This is something I always struggled with and over the years I’ve developed different coping mechanisms.
If at all possible my solution is to let a mix rest for several days, working on something else till then. But sometimes it isn’t, so:
1. Mix at really low levels for a while.
2. Take breaks. In silence if possible. If not, listen to other music.
3. Get a 20$ speaker like this and mix with it
4. If you’re mixing several different tracks, instead of going at them sequentially try to spend 20 minutes on A, then 20 minutes on B, then C, then back to A, B, etc.
5. Flip the stereo.
6. Walk around while the track is playing. Even go out of the room and listen from afar.
7. Minimize the screen and let it play entirely. Don’t touch it, but take notes or what’s popping out and sounding odd. If nothing is, then I t’s probably done.
8. If all else fails, you’re on the clock and have to deliver soon, pull up 2 or 3 reference mixes. Other tracks that are not only musically but also sonically close to what you’re going for. Level Match them to your mix, try to align their structure in time (e.g chorus hits at the same time, etc), and then quickly flip between your mix and each reference. Really quickly, like one second per track, and take notes of what your mix needs. This usually allows me to “disconnect” from my mix and only focus on global EQ, depth and dynamics.
9. Your ears are like a multiband compressor. If you’re mixing at loud volumes, they are compressing for you. It’s not linear over the frequency spectrum, and if pushed hard can give you permanent hearing damage. So take breaks often.
Basically just automating sends to delays and reverbs so the effect targets specific words/phrases.
I set up a basic example for you to check out.
Screen cap of delay and reverb throw
Audio Clyp of said throw
I'm gonna go out on a limb and say that Cakewalk is a lot better than waveform free. Cakewalk is literally Sonar platinum but free (no-cache no limiting factors no bullshit) Professionals still use it and prefer it over Ableton and another premium DAW's. You should add it to the list.
-Modern Recording Techniques (Audio Engineering Society Presents)
-Musicophilia: Tales of Music and the Brain
-Proust Was a Neuroscientist
-Writing Better Lyrics
-The Way of Zen
-The Runaway Species: How human creativity remakes the world
-The Power of Habit: Why We Do What We Do in Life and Business
From the FAQ:
> Does foobar2000 sound better than other players?
No. Most of “sound quality differences” people “hear” are placebo effect (at least with real music), as actual differences in produced sound data are below their noise floor (1 or 2 last bits in 16bit samples). foobar2000 has sound processing features such as software resampling or 24bit output on new high-end soundcards, but most of the other mainstream players are capable of doing the same by now.
Lots to go on.. but start with the Rod Gervais book
Go by this and you'll be OK. Walls with double drywall + green glue will do you right... take care with ANYTHING ELSE that goes through a wall. Doors, HVAC, electrical... that's the tricky part.
The Sound Reinforcement Handbook by Yamaha
Other great YouTube is Dave Rat
Buy this book and read it. When your done, read it again. When your finally done that, read it a third time.
Then join this forum;
And read all you can and ask your questions. You can also upload your design and the community there is very good at giving honest feedback.
Gearslutz.com is a great resource as well.
Studio building is 90% planning and 10% construction.
You really asked for a link after he gave you the retailer’s name, the product manufacturer, the model number and the price?? REALLY?
Here: Etymotic High-Fidelity Earplugs, ER20XS Standard Fit, 1 pair, Polybag Packaging
in case you're curious, the same designer who's now helping with Audacity's design previously made the designs for the upcoming Musescore 4, and prior to that most of Paint 3D and parts of Powerpoint and Ubuntu Mobile.
The Master Handbook of Acoustics is a solid choice.
You can apparently get the Fifth Edition for $10 used.
/u/CptHampton is exactly right.
If don't want to record hours of silence, and your phone is an Android, you can use an app like Smart Voice recorder which can pause recording on the silent parts and only starts recording when there is sound.
Indefinite, full-featured free trial, very reasonable purchase price. Not a huge range of included plugins but a clean, extensible workflow and the dynamic plugins are very very good quality. Side chaining and custom I/Os are simpler and more extensible than any other DAW, in my opinion.
I also add CM magazine/MusicRadar's free CM plugin suite. A bunch of good synths and plugins for the cost of a magazine.
Oh my god, my weird sound effects history is actually useful here. I have a Zoom H2 rigged up to small boom pole using a shock absorber with a good number of washers to steady a bolt onto it that will fit onto the bottom of the mic and keep it steady without too much noise when moving it.
I've totally recorded a few animals (yes, including my own cat) this way with success.
EDIT: Pictures of the setup sans boom-pole.. Boom pole in question is from this.
I made some audio examples for you:
Burnley 73 cycling from the lowest to the highest settings and back down
Decapitator at full drive cycling through each different mode a couple times
I'm the developer of BlackHole virtual audio driver. It's an open source project that allow users to pass audio between applications. It uses the GPL 3.0 license which means all other applications that use or bundle BlackHole also need to be open source and use the GPL 3.0 license. Since OPs app bundles a modified version of BlackHole it also needs to be open source under the GPL 3.0 license. So if it is indeed open source than that's awesome. If not, OP is infringing on the copyright for BlackHole and I would be extremely suspicious of using their app. OP hasn't responded to my request for the source code yet so tread carefully. It's not cool to steal. Hope that makes more sense.
Not sure which OS you are using but have you looked into Pure Data or Max/MSP these may be what you are looking for.
Pure Data is basically the open source version of Max/MSP
Reaper rocks for multi track audio; 'unlimited' free trial.
Audacity is clunky and awkward to use. I prefer Wavosaur for destructive audio editing; it's more similar to Cool Edit Pro / Audition
Reaper. You can get a free trial that is the full program for 30 days (though I'm almost positive that you can just keep using it if you don't want to spend the 40 dollars on it). It's gonna be different for you (as it would switching to any other DAW), but it can essentially do everything that Pro Tools can. I'd at least give it a try.
Look at Geoff Emerick’s Here, There, and Everywhere, My Life Recording The Music of The Beatles. Emerick was the engineer on many Beatles albums.
If you want the mic on the same table, you're basically not going to get any better noise isolation than a decent shock mount.
Fortunately, with a little ingenuity, a shock mount is not incredibly hard to DIY.
(In the image I used those cheap-ass rubber bands that turn to sticky mush in a matter of days - but you can do a lot better if you find elastic fabric strips for the same purpose.)
Option 2 is to suspend the mic from the ceiling, but that can still conduct sounds from upstairs and quite possibly from your floor (not a huge concern, but possible).
>I do not have the 200$ to spare for a first-party arm and shock mount
So get third-party. No shame in that.
Came here to post this, thankfully it's already the top comment. I ordered a set of these silicon rings off amazon for cheap. They are designed to slide onto mics and make it so they don't roll off tables, etc. I use them as something like this but for rappers. If I see a rapper cupping the mic during sound check, guess what, you earned yourself a dummy ring. It's proven to be a pretty good way to train rappers out of that shit. You kinda look silly having to use one of those, so maybe you'll learn for next time.
You might like to check out the "Introduction to Music Production" course offered FOR FREE from the Berklee College of Music (yes THE b.c.m.). I completed it and it was superb. All at an entry level.
It is offered via "Coursera" and is totally free. The teacher of this particular class is great and very competent. He is a Mohawk sporting actively gigging Bass player named Loudon Stearns with SEVERE chops. I suggest that if you're a starving musician like myself, you apply for a grant and get the Certificate.
You might even consider the "Modern Musician" specialization course. four classes all meant for the entry level musician. Here's a link to the "Introduction to Music Production" class.
"Introduction to Music Production"
These have been working great for me for many years: Etymotic High Fidelity Earplugs
Mike Senior's Mixing Secrets for the Small Studio is solid: https://www.amazon.com/Mixing-Secrets-Small-Studio-Presents/dp/1138556378/ref=dp_ob_title_bk
there's no reason to use audacity in your case.
also, there's no need to normalize, especially not as many times as you are. you aren't doing any bit-harm to your audio but it's just unnecessary. you just need to work on your gain staging of your signal thru your devices. i never normalize, literally ever.
you can run a vocal signal thru 8 different processors, send to a couple fx buses and to your master bus and you don't need to normalize, just get the input signals and output signals of each processor in good areas (input trims / faders vs. makeup gain vs. output trim / faders etc).
Former electrician here. Electrical grounding is very important for your safety. That is why the ground line to every outlet is highly regulated, and not optional. That being said, the grounding on any outlet in your home is connected to every other ground in your home. If there is some kind of ground fault, say for example a neutral-to-ground anywhere on that circuit, it can not only cause hum (sarcastic understatement /s)... it can cause a fire! Audio people tend to deal with "hum" and it leads to a phenomena called "audiophile" where bogus suppositions become regarded as audio truths, pretty much any unchallenged supposition eventually becomes a bias over time. That being said, it's not entirely untrue; Many audio cables are unbalanced circuits and generate their own noise. Induction, and RF interference are big contributors to this effect. Anyways, all I can say for your power outlets is to test them with a device mostly because I would not want you to damage your expensive equipment on faulty wiring. I would also highly recommend using a UPS between the wall and the equipment.
You will be amazed how little power that powered speaker needs for your application. That's because audio can require large amounts of power but only for very short periods of time. When the music is quiet, the power demands are quite small.
The key is that most music has a "peak factor" of at least 10. That means if your loudest peak power is 600 watts, the average power at the moment is 60 watts. Plus, when you're singing/playing anything but heavy rock, there are breaks in the music, usually every beat. And there will be soft passages. This means that about 10-20 watts average power would be enough for a typical performance.
Lucky, too, the K10 uses a class D amplifier which is amazingly efficient (around 90%). As an electrical engineer, my best guess is that one of these portable backup power units will run that speaker system for hours and hours.
Amazon has a good return policy if it doesn't work, but I'd be willing to bet on it. Me: Electrical engineer and long-time audio guy.
FWIW, your internet not being reliable and your internal network not being reliable are two different things, you might be able to get away with a basic VNC session to the desktop, since that doesn't use the internet and your internal network should be snappy enough for this.
Or get a CAT5 VGA/USB extender, like this (you'll need a second monitor, though) - https://smile.amazon.com/StarTech-SV565UTPU-Console-Extender-500-ft/dp/B00267TXZ8?sa-no-redirect=1
If you have an android phone, I have made an simple app for feedback and EQ training https://play.google.com/store/apps/details?id=com.saninnsalas.audiotraining.
I am also always open for feature requests (even though currently I have not much time available).
Edit : the first version of that was a website : https://www.saninnsalas.com/coding/audio-training/
Frequency dependent treatments generally are for situations where a particular room mode is very problematic. Broadband absorbers are most often the best price/performance ratio by far.
As for references, it’s true that few other books cover the topics involved well, and generally they’re very expensive. This work by Cox and D’Antonio is one of the best, albeit not inexpensive. Also, the work of Helmut Fuchs has fundamentally changed room acoustics almost from the ground up. I still hear consultants claim that the 1/4 wave rule is immutable while Fuch destroyed that quite a few years back. His VPN devices are amazing.
Books. Start with your local library system and find every book they have on the subject. Scan them all, and read those that seem to speak to you. Ask for book recommendations here. The one that comes up most often for live sound is "Sound Reinforcement Handbook" ( https://www.amazon.com/Sound-Reinforcement-Handbook-Gary-Davis/dp/0881889008/ref=sr_1_1?crid=32D1J9UME9UQA&keywords=sound+reinforcement+handbook+2nd+edition&qid=1564110323&s=gateway&sprefix=sound+reinfo%2Caps%2C194&sr=8-1 )
There are used copies available on Amazon for less. Even though it's from 1989 most of the information is still applicable.
I don't know if I'd call that "one of the biggst metal producers in the game right now", especially since he seems to only work in a very narrow subgenre.
Not to dismiss the guy, he's doing stuff, but let's not exaggerate too much!
Depends on the encoder/decoder.
Because of several reasons, audio that goes though an MP3 encode/decode process will be longer and out of sync. The delay/padding can be compensated when the encoder correctly sets the MP3_ACCURATE_LENGTH, ENC_PADDING, and ENC_DELAY tags, and the decoder uses them.
It looks like this in foobar2000.
I tested this with lame 3.99.5 command line, both as an encoder and decoder. The decoded WAV has the correct number of samples and sync.
The ffmpeg encoder has correct sync but is a bit longer. It seems the ffmpeg encoder puts the correct delay but sets the padding to zero, thus the decoder does not remove the extra samples at the end.
Some DAWs like Reaper use these tags when importing or exporting MP3s.
I love having clean ears and the idea of busting an eardrum gives me the willies.
Here's a link on amazon
On a recent project, a huge chunk of motion graphics landed on our desks, in dire need of some chirpy bleepy scifi-stuff. I made a bunch of assets similar to what I showed in this video but we felt it was all a bit cliché after a while, so I took a few days to experiement. What I eventually settled on was to run all kinds of broad-spectrum, high-transient sounds through a 30ms flanger with the modulation turned off. After some experimentation, I noticed that I could chuck just about anything in there -- including pure pops and crackles -- and end up with something that had a lot of body and character.
It's one of the best feelings in the world, opening up this huge space for experimentation, "feeding" a plugin all kinds of sounds and hearing what comes out on the other side.
Definitely a stand-out moment.
Check out this podcast episode with a forensic audio analyst. He goes into quite a bit of detail about things he looks for in a recording.
Not according to Michael Beinhorn, the producer. If I remember correctly, he states in an interview with Warren Huart that Chris tracked his own vocals for Black Hole Sun by himself. No one else there. Beinhorn claims it allowed him to get there emotionally ..whatever it takes
If anyone is interested, i highly recommend a small program called f.lux that removes blue light from your monitor.
It's a bit jarring to look at at first but you can set it so that it fades out gradually. After a few days i stopped noticing it and i just leave it running all the time now.
If i ever switch it off i'm temporarily blinded by the normal light of the monitor!
Not sure KRK is the best choice, but that's up to you. Monitors aside, for added studio functionality a good bet is to try one of these
Forget glue, it’s a PITA to remove and smells bad, if you’re only hanging 1” foam, go to a fabric store and get some t-pins. You only need 2-4 per panel and they leave barely any marks to fill when you remove.
You can buy a USB sound card with 8.1 output for like $5 on Amazon.
Maybe something like this?
Optimal Shop New USB 6 Channel External Sound Card 5.1 Surround Adapter Audio S/PDIF for Laptop https://www.amazon.com/dp/B00Q4WQ7XW/ref=cm_sw_r_cp_api_8r0RxbE442V2Z
Sorbothane pads. You can get an 8-pack for $20 from Amazon (https://www.amazon.com/gp/product/B00X6R1CTC/ref=ox_sc_sfl_title_3?ie=UTF8&psc=1&smid=A1EPE8IE7JPHY4), that will be enough to hold up 2 20 pound monitors. If you have monitors that weigh less or more, you'll need to get a different durometer number. The pads have to be loaded with about 50% of their weight tolerance to be effective. It is widely used in industrial settings to prevent transmission of vibrations from heavy machinery, it will be more than sufficient at preventing desk resonation. Using the right duro and thickness, it will completely stop pressure waves moving through it -
https://www.youtube.com/watch?v=FNwIc_RNOHs - just need to put some fabric between the sorbothane pucks and your desk/monitors as the polymers may cause discoloration on some finishes. If you need to boost your monitors up off the desk surface to get them to ear level, you could also throw in a couple of brick paving stones underneath, those are great for decoupling as well, not very aesthetic but at $1 each they're definitely DIY and budget friendly.
hope this link works. I like this chart
Actually it was the producer, Michael Beinhorn's decision to kick everyone out while recording Chris's vocals.
He determined that Chris produced the best vocal takes when noone else was around. It wasn't an ego thing or even Chris's personal choice. Let's make sure we dont misinform people.
The album I always go to is Voodo by D'angelo. Mixed by Russell Elevado. The drums and bass fit so infuriatingly well together while all the other instruments just float around my head exactly where they are meant to be. It makes me feel like I'm in the room.
Then my absolute favourite song is the Greatest Hits album version of Lovely Day by Bill Withers. https://clyp.it/jy3ia4yt
It is a full featured DAW that has excellent MIDI (much better MIDI than ProTools) and audio. It is used for popular music but it seems to be most popular with film scoring people for some reason. I’m pretty much a solo artist/producer and have only dabbled in writing to film a couple times but I was pretty amazed at the features.
Also, it has more than a couple academy award soundtracks under its belt. Danny Elfman is a big user for instance. The Hurt Locker was done with it. Head over, there are some pretty serious professionals over on that forum and they are pretty generous with their time and advice.
Download the demo and try it - what have you go to lose?
> and also the max rate on gen-1 scarlett's is 48khz
Wrong. The maximum sample rate on gen1 scarletts is 96khz, for gen2 it's 192khz. BTW, unless you have very specific needs, using 44.1kHz or 48kHz is absolutely fine - a higher sample rate just gets you bigger files and uses more processing power for virtually no audible benefit (unless you a are a bat...). Recording at 24bit makes sense though - you'll have more dynamic headroom to work with.
As for a beginner DAW, I'd recommend REAPER, since you can download fully featured trial version with only a nag screen after the trial period.
This sort of thing is quick and easy to set up using Max/MSP or Pd. Max will support rewire just fine if it needs to integrate into other things; I'm not sure about Pd, though I wouldn't be surprised if it did.
Measurements are presented as absorption coefficients, which are numbers between 0 and 1 (0 being perfectly reflective, 1 being perfectly absorptive) across the frequency range under test, which in this case is 300Hz-10kHz. These absorption coefficients can then be used in calculations/simulations of reverberation time.
There is an example of plotted measurement results for an office chair here: https://plot.ly/~mrlyule/108/absorption-coefficient-result-for-an-office-chair-using-microflown-impedance-gun/
Yeah, you have the wrong cables. You're feeding each monitor with the same stereo signal when they're expecting balanced mono.
You need to break out the tip and ring of a 3.5mm plug into mono 1/4" plugs. This kind of cable:
50lbs?! What are the dimensions of these panels? A typical 2’ x 4’ x 4” or 6” panel shouldn’t weigh anywhere near that.
In any event, I have 2 thoughts:
Disconnect the audio interface, restart your computer, reset SMC, reset NVRAM. Restart again, connect interface, see if the issue is resolved.
Contact Focusrite support if this does not solve the issue.
For everyone wanting to take a crack at this here's a link to the software I'm using. It's free if you register. It's easy and they don't bug you with spam.
In preferences there's a calibration feature that allows you to loop a test signal from an output back to an input (you use a cable, the software does its thing) and it generates the graphs you see.
Anyone who does this either post the graph in here or PM me and I'll include them in the imgur album.
Link to the gallery: https://imgur.com/a/f8yg7
Please tear this apart with your critique, but give me suggestions on how to improve whatever aspects you criticize. Thanks!
Also, does clyp do any data compression? I'm hearing artifacts in the upload that I don't hear in the file on my computer.
We use HDs with clones to backup all sessions. 4 TB each. To search the drives we use a utility called CD Finder. http://www.cdfinder.de/
Every drive is scanned into the database and totally searchable. ever since the 90's!
For an interesting reverb that could totally work, check out http://www.pspaudioware.com/plugins/reverbs/psp_pianoverb/ it's a free verb based on piano strings. Could be useful.
As far as wind chime sounds go, check these out:
The folks at Audacity have more information:
It's always best to download from the site of the developer. Here's the link for Audacity:
It's absolutely worth learning. While there's something to be said for comfort in a given DAW, the more you know the more versatile you are to any employer. Also DAWs have different strengths and weaknesses.
Audition is a pretty easy one to learn, and I quite like it for editing a specific waveform. (It's not my favorite for multi-track, but it has that functionality too.) Now is a great time to learn too with pluralsight.com offering free April. They have course content for audition.
Here are the measurement results for my chest, wearing a shirt and jumper. The results vary a fair bit depending on the position of the gun, but this is a decent example.
There is a massively long (perfect for our niche community) documentary about Brian Eno on Amazon. Free with commercials, covers his innovations in the 70s. So worth the time!!!
I know you didn’t ask for this and it’s not going to help your current situation but something like this could prove useful in dealing with heat buildup on your gear. I’m not sure if you could get this exact model but if you could, it’s great. Has 3 settings and is almost silent on its lowest. Basically, if you have gear that running warm, toss it under it, on it, or beside it and it will help keep the temps down.
I have 3 and use them all the time. Even keep on on my laptop keyboard since I use an external keyboard while I work.
I also have one of these for larger gear
If you can stretch to $159, the Beyerdynamic DT-770 Pro are excellent and well regarded by professionals.
I'm just going to jump to additional question number 3 and say that yes, the corner is generally a very bad spot to set up. Ideally you want everything along the same wall, with some distance between the speakers and the wall, not only because there will be constructive interference (especially in the low end) but also because they can overheat. If possible, put at least a few inches between them and the wall with some absorption pads/panels behind them.
Also, pads under your speakers are important so as to avoid some resonances on your desk. They don't have to be expensive either. Here are some on amazon.
Good luck with your new stu!
This is a good free entry level text: http://www.dspguide.com/pdfbook.htm
You'll want to have a decent grasp of maths to make the most of any DSP topic, though.
For designing and implementing audio plugins, you can either use the raw APIs/SKDs for whatever format you choose or choose a cross-platform, cross-format SDK like Juce (probably the most popular) or IPlug (not as popular).
C++ would be your go-to language choice. Something like "The C++ Primer" to get you started followed by "The C++ Programming Language" would be a good bet.
if thats all you really want to do then just drag the track into audacity and use volume/gain automation to bring down the loud parts and bring up the quiet parts.
here's a little video that shows you how to do that: http://www.ehow.com/video_4460333_draw-volume-automation-logic-pro.html
a compressor will do similar work, but it will affect the sound more. considering you are working with mp3s, you probably will only be able to use a compressor lightly before it adds noticeable distortion.
sound isolation vs sound absorbing
The easiest way is to make your own reverb is to put a speaker in a room and send your mix to it, then set up a mic on the other side of the room and record the mix playing through the speaker as a new track in your session. Be sure that the new track has the output muted, if it goes out to the speaker you might get some crazy feedback. Move the mic around to find the best sounding spot in the room. Move to different rooms for different sounds.
If you want to dig deeper you can always Build your own plate reverb
You should try it out with a Linux live USB! I'm using Ardour 3, which is what I started with. You probably used version 2, which has a laughable GUI. 3 looks really professional, and JACK has gotten really easy too. Also, something I really like and think should be more known: You can use Windows VSTs with their GUIs on Ardour (with Carla, a really nice host that supports LV2, LADSPA, DSSI, VST, VST3 and even SF2 soundfonts!) and you can also use Windows VSTs live with "vst-host", which is included in the dssi-vst package. I'm using the normal kernel, and i have a shitty Realtek sound card, but I got no latency at all with the guitar plugged into the mixer and the signal being sent to a Tape Delay VST.
Edit: By the way, you can get Ardour 3, Carla, dssi-vst as well as a BUNCH of LV2, LADSPA and DSSI plugins in the KXStudio repository. The commands for adding the KXStudio repository can be found here.
Worst thing that happened to me re: drives:
I had two drives fail on me at the same time. One physically, and one corrupted beyond recovery (and my day job is as a computer support tech - I've recovered some drives that even I was surprised were recoverable. TestDisk is pretty awesome sometimes).
Naturally, I had a third drive with most of my data on it.
...Everything except the final copy of a session I did with a band that wanted to re-mix something a while after the record was done. The only copy I had was not only missing final mixes, but some of the later overdubs. Ouch.
Then a few weeks later my house was robbed and my computers stolen, but none of my music or recording gear. But that's an entirely different story.
I'd tell him the truth, that it's a tedious and expensive process if he wants it done professionally by hand you would have to charge your hourly rate.
If not he can always try some normalization software, either audicity for free or dbpoweramp can do batch normalization/conversion/apply any VSTs.
Although it's called a converter there is nothing to stop you from having WAV in and WAV out and just using the normalization. It can also do stuff like trim leading/ending silence which might be useful too.
Or some kind of hybrid, tell him you'll teach him the process of using dbpoweramp for an hour or so at your hourly rate.
This is false. After 60 days you are legally required to buy a license, although Reaper does not enforce this via DRM (i.e. you can still use it after 60 days, but not legally..)
Either way, $60 really isn't much to pay for such an awesome piece of software.
It's GREAT for the price.
That said, try it for yourself here. There is a bit of a learning curve but it's not bad.
If you're just getting into using a DAW you'll probably want to spend some time with the manual and in the forums but that's the nature of inherently complex software.
TLDR; Try It
Depends more on your final distribution and less on sonic quality. Do you plan on releasing through Apple Music, Amazon Music HD/Ultra HD, Spotify Premium, Tidal or other high resolution channels? If so, you should master at 24bit.
Sonically, you will not be losing much by going with 44.1kHz/16bit, but it might keep you from being able to take advantage of some marketing and distribution opportunities.
Info on Apple Digital Masters
Amazon Music HD/Ultra HD FAQ
I’ve got a good one for you: “Backstage Passes and Backstabbing Bastards” by Al Kooper.
I don't think this is exactly what you are looking for - but I personally enjoyed the hell out of this.
You should be running your sessions off an external hard drive anyway. Nobody uses the internal. It literally says in the Pro Tools manual that you should never run sessions off the system drive. It's just common sense cpu optimization. Just get a T5 & run sessions off that, plug into any rig you want. You can even velcro it to the back of your laptop and you'll never notice it's there.
Phantom power is electrical power that is supplied by a mixer or an audio interface to the microphone.
Electret condenser mics like yours use a little preamp inside the body of the mic to boost their signal up to a usable level. That's what's using the phantom power in your mic.
Without that little preamp, in order to get a usable signal you are having to do the same amount of amplification later in the signal chain, were noise becomes a problem.
If you need an external phantom power supply, they can be found for about twenty bucks on Amazon.
For that plus the cost of an XLR cable, though, you might get better results with this:
It's called a "Pyle" for a reason, but since you are using a cheap $30 condenser I think your gains will outweigh your losses.
Edit: Also, I'm pretty sure that mic is side-address based on the way the capsule is oriented, so don't go talking into the top!
I liked J4T Multitrack Recorder. Unfortunately, it doesn't have a piano roll or drum loops, and it only has four tracks. I think you can bounce and import to one track when you need more, though. I didn't really go beyond recording some voice overdubs, but I know it has basic effects, looping, and editing.
This is the exact predicament that convinced me to switch to the iPhone. It's hard to compete with apps like Notion, GarageBand, Music Memos, and the Positive Grid stuff.
So i don't think this is related to your tests but it seems like you would know... is lowering the volume in windows similar to lowering bit depth as this highly upvoted post would have you believe?
First step would be to find out what the source is. Are you using Flux software https://justgetflux.com/? It's been reported that it causes clicks and pops. Setting it on 'safe mode' should stop causing issues. As for declicking, try iZotope RX declicker.
I got one, great buy.
EDIT: Nevermind, didn't notice the laptop monitor. Thought it was a mixer.
This works when moving properly recorded files between sessions at different sample rates. Tracking is different. The DAW is expecting a 44.1 signal but is getting 48k. It doesnt just squeeze it to make it fit.
The result is not a shorter, higher pitched version of the source that can just be converted back to 48k. It is a distorted, real-time version of the source. It sounds horrible.
I'll record an example at work tomorrow.
EDIT: heres the example of a sample rate mismatch i promised
Dude. I've been experimenting with distortion on my master channels and holy shit. It's that sound that i've been missing for so long. I thought it was just because my mix was bad or something. I've for so long been trying to get better at my mixing to finally get that "professional sound" I tried this on 3 other songs i've completed. the results are night and day. Just figured i'd tell you. you're the man man.
Especially now that I've applied it to my existing mastering chain.
My chain now goes:
I use Saturn of course for the saturation, ozone for the multiband compression, and Pro-L to push the gain a bit. (A/b ozones maximizer with Pro-L. Pro-L is MUCH better at maximizing)
Anyway. Thanks so much for the tip man!
I applied the satuation to a song I'm working on, check out this preview :) https://clyp.it/4zijcz2u
Nigel Godrich (Radiohead, Beck, Travis, etc.) is the king of using huge amounts of reverb in really beautiful ways. He doesn't use it all the time but when he does it's masterful.
Great thread, mesaone!
Here are my unique tips, and both have to do with omni microphones.
Omni close up on a low wattage tube amp (vintage if you have it.) Seems very simple, but is often overlooked in the recording books if you ask me. What a cool tone you get!
https://soundcloud.com/blackcircledj/niagra-balls (rhythm guitar uses above technique.)
I love using empty rooms as echo chambers. It is even better when in that room you can have a piano. Place the omni inside the piano, turn the sound source volume up coming out of the speaker, hit record, and let that microphone pick up some crazy sounding, REAL (no plug-in required) reverb. Quite a treat of a sound when done right. https://soundcloud.com/blackcircledj/andrew-neal-acoustic (No piano, but an EV635a omni is used to record this natural reverb from a spare, wood floor bedroom using an echo chamber type method.)
Enjoy all the great tips you are getting! Maybe use them all on one recording and let us know how that one turns out. Groovy.
I have a pair of his "Ultimate SDC" mics, and I love them. They can be a bit thunky on acoustic guitars when I just throw them up there for a live gig, but you can fix it with placement. It looks like he's using 991's instead of 603's now, but they're the same ~~circuit~~ everything.
They're used here for uke, bell kit, and drum overheads.
found myself really interested in this dude. he had a linkdin profile in his youtube page. check out this guys resume. he's not fucking around! https://www.linkedin.com/pub/ben-krasnow/4/6a9/679
It will play 24bit, but the iphone 8 converters+output stage don't even have enough dynamic range for true 16bit audio. I don't think any modern phones have the required noise floor.
Measurements from here:
Hi Biddly, so just running the numbers there, with a TMI of 2500pounds, divided by 150 pounds a day pay, you would need to book 16 days a month to record.
This does not take into account the cost of rent for the studio, or the cost of the monthly loan repayments.
I'm 100% about following your passion- that's what I've done and I couldn't be happier. But it's essential to maximize your profit potential while minimizing your financial risk to the greatest extent possible. Anything that requires a bank loan to startup is a serious financial risk in my opinion, so it's essential to ensure that the venture would be high profit and sustainable.
Personally, I would launch a bags/clothing/accessory business tailored to musicians and those who like live music, and then use the funds from that successful business to pursue my passion of a self funded studio for aspiring artists.
For example, lets say you designed a super slick padded guitar case that has additional pockets for headphones, wires, cigarretes and dope (or whatever musicians carry.) On the other side is a compartment for holding clothes - turning it into a musical suitcase for being on the road. You could have that made by Alibaba for as little as $3.5/2pounds. Here's one of many manufacturers:
Snap some very nice photos with bands using them, create a great story behind the design, why it's superior, with nice technical specs and a few reviews from band members, and start selling them for 30-50 pounds each. Then contact a couple of blogs who cover music, design, of bags, and boom, you're selling a bunch online. Use the money to launch new products and designs, and you've got a brand generating good, sustainable income.
Anyway, that's just one example of how you can remain in the music industry, while focussing on lower cost, higher profit options.
Any track that uses compression is doing it to some extent, whether or not it's the mixer's intent.
Check out Chris Lord-Alge's work. He's a master at using tons of compression on lots of tracks and making it work really, really well.s
You can try it yourself too - put a compressor on a snare drum with, let's say, a 4:1 ratio and a 50ms release. Pull the threshold down to get some compression happening (more compression will make the effect more apparent). Set the compressor to the fastest possible attack and listen to the snare hits, then gradually slow down the attack time - you'll hear more of the snare's attack poking through the mix as the attack time gets to 10. then 15, then 20, then 30ms and longer. In the context of an entire mix, the slower-attack snare will feel more "in your face" and present, and the faster-attack snare will feel more pushed back and distant.
Have them do this test while watching over their shoulder:
I did that last night using my studio headphones and I did no better than random chance. So much for Golden Ears. :D
This feature is built into Logic. In Pro Tools or other DAWs there are third party apps like MidiKeys.
SuperDuper is hands down the best cloning program out there. It makes BOOTABLE backups so if you really have HDDs fail, you aren't up a creek.
Raid systems are a good choice. The studio I work at has an 8 terabyte glyph drive as our main backup. It's technically 4x 2TB. We have it set up that 4tb is backup 1, and 4tb is backup 2.
The plus side to having a clone drive: you can take it to another Mac at a different studio, boot right up from it, and its like you never left your own studios computer.
Ditch time machine. It's slow, cumbersome, and is a processor hog.
Audacity is a great, free editor for clips that already exist. If you need to generate sounds, Bfxr is a great choice. It will go well with a chiptune aesthetic. You can design sounds and then write them to disk as audio files.
Thanks for the reply.
Here is a sample of the hiss.
I rent the house that we record in and I wouldn't be surprised if the outlet isn't grounded as it is an older house. If this is the case, is there a solution besides replacing the outlet?
I am using shielded xlr cables and Dynamic (Behringer) mics. The sound persists even if I don't have any of the mics or cables connected.
The mixer has its own power source. I have tried it on the same outlet and and on a different outlet (though this may not make a difference as I imagine it is on the same circuit.
You could use any number of free AI audio processors to separate the vocals from the rest and then manually adjust the vocal level in your DAW with the 2 new stems.
or install it yourself. Search for Spleeter
If you have Ableton Live 10+ you can install a real time plugin
Have you looked at something like Audio Hijack by Rogue Amoeba? It lets you set up almost any routing you can think of, probably a good way to have everything going where you want, whilst avoiding audio going to places you don’t want and causing feedback and echoes.
I would like some money if it’s going plz
You should be asking in http://www.reddit.com/r/AudioPost. The sound-for-picture guys are the ones who use SFX all the time, and there might be a technical theatre sub that is appropriate too.
The first place I look for sounds is freesound: