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Post by Hexspa on Apr 18, 2024 14:55:19 GMT
You can go to audio test kitchen (dot com) and listen to hundreds of mics. They let you filter by cost, brand, and other options. I particularly like the Shure KSM range but the Audio Technica stuff is good too, just a different voicing. None if it is that expensive. If it's out of your budget, simply buy used. With a little patience, I've been able to get discounts of around 50% on premium items. There's absolutely no need to dip into the cheapest, cost-cut items if you follow these steps.
It's important to remember that the price of something is what the seller hopes to get, not necessarily the minimum that they (or someone with something similar) will accept. By no means am I saying be cheap and always play hardball but let's just say you can save money if you want to.
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Post by Hexspa on Apr 13, 2024 8:41:44 GMT
I think it's important to remember that a lot of the reason that cheap things are made in China has to do with greed. Companies who are not based in China, like Shure and sE Electronics manufacture there but are generally not associated with producing cheap products. sE Electronics maybe more than Shure but they make the RNT there, afaik, which is a microphone made in collaboration with Rupert Neve that I think sounds pretty good. It's also not cheap. Actually sE Electronics is based in China now apparently but was originally founded in California.
So there you go: it has to do with the capitalists who outsource for reduced labor costs then push material costs further down by using poor designs.
Let's keep the discussion to the quality of circuits and components or market forces and minimize emphasis on country of origin.
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Post by Hexspa on Apr 2, 2024 3:06:13 GMT
I'm wondering about 3" depth quadratic diffusers. Low frequency cut off of about 1600hz. Say a reverberation field of 1600-8000hz Since I will have very little to no reflected or diffused energy below 1600hz, then maybe a diffuser in this frequency range could work well. All I really know about diffusion comes from the EBU white paper I always reference. They call for a pretty linear decay profile throughout the mids and highs except for between 4-8kHz iirc. To me that indicates a preference for diffusion. Given that the bandwidth is just one octave and the frequencies are high, there's no need for deep diffusers if you're going according to them.
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Post by Hexspa on Jan 12, 2024 7:13:12 GMT
That's impressive. Well done and congratulations. Thanks!
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Post by Hexspa on Jan 6, 2024 21:55:46 GMT
Hi, just wanted to thank Ethan for his help again. It's nice to be able to work through acoustics problems without depending on advice from random people online. These seem to be the best measurements I've taken so far and I wanted to share. Yamaha HS50 and HS10w (-10dB @ 30Hz) @ 1m @ 74.4dB. 18 measurements around listening area, nine at standing, nine seated, averaged then processed in REW to produce filters which I then imported into EqualizerAPO. Looks like +-4dB SPL below 200Hz (not smoothed), -32dB impulse response envelope @ 20ms, the "20dB decay within 150ms above 63Hz spectral decay target with a smooth taper" target is mostly there and a clarity graph which I don't know how to make sense of except some people like it. This is what many years of interest in acoustics and a drinking problem will get you folks! Anyone could probably do it in half the time. Discuss?
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Post by Hexspa on Jan 3, 2024 21:12:55 GMT
Translation is one of the Great Pillars of mixing. Beyond technical requirements, you have to account for preference and people differ.
A debate that still rages is about the merits of "flawed but popular" vs "objectively flat" speakers. This is an oversimplification but your question about using your TV speakers to mix is in here. The answer is it depends but it's probably not going to work.
Of course, it's possible that your speakers suck in just the right way to compliment your room, where they're placed, and your unique hearing ability. However, it's more likely that using neutral monitors in a neutral space will help you produce a product that will sound good in the greatest number of systems. The logic is 0-y is y. In other words, if your room and speakers neither contribute nor detract from the source material then whatever crap your audience listens on will only make it deviate from neutral. In contrast, if your whole system had a 10dB peak at 1kHz, you'll under-represent that range. Then if they have a null there, then it's bye-bye intelligibility.
By all means, reference your mixes. I'm of the thought that your main source of speakers a probably better off flat. If you want to juice them, then sum to mono in one speaker (true mono, not dual mono) with a high pass filter to about 250Hz. The combination of those things will give you a fresh picture of your mix, remove low frequency masking, enable you to focus better due to listening to one speaker (Harman) and mono itself makes stereo mixes sound different as well.
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Post by Hexspa on Nov 25, 2023 18:21:19 GMT
You agreed with me that there's no stereo in the real world. Sure, we have two ears but I was referring to sources.
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Post by Hexspa on Nov 20, 2023 14:15:58 GMT
Re #4 "Why would you want mono to sound "wide"? The whole point of mono is that it is not wide." Ok, you are sitting a concert hall and there is only one piano played and you hear it by both ears and it sounds like stereo, right? So why would you want it to sound unnaturally narrow if you have only one speaker like old vinyl players? You would definitely place the player somewhere to have a wider sound. The thing is that room itself is a powerful stereorizer but if you put speakers in equilateral triangle along the longest side it will be much worse (no stereoscene at all) than putting in corners with some diffusion by carpets and bookshelves. Plus a benefit for listening old mono records. Plus you can more freely move, shift your head from the centerline with less adverse effects. Say you are listening together with a couple of friends. The central image can be stretched though to turn singer into a giant but it can be to some degree fixed by dampening and diffusion. Another benefit as i have written before is that all your apartment has the same frequency response and you enjoy music from any square meter,not just as part of that triangle. Cinema houses work this way by the way... you should try getting the same effect and i firmly suspect it's possible even with one speaker. Sound in the real world isn't stereo; stereo is a mechanical reproduction via two channels. Mono is a function of the same repro system so any correlation is bound to be incomplete between a physical source and a stereo one. For example, a piano or a vacuum cleaner has many sources of sound, each contributing to different parts of a composite whole. There are "sound cameras" which let you visually observe this. In contrast, a loudspeaker generally has a tweeter and a woofer and they're close together. It's obvious that the two, a piano and a loudspeaker, can never produce the same exact sound in a space. It's well-established that stereo has its limitations - therefore there's no disagreement with "official science".
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Post by Hexspa on Nov 20, 2023 14:12:37 GMT
I have read the entire book by Ethan, so there is no need to repeat the statements from the book, especially about subjectivity. Nulling tests don't prove anything by one important secret reason: say reaper daw is secretely designed to spoil sound (which is true imo!), you put dualmono file on track, invert phase on one channel and voila - sound disappears. Does it prove anything? Only that channels were identical. But it does prove that reaper has not made any adverse manipulations to the sound? No! Because nulling is made between l and r channels but not between original left channel and same channel after import into reaper. So it explains why daws produce very different timbres due to different resamplers and plugin algorithms. And those of reaper are of very low quality. The majority of delay plugins do not work in the honest way and you can detect it only in one way: if you try sweeping from 0.1 to 10 ms - comb filtering should be severe and very audible. If you test readelay you will find that comb filtering is very weak and barely heard with same crap happenning when shifting clip's duplicate - i.e. some muddy hazy sound instead of crisp comb filtered result. And 99% of delays do the same crap. I won't discuss drawbacks of other reapack's crappy plugins. I just wanted to point out that unfortunately Ethan's book does not reveal all secrets and manipulations from corporations. I can only recommend using very old plugins. The secret malicious trend by corps is proposing complex plugins like multiband crap, etc, instead of giving simple but honest tools to people. Re point #5. I have tried recently to create a comb filter as 5-band readelay with bands created by 5 12-pole filters and 5 readelays. All this worked in dualmono and to my suprise sound just desintegrated even when delays were set to 5ms difference and changed timbre even with 0.1ms delays. You may try this test yourself. Sound remains mono but it kind of loses focus like if you record reverb instead of direct sound and all this happens in mono! Unbelievable stuff! Correlometer won't show anything. So, you again can't detect phase shift but you hear it perfectly. Same crap happens with min phase eqs... for that reason linear phase eqs are better even with ringing.... And why honest delays are hard to find? Because in comb filtering mode they can fix some crappy room comb filtering which was recorded in spaces less than 100m2 or just crappy timbre. You can't fix it by equing as it requires thousands cuts, so corps make people buy more and more eqs in the hope to find a magic wand. And any daw/tool does not reveal its own spoiling caveats when being used for nulling tests. Perhaps old hw oscilloscopes are the way to go. I just want to underline the fact that my own listening experiments have revealed all described above crap and nulling tests helped in no way. I would switch to some old daw from 1997-1998 if i were a beginner. At that time malicious algorithms had not yet been implemented imo. Yes, there were aliasing artifacts and 16-bit cpu friendly processing but those algorithms were honestly working as good old hardware. And by the way i am the first to have disclosed the truth about such simple yet powerful tool as delay and corps' malicious evil deeds. So, just repeat the tests yourself if anyone is hesitant. By the way i agree with Ethan that sonitus and Budde's plugins were not that bad and Budde has even created first real vector sampling software which promptly was monopolized by corp acustica audio selling now pseudovector software. Re #2 I don't see the point for debate. In some cases lowering mids produces better result as it decreases collision of mids of different instruments where raising highs would not help - the simple truth not formulated by anywhere for a strange reason. My ideas about cheap chinese mics: there is like can timbre with them which speaks of some comb filtering. As Ethan have written their plots' resolution sucks and indeed how will you show thousands of peaks and nulls for highs in a viewable plot but the averaged plot may boast of raised crisp highs. Same crap about chinese plastic guitars and their strings: they may be resonant, solidly made and painted but with some crappy toy like sound. It's like genetically modified food: you can eat it but the results will be adverse. Aaron Russo's interview about the elite is quite informative about present-day products and intents: corps just do not care and do harm to people and it's done in all fields including audio. I'm not going to address speculation on Cockos' hidden agendas. That's called conspiracy theorizing and I'm not into that. Regarding how delays work, there's probably reasons for why they work how they do. Obviously digital delay works different than analog because the processing has to be broken into buffer chunks; that's as much as I know about it. As far as Reaper's resampling being "very low quality", I suggest you talk about it on the Reaper forum where someone there can challenge your ideas. Posting here about it is like someone complaining to a pastor about a politician - go to the arena where they fight the battles! Go to a DSP forum if you think you have a valid contribution. Instead you are tearing everyone down outside of their presence and that is cowardly. As far as the null test is concerned, simply import a file into two separate daws and render it with identical settings. If they don't null, you'll have a leg to stand on. Until then, it's all idle speculation and I will no longer contribute to it.
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Post by Hexspa on Oct 16, 2023 13:05:36 GMT
The forum seems dead as well as audio industry no longer required for brainwashing. No possibility to earn anything on music.... Not totally dead. Your reference level has no way of "meeting" the "fletcher-munson curve". The fact is our hearing is not linear in terms of frequency-to-amplitude. While most experienced mixers agree that the bulk of mixing is best done at moderate levels, you have to turn the audio up to hear the frequency extremes. Nobody is trying to "meet fletcher munson" as much as there's a framework and you have to tell beginners something. It's a complex issue that you can't expect a beginner to understand. Even for someone experienced, it takes concerted effort to appreciate the nuance to this topic. You might be referring to the axiom of "better to cut than boost when eqing". Technically and in isolation there's no difference between creating a curve via negativa or via positiva but within the context of a mix you have to consider headroom. Plus, to your point #1, louder sounds better and boosting is adding loudness thus deceiving you. In many creative fields, you're always battling noise to some extent. Digital has helped but even then you have physical sources of noise. Saying that "chinese mic do not capture high freqs at all substituting them by noise instead" is a gross generalization which can be easily disproven. You then say "cheap chinese mics" but then that is still too vague. If you want to talk about the response of a particular mic then that will be much more fruitful. In the mean time, there are hundreds of mics on audio test kitchen complete with frequency response plots created in collaboration with Harman. You will not find a more comprehensive source anywhere - especially with audio examples. Based on your rendering settings, you can get a different result with a printed file than you hear live. Things like lossless vs lossy encoding, sample rate and bit depth (in extreme cases) as well as resampling modes can all affect the sound. However, if you expect to be taken seriously by anyone who can do anything then you need to speak in objective terms. All these subjective words you use mean nothing to anyone but you. "Hard, crisp, worse, soft" etc. are all sales terms used to bamboozle the foolish out of money. You've come to one of the few places on the internet where this kind of discourse will not fly.
Why would you want mono to sound "wide"? The whole point of mono is that it is not wide. Some listeners do prefer more 'live' implementations of their listening room but not all. If you want a source for this kind of approach, check out noted expert Floyd Toole.
I'm not a psychoacoustics nor hearing expert but how people hear is pretty well known. Interaural Level Differences (ILD) and Interaural Time Differences (ITD) account for all of our frequency range in varying amounts, I believe. William Moylan's Art of Recording goes into this in some depth. Mixers can use speakers or headphones. I use both and use speakers as the final tiebreaker. Saying that "if it sounds good on calibrated cans it will sound even better on any speakers but not vice versa" is using subjective language and is probably misleading at best. You also say a "well treated room ... is costly and ... inconvenient for living" is another subjective and potentially misleading statement.
Whether you think you can or you can't - you're right!
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Post by Hexspa on Sept 17, 2023 20:19:33 GMT
Welcome to the forum! First and foremost, did you read the stickies and the top and all Ethan's material? If not, please do so. In general, everyone has the same problems so the answer is pretty much the same for everyone. After you measure your room to get a baseline, you'll systematically plan, deploy, remeasure, repeat. Here's a link to another thread, please watch the videos: the-audio-expert.freeforums.net/thread/712/acoustic-analysis-drunks (watch number 1 at the bottom first) After that, I hope you have some different questions. I will mention that most of us here are using porous absorbers for RFZ and bass traps and that's what we overwhelmingly recommend. Hey, rock. Thanks for recommending that article. However, I've since taken that material down and just deleted the thread. Without going into all the reasons why, I'm concerned it may not be the kind of messaging I want to perpetuate going forward. If you could update your post by removing the link, I'll appreciate it. It's not a huge deal, I just don't want anyone to click dead links. Cheers.
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Post by Hexspa on Jul 31, 2023 19:21:05 GMT
All I can recommend is some active pickups like EMG and Fishman or some kind of "noiseless" pickups which I believe are actually some kind of humbucker. Speaking of which, there are also single-width humbuckers; though of course will differ in sound.
My experience doesn't give me the ability to recommend "electrical" solutions because I'm only now learning about resistors and computer circuit modelling via SPICE!
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Post by Hexspa on Jul 26, 2023 16:02:24 GMT
Generally 20, 32, and 45% any given dimension is the mathematical best location for even bass so I'd start there.
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Post by Hexspa on Jul 25, 2023 15:36:53 GMT
I had juli@ and still have delta 1010lt. Their inputs were very noisy for recording with noise floor -35db. I suspect all pci cards behave the same. After acquiring rme hdsp noise floor became -48db so at least i can record with noise reduction by audition. But all pci cards are better for realtime latency than usb/firewire. So it's best to record via external card, send recorded sound via spdif into pci card and set the daw to use pci card for lowest latency and of course on linux with rt kernel. I guess it is a marketing trick to make inputs of pci cards noisy because someone claimed that old soundblaster's inputs were not noisy. By the way agp port for video was a good thing for rt as soundcards used pci and they did not conflict by irqs. Pcie and pci seem to conflict. I'm surprised you're finding the RME soundcard to have such high noise. These technical matters are not my forte but make sure you're measuring it right. You can also visit the RME forum if you aren't satisfied with the performance. They've been pretty responsive for any questions I've had.
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Post by Hexspa on Jul 25, 2023 15:31:09 GMT
Keeping it open is probably better. The best thing would be to knock down that dividing wall and use the whole space. This would give you lower fundamental modes and lessen the strength of reflections. Sylvia Massey does a similar one-room approach.
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