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Are you looking for a standalone solution, or some sort of plugin? Is it about loading up some wavs, or is the step sequencer the essence? Anyway, I just had a flashback to the late nineties, where Rubberduck DrumStation was a shitload of fun and it's free now, so that's somewhere to start anyway  Not as versatile as FL by any stretch of the imagination, of course.
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Mar 31, 2016 14:53
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If you're already using a DAW, I highly recommend finding something that integrates with that. The VST thread can probably help you with that, but most DAWs have a plugin that comes with the product for triggering drum samples at least. Always worth looking into, especially if you're rethinking your process anyway. Drum pads made for playing with sticks don't come cheap, but if you're ok with finger tapping on small squares, the there's cheap stuff from Akai, Novation or the Korg Nanopad.
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Mar 31, 2016 19:26
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CaptainViolence posted:I have an e-drum kit and a keyboard, both of which have MIDI cables strung over to my audio interface. Neither of them has MIDI Thru and the interface only has one MIDI input, so right now if I want to switch between them I have to unplug one and plug in the other. What do I need to be able to switch back and forth between them without plugging/unplugging things? Searching for a MIDI switch just gives me a bunch of MIDI control pedals that cost over a hundred bucks and don't look like they even do what I want, but I can't think of what else to call this theoretical box that I can't imagine costs more than ten or twenty bucks. I found it googling for "midi kvm" which could be another useful term for searching. Alternatively, for slightly over twice that amount, you can get a midi merger like the Miditech 4merge USB, which lets you use your kit and keyboard simultaneously, which is the way I would go. Sorry for the euro links.
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May 3, 2016 22:18
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SD cards work pretty much just like usb sticks. Many laptops have an SD card slot and for desktops there are usb card readers. You plug them in, put the card in the slot and the stuff appears in explorer or finder (or whatever that's called on Mac) as a new drive letter. From there you have access to the files and folders on the card, just like you would on an internal drive, or a cd-rom or a usb stick.
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Jun 4, 2016 09:49
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Perhaps the shortest shortcut out of this madness is if you give us brand and model of the mic.
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Jul 13, 2016 18:15
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Right. The black box it comes with has a jack labeled audio out. Do you have anything plugged into that? And if you do, is that what goes to the red mic jack at the back of your laptop?
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Jul 13, 2016 19:15
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Yeah, that's not how it is supposed to work. Either plug the microphone straight into the mic jack of your laptop, or plug the mic into the pink jack of the black box and then plug in the usb cable and nothing else. The black box will then function as the analog to digital converter instead of your computer's onboard soundchip, which is the point of the usb part of a usb mic. If you do the latter, you'll probably have better quality sound, but maybe you'll have to go into the audio settings of whatever software you're using to change the input device or change the default input device in Windows' audio settings (unless you already did that). The double connection is likely what's creating your ground loop. It also has the potential to create an audio feedback loop. There is no reason to have it anyway. The box is made to get your audio into your computer over the usb cable. Without the manual in English I can't tell, but the audio output on the box is either for direct (headphone) monitoring what's going into the mic, or possibly presents itself to the system as an alternative ouput device. It's not made to connect to a microphone input either way.
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Jul 13, 2016 20:32
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Officially, the opportunity to update from 7 to 10 for free ended on july 29th, if that's what you were thinking of doing.
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Aug 2, 2016 19:06
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well why not posted:What's the best way to ensure all my tracks come out at a similar level / loudness? Normally I make sure nothing exceeds 0 when doing a render, but they still seem to be really different sounding. To give tracks a similar loudness (to each other as well as to chosen reference material) is traditionally the job of a mastering engineer, because it requires listening skills, good (quasi objective) judgement and -commonly- high quality compressors/limiters. It's pretty much a job and an art in and of itself. While the digital mixer in your DAW will most often still use peak metering, some have loudness metering built in on the master bus these days. Loudness metering is also available as a plugin. There's the older K-System and a competing EBU R128 standard that expresses itself in LUFS (loudness units) instead of dB, with varying time windows to average over. Both are attempts to both give you a visual impression of how loud something actually is, but also are an attempt to standardize loudness levels throughout the industry (though conservative recommended peak-to-loudness ratios). These things take into account the sensitivity of the human ear to varying frequencies at varying levels (Fletcher-Munson curve). If you read up on these things, a lot of discussion will concern the standardizing part of that and you might not care that much about it. But you can still use one of those plugins to eyeball loudness levels and do quick adjustments to your mixer levels and master bus compression to get things into the same ballpark. Still listening to what sounds good, obviously! This is all pretty involved and alternatively, a program like iTunes has its own volume leveling and can bring all your music to a similar loudness on playback, though it can only do that by bringing down the level of the loudest tracks; it can't bring up the level of a track that would then clip peaks. Lots of media player programs can do this and some can embed the necessary information in mp3 tags so the file will play back at the same level anywhere, but iTunes is the only one I know that uses the EBU R128 standard. Lots of players use more primitive averaging algorithms. It's a limited solution to the problem anyway, so it depends on what you're actually looking for. Personally, halfway through the mixing, I'll start using a K-meter plugin (because that was the easiest to find for free) to see, mainly, if I'm overdoing all the compression. I'll stick to the K-14 standard in the loudest parts of the music (roughly equivalent to a 14dB peak-to-loudness ratio), which has plenty of dynamic range compared to how electronic music usually is mastered still, while never ending up accidentally being obtrusively quiet. This gives me a fairly consistent starting point to squeeze out a little more in the self styled 'mastering stage' by ear. So basically, the answer is compression, but the metering plugins can help you get more consistent results.
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Sep 16, 2016 12:40
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Knifegrab posted:I do mostly vocal stuff, as such I just use stupid and bad things like audacity. It works for everything I need. However I was trying to do something the other day and could not get it to work.
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Sep 27, 2016 18:25
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That would depend on the pan law setting of your DAW. e: Wait, no, it shouldn't. Interesting, often overlooked tidbit on panning though. e2: compared to "just the one mono track panned to the center", yes, pan law matters, sorry 
Flipperwaldt fucked around with this message at 21:13 on Jan 4, 2017 |
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Jan 4, 2017 21:09
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To be clear, this is (twice) in addition to the level increase you get from doubling up on the sound sources, I suppose. If I've properly grasped this conceptually. My brain is not made for math.
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Jan 4, 2017 23:12
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If you want to mix two sources, a small mixer might be the thing you're looking for.
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Jan 8, 2017 00:01
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Immolat1on posted:Ah, thanks! And dumb question: with that one I'm assuming the headphones plug in where it says "PHONES" and then the amp and phone cords would plug in where it says "2-track"? Would both go in where it says input underneath? Does the Left vs Right matter? I would want the headphones to play both inputs through both headphones (not one in one ear and one in the other).
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Jan 8, 2017 23:44
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Immolat1on posted:Do I need to buy an adapter that splits each input into a left and right for it to work? Stereo output from the guitar amp is mostly important when it has built in effects. Otherwise, you could run a mono-jack-to-mono-jack (aka "guitar lead") cable from the headphone output of the amp to the left input of the mixer channel and it should still end up in both ears of the headphones. For the phone, the adapters I posted earlier with a stereo-mini-jack-to-dual-rca cable is going to be the easiest.
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Jan 9, 2017 15:43
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Is the adapter to convert to 1/4" of a tip-ring-sleeve configuration? Then you're being hosed by balanced vs unbalanced connections. Use an adapter that is just tip-sleeve instead.
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Jan 13, 2017 01:09
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There were a couple of toy-like Casios (SK-1, SK-5) and Yamahas (VSS-30) in the eighties and nineties, yes. And the show may have referenced those. More current (still out of production though) is the Korg microSampler. If you don't mind the intermediary step of recording sound to an iPhone first, the Korg Volca Sample should be cheap enough to just gently caress around with. Even cheaper is loving around with iPhone/iPad/Android apps, like Caustic or whatever. Top of the line I'd name the Teenage Engineering OP-1.
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Jan 28, 2017 17:43
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I'm sure you'll find opinions on that in the synth thread.
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Feb 6, 2017 21:19
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FeastForCows posted:Is there any thread where I can ask for microphone recommendations (maybe this one?)? The general overview in the sticky was not really helpful for my use situation.
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Mar 8, 2017 13:30
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I'd recommend asking in the Home Recording thread then anyway, but the AT2020 is definitely a good choice for what you want to do with it considering budget.
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Mar 8, 2017 14:35
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Rupert Buttermilk posted:http://www.wikihow.com/Rip-DVD-Audio-to-MP3-Using-VLC-Media-Player You can just use VLC to rip the audio straight from the disc to wav. e: Like this if you need guidance. Flipperwaldt fucked around with this message at 19:31 on Mar 17, 2017 |
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Mar 17, 2017 19:26
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Rupert Buttermilk posted:Ah, I actually wasn't aware that VLC did that directly! Neat! And I was just referring to taking the audio, bringing it down to mp3 quality, and then burning it onto a cd as a track anyway. Why not just use .wav instead, you know? I'm just mad at the moron who wrote that how-to, mind you. The disc tab is even in the screenshot of VLC!
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Mar 17, 2017 20:12
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Rupert Buttermilk posted:If you all had to make a seamless loop of some background ambiance from a sound effect (raw audio file, not midi), what would be your preferred method? Edit: Like, is this what you want to happen? https://www.youtube.com/watch?v=yjGDW4wVkEQ How to get that script Flipperwaldt fucked around with this message at 17:06 on Mar 21, 2017 |
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Mar 21, 2017 16:57
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Sorry about the ambiguous formatting, but there are different links behind either word in "two leads", if you didn't notice. The first one points to some native reaper functionality, which is not all that convenient, but should work, I'd expect.
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Mar 21, 2017 18:09
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To do brass right you need to control a handful of performance parameters simultaneously. There are a lot of idiosyncrasies that don't easily translate from a keyboard, like bending up towards a note and tonguing, muting, falsetto and stuff like that. The quality of the timbre itself isn't all that critical, relatively. It's easier to implement in synthesized timbres than with sample libraries, that would have to have sample variations for all sorts of combinations and would have to be scripted to hell and back for performers with more than ten fingers. On a budget, it's maybe more sensible to get some older module that has accommodations for a breath controller (ie has midi parameters supported beyond pitch bend and actually does something with them) than to go with whatever has the most GB of samples. Although really clever scripting does exist in software these days. And it would require some specific training to get the most out of such a module if not used with an actual breath controller. Here are some brass programming tips from Rob Young's book The Midi Files that I posted earlier in the context of a similar question in another thread:  I don't know if it will help what you're doing but it gives an idea of the sort of considerations that make a brass patch sound more like brass. In short, it's a hard thing to do right.
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Apr 16, 2017 20:44
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The Science Goy posted:My band recorded a short album and ordered a small run of CDs for physical distribution (custom printed CDs, decent disc jacket, etc). To save money, we ordered the CDs blank and are burning the songs to disc ourselves. So far we have burned as mp3 CDs, but some people have reported issues playing them back in some cars, etc.
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May 5, 2017 21:41
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"Sort of" is hedging your bets too much. It's exactly the same thing. The reason I asked the question is to find out whether the other poster was using redbook already, but maybe using the wrong terminology. In which case this would be a different type of problem.
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May 5, 2017 23:07
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Rupert Buttermilk posted:I thought you were just referring to a data disc with mp3s on it, and not an actually-formatted MP3-CD Here are the highlights from the wikipedia page: quote:A compressed audio optical disc, MP3 CD, or MP3 CD-ROM or MP3 DVD is an optical disc (usually a CD-R, CD-RW, DVD-R or DVD-RW) that contains digital audio in the MP3 file format. Discs are written in the "Yellow Book" standard data format (used for CD-ROMs and DVD-ROMs)
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May 5, 2017 23:28
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Drink-Mix Man posted:I have some tracks that are 50 cents flat and I'd like a quick and dirty way to tune them up in Cubase. I have a pitchshifter vst but it colors the sound in an undesirable way. I was thinking it would be nice to be able to tune things in my DAW the way you can with an old school tape cassette recorder, by speeding them up or down slightly. Is there a VST that will let me do this with accuracy, or a way to bounce the tracks mess with the sample rate for precision tuning? You'll have to do the calculations of how many % you'll need to adjust manually though. There's a guy explaining how here, but maths break my brain. The pitch shifter in Audition suggests around +2.93% for 50 cents upwards, if I'm reading this right. Chances are I'm not, since its scale references play duration instead of speed offset. But it should be in the ballpark. If you have a tuner app like pitchlab on your phone listening to the preview, it'll probably help. A VST can't do it, because it can't feed back the change in play duration to the DAW, it can only process what it's fed in real time. I didn't really find a free solution that was more elegant and convenient, which surprised me. There are a lot of online calculators that calculate everything but this (+/-pitch adjustment in cents to +/-sample rate or +/-%).
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May 22, 2017 11:59
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You make a website, get 10000 followers on youtube, facebook and twitter and soundcloud, constantly promote yourself, be charming, funny and interesting, network with people at gigs to hell and back, particularly those organizing them. You make free remixes or covers and post them to the original artists on twitter sparingly, hoping to piggyback on their retweets. Maybe hassle popular youtubers to use your music in the background of their videos. Do goddamn press releases. All without coming across pushy or desperate. Basically take on a second and third unpaid, fulltime job managing your presence like a business with your music as a loss leader and maybe you get noticed by someone. All assuming your music is any good to begin with. You perform in front of a public any time you can, even if it costs you money to get there and hopefully after a while there's maybe ten-fifteen faces you start to recognise. These are the people who will repost your new releases to their friends. From that point on it's a lottery. With lottery-like odds. You need ambition like someone wanting to become an astronaut and you're equally as likely to be forced to settle along the way, because you're just not good enough and more importantly, not lucky enough and don't know the right people. If none of this comes naturally to you, learn to enjoy making music for the sake of it would be my advice.
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Sep 8, 2017 00:36
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MrSargent posted:Just curious as to how you are measuring the mix level of a professional track? The standard is imaginatively called ITU-R BS.1770-3 and was introduced by the European Broadcast Union. It has seen some adoption in recent years in radio, tv and streaming. The general idea is that by playing everything at a moderate "average" loudness that leaves room for peaks, they are taking away the incentive to squeeze the crap out of your mix for maximal loudness. All that happens if you still do that is that your music will be played back at the same level as music that doesn't, but it will lack impact because you're not using the available dynamic range. Well, the main purpose is to prevent the need of the listener to constantly adjust the level on their playback device, but it's nice that it stops the perceived need to push everything into the red constantly for producers as well. Compress and limit for aesthetic value, to change the balance within the mix, instead of as a louderer-mechanism. Mix with relative levels in mind, instead of absolute. Mixing towards the standards CaptainViolence mentioned is sane, as is mastering that mix fit for purpose later. CD players don't have volume leveling built in, so if your stuff is going to end up on a compilation CD, you're going to have to participate in the loudness wars still, to some degree at least. If you're going to distribute your stuff as files on some platform that hasn't adopted volume leveling, by all means, take a look at what everyone else is doing. You want vinyl releases or archival quality mp3s, it'll probably pay to be a bit more conservative. -14 or -16 LUFS is probably a lot more dynamic than most of us are used to, but it's a workable compromise, even for electronic music. It won't stand out excessively in non-volume leveled contexts except against the most extreme examples of late 90s, early 00s crunch. If you make folk or acoustic stuff, -23 LUFS is probably more appropriate. You are by no means forced to oblige though and it's still too soon to call all this effectively an industry-wide standard yet. That you can see articles contradict each other reflects that. Some may even be from before the invention of the standard. Others aren't aware of it or sold on it. There is still some idealism required to adopt it. With every additional streaming/mp3 selling service implementing it, the point becomes more moot though. But -6.5 LUFS is just about enough headroom to prevent your music from collapsing in on itself and it will certainly not raise eyebrows anywhere. Is it best practise? That's up for discussion and depends on context, I guess. well why not posted:YouLean or Insight on a track in Live. Might not be the correct way to do it, now you mention it.
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Sep 9, 2017 20:18
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MrSargent posted:If you drag a track into Ableton though, you are dragging in a mastered track so wouldn't the levels you measure only be relevant for comparing to another master? On the other hand, you get people doing their own mixing and mastering, often not really as two distinct stages. The language gets mixed up. I'm pretty sure the question was about what sort of levels are appropriate for distribution, conceptually implying mastering as well. It's not some major gotcha, in the sense that the ideal mastering process is just about getting everything in line with what's appropriate for the distribution media and if that means nothing needs to be done to the mix, that's platonically perfect. In the past this was all just theory, because you were in a competition for a loud enough mix and the mastering engineer had the superior, super expensive compressors, limiters and monitoring. You'd undershoot the levels and the mastering engineer would bring them up as transparently as possible. This was always somewhat artificial as a goal and the mastering engineer would always aim for doing the least damage. With today's technology and goals, provided you have quality monitoring, what you (as mixing engineer) are shooting for and what the mastering engineer tries to achieve, can be closer to each other than ever. Level-wise and for all platforms that have adopted volume leveling, at least. Instead about levels, the question is about what use do I make of the available dynamic range, which becomes an artistic decision, which is a matter at least as much in the hands of the mixing engineer/producer as the mastering engineer, if not more. So it used to be that you got warned against comparing your material against mastered tracks, in essence because you couldn't get there yourself at home and should be shooting for something else and thus comparing it to something anything but mastered material. However, if today you properly volume level your mastered reference material against your own unmastered music, you're not comparing loudness anymore, you're not fighting the technological limitations anymore, you're not mixing towards something that must be different from the end result and you can just make the mix that sounds best to you. The reference levels in LUFS become flexible sanity checks instead of the hard limit of the 0dB line. There are still jobs for mastering engineers to compensate for your lovely home monitoring, obviously. And again with the caveats of adoption rates of volume leveling according to those standards and what not. But it has its place in a discussion about best practise, imo.
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Sep 10, 2017 15:28
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All this is true for analogue and perhaps early standalone digital mixers. The mixer inside every current DAW uses floating point math and you can only cause the master to actually clip. The individual channels can go into the red on the meters, but won't ever clip. If you've got no processors on the master bus, lowering the master fader in that situation is identical to lowering all the other channel's faders. Mathematically. You can however make plugins used on individual channels distort if they are made to do so and/or don't use floating point math internally and a traditional approach to gain staging comes back onto the picture as best practice. A mastering engineer obviously can't know what kind of lovely setup you're using and telling people not to touch the master fader is safe advice for all situations. Also, the point might not be to get the levels lower in absolute numbers, but to get you to back off on that maximizer/limiter on the master bus that you're pushing way too hard. In which case lowering all faders will produce that result as a side effect, where lowering the master fader won't.
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Oct 26, 2017 18:09
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BIG TIT LIL NIP posted:I'm trying to control my korg monologues sequencer with the trigger out from a tr-08. When the tr-08s trigger out is plugged into the monologues sync in one trigger will advance the sequence multiple steps. Is there a way to get one trigger to equal one step with this set up? But, my man, have you tried our excellent synthesizer thread?
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Jan 29, 2018 13:25
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Some people have also gone full Stockholm syndrome for the crunchy sound of digital distortion. I absolutely hate it. It makes the instrument pull through the mix, so obviously put it on everything!
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Jun 10, 2018 22:36
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Reaper's Dynamic Split function can probably be teased into auto trimming silence, probably in batch, kinda, I'm not entirely sure. Anyway, Reaper is available on macOS and free to try.
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Oct 22, 2018 15:06
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https://www.audiofront.net/MIDIExpression.php http://www.tecontrol.se/products/usb-midi-pedal-controller These seem compact and priced reasonably.
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Jan 27, 2019 12:43
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It's a 'quirk' of the Scarlett's preamps that the gain range doesn't quite cover low efficiency (ie most dynamic) microphones. But if turning up the gain in software gives you the signal strength and quality you need, that's ok. Otherwise you're looking at a cheap efficient condenser microphone, a cloudlifter or (a different interface with) a preamp with a wider range.
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Jun 10, 2019 16:00
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Either another preamp or the cloudlifter would do. You wouldn't need both.
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Jun 10, 2019 20:38
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Oh, and as far as just bumping up the level in software goes, I recommend getting intimate with the workings of a dynamic expander plugin to keep noise levels sane.
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Jun 10, 2019 22:39
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