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I have a microphone cable that goes from female xlr to male tr jack and it picks up hum. I screwed open the connectors and I see nothing wrong. I'm not sure where to go from there to fix it. Seems like I could only do a worse job resoldering it. With the same microphone in the same interface input and an xlr to xlr cable, it doesn't happen. I've also got a couple of cheap microphones with a lead ending in a jack permanently attached to them and they don't show the problem either. So I'm guessing it really is that specific cable. Right?
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Jun 17, 2015 11:01
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Hollis Brownsound posted:Well an XLR > TR isn't a balanced connection, so it could just be picking up RF interference. Guess there's nothing I can do to improve the cable itself then. Thanks.
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Jun 17, 2015 14:15
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Selim Sivad posted:Does anyone know of an Omnichord VSTi? As a guitar player who learned everything in terms of chord shapes, I really like the idea of being able to play a chord on keys without counting steps, or even thinking about what notes are actually in it (I am a bad). The only one I could find is for Kontact, which I don't plan on buying for just a crummy Omnichord. Cthulhu doesn't literally (visually) mimic an Omnichord, but it allows you to map a variety of chords across the keyboard. Since you'd access an Omnichord vst through the keyboard anyway, it seems like it would end up being pretty much the same thing, but more flexible (map whatever chords, play any vst through it and a good arpeggiator to boot).
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Jul 1, 2015 00:43
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NTT posted:I'm getting these strange 'not perfect L/R output NTT posted:does that mean I need to have an adapter that specifically has tip/ring?
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Jul 2, 2015 09:50
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NTT posted:
Maybe the headphone out on your interface is broken as a consequence of all the jiggling. Or -I'm not really familiar with the Audiobox- you simply have routing issues. Like, say, you're recording a single microphone onto a stereo track and it all ends up in the left channel. Jiggling the plug could make both the tip and ring (or ring and sleeve, I forget) make contact with the same contact point, sending the audio that by all rights should only be in one channel to both, giving you the impression that you 'fixed' it. Where you probably should just tell your DAW to record to a mono channel instead, or say in the driver control panel of the interface that the input you're recording from isn't part of a stereo pair or something like that idk.  Could just be a poo poo adapter, hard to tell. Suppose you don't have another pair of headphones with a 1/4" jack to try if that works properly?
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Jul 2, 2015 12:27
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I have had zero problems with 1/8" inch jacks over the last 20 years, but I guess. The solution for the littlebits thing seems it should be the same as for the Volcas, that have the same problem of out of phase 'mono' with a stereo jack: buy an adapter that splits the stereo in two mono jacks and then only use one of them.
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Jul 2, 2015 12:46
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chippy posted:I'm hoping to use Ableton as a looper rather than a separate pedal that's just feeding in to the mix. Really I just want as low a latency as possible as I seem to notice it quite a lot, and it seems like looping might be kind of difficult if what I'm hearing is delayed. Radiapathy posted:If the OP is going to recommend brands/models, I challenge the blanket suggestion of Focusrite interfaces. No experience with the Saffires (although I hear they're good), but the entry-level Scarletts (2i2, 2i4) have quite possibly the worst latency on the market, and generally have fewer features and less impressive specs than several other current interfaces for the same price.
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Jul 16, 2015 12:48
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Sexy Randal posted:Is this a limitation of the mixer (Peavy PV6 USB)?
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Aug 2, 2015 19:52
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strangemusic posted:Definitely don't just push the master down.
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Aug 27, 2015 12:47
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^^^^ E: The static is there alright, it's even clearly visible in the soundcloud visual preview, which means it's objectively pretty bad. Yeah, that's the self noise floor of the mic/built in preamp itself. What are typical distances between the sound source and the mic? If you mic up closer, your signal will hotter, allowing you not to turn the gain up so high. This will keep the noise level lower, relatively. You will lose some room ambience, which you can compensate for with judicious use of reverb. If that doesn't cut it, a new mic with lower self noise is the only other solution. At the price point, the new Rode NT1 is supposed to be amazing on that front, but you're looking at $300 - $400 if you are going to have to buy an audio interface with ok mic preamps as well.
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Sep 22, 2015 12:47
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Vulcan posted:Can anyone tell me if I can consolidate my audio equipment? It will take the mic, it will take the soundboard, it will take the audio output of your computer, mix it all together and feed it back into the computer towards Mumble or whatever, while still letting you select which inputs are going to your headphones and which aren't with the cue/pfl function. Sound quality will probably be on par or better than what you're using now. Not pro recording quality, but decent. Headphone out probably not top of the line, but fair, considering what you are doing now.
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Oct 22, 2015 12:13
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Cage posted:Hello thread. I haven't dabbled in recording in probably 8 years or so. My setup back then was an a m-audio mxl 900 microphone, a too large behringer mixer to power it and then connected that to my creative audigy sound card. I want to simplify everything if possible. It seems like I can use the recommended Focusrite Scarlett 2i2 to power that condenser microphone instead of using my behringer mixer right? And using a usb interface will let me bypass any fancy sound card I used to use because the audio is being sent via usb to the computer right?
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Oct 28, 2015 22:08
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Cage posted:I've seen a lot of driver issues on the amazon reviews. Anything you would specifically recommend? Anyway, I get triggered by the default recommendation for the Scarlett. I'm not convinced the case for it is as ironclad as commonly is assumed, seeing as it has double the latency some others in its price category have, for example. But I'm really not up to speed anymore and things got a bit more complicated with usb 3 tripping up otherwise perfectly fine recommendations. And there's new stuff coming out all the time. All I know is in the audio interface thread anyway, so browse through that a bit.
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Oct 28, 2015 22:44
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Get a Roland A49. Anything labeled as made for Ableton will likely work as a generic midi controller, but will lose most of the special features. Anything that requires feedback from the daw to the controller basically, like different colors of light behind drumpads meaning anything, for example. Might also miss out on some smart presets and automapping maybe, in some cases.
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Nov 4, 2015 00:39
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Southern Heel posted:I can't switch to the ME-80 ASIO driver because it doesn't support audio-out via my soundcard. Use the ME-80 asio driver, get rid of asio4all, connect your speakers to the ME-80's rec out, make that the default output for everything, you should be sorted, right? Barring the desire for surround sound for other purposes, maybe. If not, you need another real asio capable audio interface and record the analog output of the ME-80 into that. Asio sort of requires you to use the same device for input and output, anything else is asking for trouble, especially if you throw onboard audio into the mix.
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Nov 17, 2015 15:41
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Southern Heel posted:So if I understand correctly, the solution would be to pipe EVERYTHING from my PC through the ME-80 - system sounds, videogame sounds, recording input, etc. ? I'm trying that now. Southern Heel posted:If I can't hear pops and crackles in REAPER but I can in my outputted MP3, is there an easy fix for that? Here's the settings:
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Dec 4, 2015 12:44
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The StripBus effect he's using is basically an equalizer plugin. It was affecting the audio output of the Addictive Keys piano plugin. It wouldn't at any point affect midi data. It would ignore midi data and even only have the opportunity to do so if Addictive Keys passed through the midi data (which typically is the case), because the plugins indeed work in series. But it's not relevant to the StripBus, because it's an audio processing plugin. What he's doing is taking the audio output of Addictive keys, processed by the StripBus and turning it into a wav. At that point no midi data is required anymore to hear sound, so he disables the midi clip. And because he has the wav on the same track and doesn't want it to be processed a second time by the StripBus (and the piano plugin doesn't get any input anymore), he disables the plugins for that track, thus saving cpu cycles. Because playing a wav is easy compared to generating and processing it. In audio, bus (or buss) can have a couple of meanings, but in this case it's a bit of a red herring, I think. Nothing is actually being bussed. It's just the name of this vst combining a vu meter, a gain knob and an equalizer. Some plugins use this word in their name to indicate that it might typically be used on a bus (like where you combine the audio output of several channels into one), whereas strip typically is used to say the plugin combines several functions you might find on a mixer's channel strip (like metering, gain, pan, eq, compressor, exciter, volume, etc) Hope this clarifies rather than confusing further.
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Dec 5, 2015 21:00
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Wrath of Mordark posted:I see. Yeah, no way that Stripbus plugin is doing anything useful there, afaik.
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Dec 5, 2015 21:19
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Vocal mic = Neumann U87 of some particular vintage or just a new Neumann U87 ai. Guitar mic is maybe a black Neumann U87 ai or maybe a new Rode NT1 with that white dot. There isn't really a good shot of that. No idea what's used on the piano. The NT1 isn't half bad in any case -very low self noise- and not excessively expensive. Those are my guesses anyway. There's a lot of other things in play, obviously. Room acoustics, mic setup/distance, preamp, possibly equalizing, the timbre of his voice. Don't think buying a particular mic will make you sound like that just like that.
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Dec 9, 2015 14:39
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Captain Apollo posted:Also - I should always be recording in stereo right? It's going to depend on your DAW whether it otherwise would be a bit of a hassle, but I'm glad not to have to think about whether a routing mistake on my part is going to make a signal end up in the left channel only, I'm glad I don't have to deal with separate mono-to-mono or mono-to-stereo versions of effect plugins (or know whether each plugin inherently supports multiple modes). What I trade in is some extra disk space and cpu power, as a two channel file from a single channel source gets needlessly stored and processed in duplicate, obviously. If you want to run a lean project, or bump into the limits of your machine regularly, the proper way to do things is using mono tracks for mono sources and keep them as that in the chain as long as sensible (ie. until you start to apply stereo modulation or delay effects or other spatial stuff). I imagine it is of a greater importance when you're recording a whole orchestra at once and you risk bumping into read/write speed limits of harddrives and things like that. As far as actually recording things in stereo, with two mono or one stereo microphone goes, it's not always a benefit when mixing. You can get weird phasing and comb filtering effects if the mics aren't aligned properly and the signal in both mics doesn't differ significantly enough. It can be a nightmare to place a recording like that in the mix where you want it. In the tetris of mixing, simple shaped blocks simplify your life enormously. Basically only do it with explicit intent and in full confidence you can place the mics exactly where you want them so that what comes into your interface has the exact effect you want. You can obviously experiment and maybe toss one side (one mic) out, post recording, with no ill effects if either of the mics picked a sound you liked. For vocals, there's no point to it (also because you will want them to be dead center in most mixes). For acoustic guitar or acoustic piano, you might, but it's an art and the key is in having each end pick up a signal that differs sufficiently from the other. Left and right sides of a piano live in differing places in the audio spectrum because of the high notes and the low notes being on each side. Acoustic guitar close up might have sharper attack vs midrange on different places along the length of the instrument that could be worth combining in stereo. Note that that is different from just plopping someone in front of a stereo microphone and hoping for the best, just because stereo would be something that you "should". There is nothing wrong with putting one mic (per instrument) in the best place you can find and going with that; it's in fact quite a common practice that can save major headaches.
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Dec 21, 2015 14:13
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Roland Duo Capture Ex is $20 cheaper than the 2i4, has half the latency and cleaner preamps that can actually properly accommodate instruments.
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Jan 21, 2016 18:28
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Just buy an extra preamp so you can plug in whichever mic you like.
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Feb 18, 2016 21:04
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mike12345 posted:I finished mixing my song (to the best of my abilites), and now I want to export it to wav and mp3. But how do I know if it's loud enough? I mean I could dl some current pop songs, load everything into my audio player, and fiddle around till it matches up. But there's probably a standard, right? This means that as a producer, mixer or mastering engineer, you can still make your music however quiet or loud as you want, but on playback through these channels, the level of your material will be adjusted to +/- -23 LUFS (a perceptual loudness unit). If you made your material louder than this through compression and limiting, it will be quietened on playback and you will have lost dynamic range for no gain. Although these standards are only adhered to in broadcast and you can still crunch the gently caress out of everything on cd or soundcloud "because everyone else does it", if hypercompression isn't a stylistic choice for you and you don't know where to start, it's not a bad way to go. It will be fairly conservative in comparison with what has come out in previous years, but it will not be "objectively too quiet". You will need a loudness metering plugin. Newer DAWs may have them built in. There are also a number of plugins available. They can be found by googling "ebu r 128" with the plugin standard you need. Ideally you'll also have to read up on how to interpret the numbers correctly, because loudness can be averaged over different time periods, for example. "Integrated" (the whole length of the song) is generally the number you should be looking at. Here's a free vst plugin for Windows. Even if you choose not to follow this standard, a loudness metering plugin is a useful tool to objectively compare your stuff with whatever reference material you choose.
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Apr 1, 2016 16:46
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mike12345 posted:Alright, I watched some Youtube-video, and played around with Hard Limits and Amplify. That Loudness app crashed Audacity, so that probably means I'm super-loud. Comparing it to other Retrowave/Synthwave songs in Foobar still makes it sound anemic, but whatevs. Would any of you guys maybe give it a listen and provide some feedback? I'd pm you the link, or email. Audacity crashing doesn't mean you're super-loud, it means the vst host functions of it suck. Consider trying Reaper as a host instead. If it sounds anemic, then it's probably quieter than what you're comparing it with.
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Apr 1, 2016 19:01
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mike12345 posted:Yeah, I was joking with the loudness crashing Audacity. Ok I dl'd Reaper and dropped the file into its window. When I run that loudness plugin it just says OVER. Hmm okay.
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Apr 1, 2016 20:01
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Oh, if that's the sort of price range we're talking about, I'll be sure to mention the Rode NT1. Ultra low self noise opens up more possibilities than you think in positioning as well as post processing.
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Apr 3, 2016 23:37
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Didn't see him mentioning it was an mp3. Also, mp3 encoding at normal bitrates is pretty good these days and you wouldn't hear a difference between it and a level matched wav. My guess is the wav is playing at a different level and it's Fletcher-Munson rearing its head. Like, my audio interface has different maximum levels in ASIO and WDM modes and if I export something out of my DAW and play it back in a media player, suddenly it all sounds just that bit different, but it's just a couple of dB louder. Another rookie mistake could be dragging the exported file back into the project and forgetting there are effects enabled on the output bus. Normalizing from peak -1 to peak 0 is pointless at best. But doing the mix again and being happier with it is great. You were worried about maybe it being too quiet. At another point you said the old mix sounded wimpy. These are not the same thing. You can test how far you can go into sausagifying a punchy mix and end up with something louder but still cool. You cannot turn a confused mix into something punchy by turning it into a sausage, nor will backing up from the compression on a loud and wimpy mix automatically give you a punchy one. So if you have a mix you're happy with, however relatively quiet it is, you're in a good place.
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Apr 12, 2016 14:22
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himajinga posted:This might seem like an insane question but is there any way to take a wav file that has had eq applied to it and exported to the original wav file pre-eq to create an eq difference pattern and apply it to another wav? How well this works, sort of depends on how close source and target material match. It's not magic.
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May 4, 2016 17:15
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Ball Cupper posted:Done a bit of poking around. Tried looking at the power saving settings again but no luck. Reinstalled the drivers, didn't do anything. Plugged it in to the USB 2 port instead of the USB 3 one, didn't do anything different. If not; pretty weird.
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May 11, 2016 13:14
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There are loads of interviews with recording and mixing engineers and related articles in the free archives of the SoundOnSound magazine. Also Q&A's and how-to's. They are full of little tidbits like that. Like, any tip brought up here in the last couple of posts is something I've read before, almost verbatim sometimes. Anything from 2000 on up until 6 months ago has the full article freely accessible online, I think. Open a couple of tabs while you're on the shitter. Probably skip the gear reviews for these purposes. I'm sure there are books, blogs, podcasts and video channels a-plenty as well if the article format doesn't suit you. What I'm saying is don't wait for problems to crop up to then google for a solution someone almost equally inexperienced posts on some random forum or re-invent the wheel yourself. Pre-emptively read up a bunch on the experiences of people that have been doing this for a job for years. None of the problems you're (bound to be) having are new.
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Jun 2, 2016 00:19
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CaptainViolence posted:I've definitely read the SOS Classic Tracks articles and got some (but not a ton of) useful advice from them, but you bringing this up let me stumble on the Secrets of the Mix Engineers series of articles, and holy poo poo. This is exactly what I've always wanted to know about mixing: how they set up their tracks to mix, exactly what effects and plug-ins they use to get whatever sound they're looking for, what sort of EQing and bussing they do. It's actually incredibly intimidating, and I'm just now realizing how basic and amateurish my mixing is compared to the professionals. This is amazing. But also general good practice, like eq'ing without soloing, which at one time was something that really went against how I used to work. Basically only solo while fixing real technical shortcomings. Otherwise let the context point you in the right direction. Or the simple realisation that even a basic folksy track in a professional studio is built up from five times the number of tracks I would use at home for anything, with stuff double mic'ed from different places, an ambience mic, re-amped guitars and extensive level automation. Like, if you're also a performer, it's good to know that even the most legendary musicians' recordings go through a bit of tarting up and that you shouldn't be embarrassed per se if your vocals aren't perfectly level already going into the mic. You should work on that, probably, but not cower in shame when comparing it to finished professional recordings. I also love the interaction between people involved. Some are closer to the process than others and each has different vision and input, which can contrast with the tunnel vision (or lack of vision) when you're just one person doing everything at home. It can be really hard to be in that situation and also try to be the guy that can say: "this isn't it; let's wipe the slate clean and start over". This broader stuff I find more interesting than what specific mixer or compressor was used with what settings. I'll never be able to afford some vintage Neve desk anyway and settings are heavily programme dependent to the point there's no use copying them. Though I do love me the Q&A section as well. How do I mic up a banjo? In essence, I don't care. Wouldn't have come up with the question in a thousand years. But the answer will probably make a comparison with general guidelines on how to mic up an acoustic guitar, which maybe is useful to me. Or tell me about how microphones in general perform differently under different angles.
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Jun 5, 2016 09:56
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There is an audio interface thread, but it's very low traffic these days. One of the surprises that came forward in there was the Zoom R8 as a low cost, low noise, low latency marvel with a bunch of added in extra conveniences. These plusses would also go for the Zoom R16 that can record 8 inputs simultaneously. If you're vaguely swayed by internal effects and stuff, this thing is great. Also works as a stand alone, battery powered recorder to sd cards. Doubles as control surface and what have you. I just had my hands on a predecessor of that Yamaha mixer and it's noisy as hell. Don't know if that counts as reliable data on the new one though. There's nothing wrong with the 18i8, but I'm personally absolutely swayed by the novelty factor of that Zoom thing. I'm not sure if combining the two Focusrite interfaces through the digital connections is possible (ie use one as a sort of port extender on the other). But that would set you for inputs for the future, of course. The new behringer mixers with the red sides can function as a multi-in audio interface as well, and are supposed to be better than the previous generation as well, but you're not saving any money with them. Only if you were looking for mic inputs and hands on onboard eq in the first place.
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Jun 5, 2016 15:54
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I mention it because according to the introductionary chapter of every book on mixing I ever read, the overvaluing of memorizing compressor and eq settings is a widely held mistaken belief. Even though I've never seen anyone really ask for stuff like that and most books subsequently can't help themselves providing tables with instrument specific frequencies and good starter settings for compressors anyway  That Muse setup sounds insane and borders on acoustic synthesis.
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Jun 5, 2016 23:59
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If any of it works, it's on headphones only. It can't be done for a typical stereo setup of two speakers in a room. You can use some sort of binaural panning plugin, and create an impression of something moving up and down on headphones. Doesn't work as well for a static placement, because the disembodied sound you input in typical music production is so abstract, you wouldn't know the difference between it being equalized to be brighter or duller as an artistic choice or for the purposes of vertical placement. To create the perfect illusion, you need more than what these plugins typically do and have a pretty much perfect room simulation integrated with that and then other sounds also taking place in that room to give you a frame of reference, because these all contain cues the brain uses. And again, that poo poo plain doesn't work on speakers unless an actual top and bottom speaker pair is added. It's interesting from a research and experimental point of view and the easiest way in is getting a pair of binaural mics and a styrofoam head to record stuff in actual spaces, because the way eg. reverb typically simulates a space simply won't do. But the practical applications in music are very limited.
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Jul 4, 2016 23:08
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That's fine, as long as you're not sending the mix into the red during the louder passages. Simple level riding is something that's done quite frequently and into detail (word or syllable level) in pro mixes to get stuff not buried. Though if you're recording the vocal yourself, you can get more creative. Like recording the passage twice. Singing loudly and quietly will give your voice different timbres. So will varying the mic distance, where closer will be more intimate and suited to the quieter section. You could accentuate this by adding the slightest touch more reverb/delay to the louder section, maybe. All artistic license. With prerecorded or sampled vocals, you've got to look at if it's super noticeable that you've copy pasted the stuff and whether that's just cool or an annoyance. You can split the vocal up into words and shuffle the timings around a bit into an alternate phrasing that works equally well, giving more of an impression of a different take. If you're obsessive about it, that is. If the verse is louder because the backing track has extra elements in it, you could of course ride those levels as well, trying to make the summed level more similar, making large adjustments in vocal level unneccesary. There's certainly a good compromise there, where you push some things back a little as well as pull the vocal to the front. Anyway, it is a bit of an odd question, in that it really doesn't matter how you get there if it sounds right. If the simple thing of changing the level works, I can't imagine what would be smarter. Unless it doesn't sound right. In which case, try to describe what sounds wrong about it.
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Jul 23, 2016 10:41
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What I read in that description is sidechaining to make a compressor pump with the rhythm, maybe?
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Jul 24, 2016 19:51
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NonzeroCircle posted:Perhaps a judicious multiband compressor would let that through without squishing the poo poo out of everything else when it hits. What you want of the note, is to be as loud as it was without peaking so high. This can be approximated reasonably well by keeping up the average level of it, while reducing the peak level, aka compressing it (without makeup gain)! Probably even with a multiband compressor that passes through everything else! Pretty much de-essing, but at another frequency range. After that, the level of the whole mix can be brought up as normal, without unwanted ducking.
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Jul 31, 2016 15:06
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thehustler posted:I have a wireless mic set and the output from the receiver is a balanced 3.5mm minijack. thehustler posted:The PA system I want to plug into is a little church disco kinda all-in-one PA and it has some phono line-ins (obviously unbalanced). It helps if you post brand and model of gear you're talking about. But if you're going line level to line level, something made of cables and adapters should work.
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Aug 2, 2016 16:29
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thehustler posted:Manual for the wireless mic says the receiver can do line or mic level output, set by DIP switches, and explicitly says balanced. In this picture, look at number 10. It says to go from balanced trs to unbalanced rca, you just have to ignore the ring bit.  In the following picture you see that that's equivalent to using a stereo trs mini jack to rca cable, but leaving the right (red) channel unplugged.  You could use two of those cables or put the receiver in mix mode and use one cable. As long as it's all mixed down to mono at the end and the sound is only coming out of that single PA speaker, there's no difference. It doesn't look like there is a panning or balance control to worry about on the PA speaker anyway. In a stereo setup, you'd have to think about one mic coming out of the left speaker and the other one out of the right one. Or both left if only using one cable. That would be a pain in the rear end associated with using that aux input. I'm assuming you're doing this because the other mic inputs on the PA thing are already taken. In the first picture you'll see what's needed for using those at number 9. But that would probably mean more adapters anyway. On the other hand: no worries if this is expanded to a stereo setup through the aux out. Looks like using the aux in will also mean you have to set levels on the receiver end. You might also be bypassing any eq and echo options on the PA. Might not be a problem; just saying.
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Aug 3, 2016 12:23
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The recommendation for using a 3.5mm stereo jack to rca cable was made with the condition in mind that it wouldn't require soldering, because they are ubiquitous and cheap. And again, the portable PA speaker has mic inputs that will allow you to use the level and tone controls. If you're not plugging any other mics in there and you're having a cable made anyway, might as well use those inputs. Emphasizing that, lest your supervisor think what you're asking him to provide is odd and people on the internet crazy.
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Aug 4, 2016 20:56
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