blowing through a funnel backwards which immediately disperses the air front.
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Post by svart on Jun 23, 2022 15:22:01 GMT -6
Whew - boy, do I love my KH310s. They sound so natural to me. I've still never heard the LYD48s, but I'm sure I would like them, too. But seriously, wow on these KH310s. The thing that worries me about the KH310 is the wave guide. I know the wave guide is seen as a benefit to a lot of people, due to the wave guide's ability to vertically and horizontally narrow the dispersion field in order to reduce first reflection issues in a small or low ceiling room. However, I've have a large room (24x30) with pretty good treatment, and I would actually be worried that I would end up with a narrower sweet spot than I'd prefer. The waveguide on the tweeter is horizontally wider. Less reflections from above or below. I also think with a treated room, reflections from a distance will be much less of a problem than a narrow sweet spot. I much prefer a wider sweet spot that I can move around in within reason.
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Post by svart on Jun 23, 2022 15:19:27 GMT -6
Go check out my KH310D review in the review section.
I like mine too. I don't see how I could really need anything else at this point. They do what they're supposed to do and they don't get in my way. What else can I ask for?
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Post by svart on Jun 23, 2022 9:42:57 GMT -6
This is why once my recording machine is stable I turn OFF any and all updates.
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Post by svart on Jun 23, 2022 6:27:15 GMT -6
I haven't really seen any compressor take much resources at all. The only ones I see using any significant resources these days are reverbs, autotunes, amp sims or ones that need to process large amounts of signal (like Soothe, etc).
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Post by svart on Jun 20, 2022 21:29:48 GMT -6
I don't know if this is still relevant ( svart ), but I remember reading a decade, or so, ago that chips are either optimized for single or double rates, then scaled either direction depending on the selected rate. Anyone know if this ever was, or is still, a thing? Modern Delta-sigma converters are all oversampling. They oversample 128x to 512x times the audio rate. A master clock of 22.579Mhz would divide by 512x for 44.1k, 256x for 88.2k, 128x for 176.4k. a master of clock of 24.576Mhz would divide by 512x for 48k, 256x for 96k, 128x for 192k. Oversampling significantly lowers the antialiasing filter requirements and significantly lowers the quantization noise. There's almost no drawbacks this way. I should also add that it's much easier for manufacturers to build crystal oscillators in this frequency region as well. The size of the quartz crystal in the oscillator is relative to it's frequency, so the higher the frequency, the smaller the crystal can be. You can't make a crystal large enough for direct audio frequencies, and you can't make them smaller than what would equate to around 100Mhz.
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Post by svart on Jun 17, 2022 13:21:59 GMT -6
I’ve never tried a BeezNeez mic. I hear great things about their 87 though. I love my BU87. It does the U87 thing as well as any other 87 variant and/or clone. I consider it my primary vocal mic now.
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Post by svart on Jun 17, 2022 13:03:25 GMT -6
Seems that they are selling for 1500$USD. That's not terrible. I had hoped to be more around 1200$ though. It's intriguing at 1500$ but I'd buy today at 1200$.
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Post by svart on Jun 17, 2022 9:28:16 GMT -6
Neat stuff. I love designs like that. Too bad I'm not rich.
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Post by svart on Jun 16, 2022 11:17:30 GMT -6
Yes. Just make sure the seat pan has no cracks anywhere around where it mounts to the main frame. Used ones might have cracks and break around there if people aggressively bounce on them or sit down hard over time.
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Post by svart on Jun 16, 2022 11:06:10 GMT -6
Wow, svart , really nice! Definitely mathy but not too mathy. This link gets really mathy but it's fascinating, in a way: www.sciencedirect.com/topics/engineering/quantization-step-sizeSection 4.2.2 is what I read. I like their little graph of quantization error, this funny Egyptian looking little curve. They describe it as a white noise with no mean, dependent both on bit depth and sample rate, in a way I don't fully grasp (and might not need to.) The "stair steps" doomsday analog cult were right, in a way. No, we don't play back stair steps. But yes, we hear the white noise of their residue. But we don't because the noise would be in the form of harmonic content related to the quantization frequency step size, which should/would be much higher than the carrier wave being reconstructed. The digital filter(or analog in cases of a DAC) should be designed to filter off the residual harmonics.
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Post by svart on Jun 16, 2022 8:34:26 GMT -6
Some of those AT mics are dark like an old vintage mic. Any of us could put out a decent project with them. Limiters often add highs so just slamming the limiter brings life and sounds good. There is a limit to how good you’re gonna get though, and that’s why that guy probably gave up He gave up because it wasn't challenging for him. He went off and did video for a while and now he's flipping houses. And it wasn't ever about just the tones and fidelity, it was everything. Songwriting, arrangements, everything. It was effortless for him so he just never really seemed like it mattered. He was never one of us tryhards that have to listen to every tiny little detail. He just made changes and moved on. Like I said, effortless.
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Post by svart on Jun 16, 2022 7:21:21 GMT -6
I am glad I am able to read all of this now without feeling like I need to buy/try every single piece of gear or method that has been mentioned. When I first joined this forum, I was allowing myself to get caught up in all the “gear you must have” and spent tons of money buying stuff I honestly didn’t have the time to figure out how to use. Plus, being a songwriter, writing songs kept “getting in the way” because the songs have to come out. I read all of these types threads now for information and learning instead of for what I “need”. That has been like as dr. bill and monkeyxx as well as a couple of others have said, totally freeing. I have chosen a mixer and I pay him a sum to mix for me now. Just like svart I meandered around for a few years (during my "10,000 hours") until I made the discovery (we're a few decades on since then), also I met a few engineers that could use nearly anything (one was a Presonus interface with a few plugs) and it was utterly depressing how good they were.
I knew a guy who went down to GC and bought himself a Presounus interface, some KRK Rokits, a couple headphone and some AT mics on sale and went home. He produced multiple bands within the year and his projects sounded better than mine after 15 years. I was so grieved that I almost quit.. He was so good at doing it all with nothing but a few plugs while sitting in his untreated living room and here I was behind tons of higher end gear and decades of trying. After a few years he got bored and stopped recording. WTF.
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Post by svart on Jun 16, 2022 7:14:38 GMT -6
So, the consensus is that it's a non-issue? It's starting to drift into 48 vs 96 territory, but I guess I'm more interested in any technical issues with 48 upsampled to 96? I plan on running these drum tracks through outboard, so there's going to be more conversion trips through my Aurora (n). Then again through the drum buss and master buss. If you want it to be "absolutely perfect" in techincal terms, some SRC algorithms are cleaner than others. Here's a handy reference: src.infinitewave.ca/Altough, I wonder, if downsampling is "dirtier" (due to filtering,) than upsampling, which in my mind just adds a bunch of "nothing bits" above the previous cutoff frequency. Maybe @tomegatherion would know. So here's how it works. Upsampling, or adding more sample points to an existing waveform, would normally analyze the existing points and between each point would estimate a curve of some type. The algorithm would then estimate the value of the new sample based on this interpolation curve. The drawback is that the quality of the interpolation matters greatly. Some interpolators will use high order polynomial curves (with matching the order value to the signal being critical), which are more accurate but uses more resources and take more time. Some will do linear interpolation which is much faster but less accurate. Either way, there will be some error that exists, which is the difference between the estimated(interpolated) point value and what a real sample value should be. This error in value is also quantization error. A way to reduce this error is to oversample the original signal at some very high samplerate so that the quantization steps are very small at the cost of much higher processing and resource requirements. Once you get a higher accuracy signal that has been upsampled, you can then reduce the sampling rate once again. This works for A/D converters as well as software sample rate conversion. You then have to Low Pass Filter so that you remove the small amounts of harmonics that appear at higher frequencies from the resulting quantization steps. Downsampling is effectively the inverse. There's a dozen different ways to do it but the most common algorithm will decimate, or remove certain samples based on a ratio. This filters the signal then removes the samples required and may or may not apply more filtering. Neither one would realistically be "dirtier" than the other if properly done but upsampling would have more chances to introduce error since the process is more intensive and introduces "ghost" samples.
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Post by svart on Jun 16, 2022 6:28:57 GMT -6
I worked with an artist who's album was primarily tracked by another producer where I was brought in to cut some overdubs because I was local to the artist. When things were tracked and it came time to mix, he liked my roughs better than his producer's mixes, so I ended up eventually getting the mix gig. I did a couple of the mixes that the artist loved and then got a conference call from the producer saying he was "concerned that we are compromising on quality" because I was sending back mixes @ 48khz instead of 96kHz. Meanwhile, while he was concerned with what our furry friends can hear, down in the audible spectrum, the sessions he was sending me were littered with bad edits and poorly intonnated guitars. So yeah.... seems like people get too hung up on the wrong things sometimes and miss the bigger picture. So many stories like this. I had a guitarist (seems they are generally more anal about small things than anyone else, second only to vocalists..) who would nitpick various small things in his tracks like single string plucks.. But not be concerned that string bends would land on a sour note or that his guitar intonation was off at higher notes. Or a vocalist that was super concerned about a hard T sound in one word on one track and we did maybe 30 takes over a day on this one phrase and I could barely tell a difference between the takes.. In the end the artist was so upset it wasn't working she was about to ditch the whole tune out of anger. I ended up ducking the T a little in automation with a little de-esser and it worked well enough she was OK with it.
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Post by svart on Jun 15, 2022 8:01:16 GMT -6
Upsampling will simply sound like 48K but at 96K samplerates.
BTW, I can hear a difference between 48K and 96K when A/B'd. Will it make a difference in your mix? Maybe, maybe not. Depends on the quality of the 48K. I've heard 96K that sounded worse than 48K, so YMMV.
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