Podcast #128: We chat with Kent C Dodds about why he loves React and discuss what life was like in the dark days before Git. Listen now.

Hot answers tagged

11

I guess you'll be able to reach the desired result by applying these effects: EQ - Boost around 2KHz and cut everywhere else. (You can play with the frequencies and see which one suits you the best. To do that just boost one frequency and swipe it across the area). Distortion - After setting the EQ, you could add just a bit of distortion to the signal. P.S....


8

A lot's been said about compression but nothing's been said about intelligibility. If you're producing VOs, trails, bumpers, idents or jingles with vocals over a bed, EQ is also your friend here. If you critically listen to professional productions, they often lowpass AND highpass music under speech in addition to heavily compressing and EQing the vocals. ...


7

If you have a lisp, then it's a natural part of your speaking voice and you won't be able to get rid of it entirely through processing (and you shouldn't, in my opinion). That said, the "ess" and "shh" sounds exhibit quite a lot of sibilance, which tends to show up in recordings, like you said, as a burst of volume and noise. It can be very distracting to ...


7

Just add a compressor on your voice-over. Set the ratio to 2:1. Set a middle long release and a bit faster attack. (Or auto if you have it.) Then you lower the threshold until it gets better. Just play with the ratio and threshold until you get it right. What the compressor does is that if the audio gets over the threshold in volume it will wait the number ...


7

To make feedback a good way is to use the sends. Put on your source, for example just the default Operator, on your track. Send 100 % to Send A. Right click on the Send A wheel on the Return A channel and choose "Enable". If you increase the Send A now you will of course get feedback, but it will come instantaneously and will not output any sound. I don't ...


6

As the other comment said. Try a low-pass parametric filter. Adjust the frequency down to about 300 Hz, maybe lower, and see how you go.


6

Notch filters are perfectly valid in certain key cases to deal with a particular problem with the sound. For example, when trying to deal with a single frequency buzz (like a 60hz line buzz) it is ideal to cut a very narrow band to deal with the problem. In general however, this is only true when troubleshooting a problem with the recording. For a general ...


6

It really depends on you. If your DAW (Ableton) does everything you want, then you're good to go. VSTs come in when you want to do more. You may decide the reverb isn't quite what you want, or you'd want a vintage synth, or orchestra, or electric piano sound, or better drums beyond what comes in the package... that's the time to look at VSTs. That's not ...


5

For post-production, I'm not aware of any hardware device that can do what the software equivalent can't, with the exception of noise reduction hardware, such as CEDAR. That, and perhaps 5.1 processors, like the TCE Reverb 4000. But after speaking with Doug Jackson a couple weeks ago about this, he showed that the TCE reverbs on the dub stages he works on ...


5

As well as the technical means mentioned in other answers, you should also consider "working the mic" - understand how various syllables produce air out of your mouth. Position yourself slightly off-axis from the mic so the blasts of high frequency sound and high speed exhalations don't hit the mic's diaphragm head-on. 30 or 45 degrees off-axis can produce ...


5

I agree with what's been said, although don't forget that part of what gives a sound its off-axis/down-the-hall timbre is how it resonates through the building materials. So yes, highs will drop off but you'll likely need to bump sonewhere between 180 - 400 Hz where there's a nice resonant quality, just be careful of the 300 Hz muddiness. This is where a ...


5

Here's my process for creating loopable sound files: Select a reasonable amount of audio from the tail end of the file ("reasonable" is variable and entirely dependent on the content of what you're working with), cut it off and paste it onto the front of the file. You'll need to make a crossfade from the tail end of the region you just cut onto the head of ...


5

I wouldn't record using FX unless I was very sure about the end result I wanted and how it would sit in the mix, just to leave my options open since you can't go back and remove the FX. On the other hand, if the FX are an integral part of the performance then it might be a good idea to record after the FX. Sometimes the performer might hit a 'sweet spot'that'...


4

For Foley Usually LDCs (large diaphragm condensors) since of prime importance, they retain a lower noise floor. U87s are often used for this, sometimes with an MKH 416 or KM81/82 as a room mic. Octava makes a good LDC I've used, although I cannot recall the model number - it's cylinder-looking one. AT2035 would more in the bargain range. For FX, it can ...


4

I've had great luck using a home made impulse response created by slamming a 2x4 plank on the floor upstairs with a hammer hard, while recording it downstairs. Try taking the top of your sound off with a filter and send it through an IR like that.


4

Creepy is my middle name. Here are some suggestions: Regarding noise, if you have any noise on the original tracks, it will be difficult to remove it and this gets worse if you add multiple tracks with noise. Best to have the best recordings you can get your hands on before processing. Also learn to adjust and mix multiple tracks before doing this so you ...


4

Two thoughts... 1) The song is currently in the wrong key for the sound you are aiming for, vocally. The only fix for that would really be to re-record it in a more suitable key. 2) There are plugins available that can adjust the formant** of a sound without changing the pitch. Probably the best tool available for that is Melodyne [imho] which can adjust ...


4

This sounds like a live setup so I'd say the SM58 is an excellent choice of microphone for the purpose. I'm more concerned about the amplifier. A bit of research tells me this is an acoustic guitar amp which can also take a vocal microphone and although I haven't heard this model that setup is hardly ideal. Might be OK for practicing but if you're playing ...


3

Even if you get a really awesome sound effects pack, I wouldn't recommend using them as they come. You should try layering different and apparently unrelated sounds to get just the right sound effects. That said, it doesn't make a great difference what you use as starting point.


3

Hi pepe, While at Sounddogs.com some years ago I mastered the Masters Workshop vintage car collection for the in-house library. They have a 34 Rolls, 39 Packard and more. Not perfect matches but perhaps useful. The recordings are older (DAT), but they were recorded exceptionally well. Masters Workshop (a post facility in Toronto) had a recordist (Jamie, ...


3

Using an EQ for mastering should be a corrective process; so you need to know what you are trying to correct in the sound. Don't just use EQ or dynamics because that's what you think mastering is. If the sound is working, then leave it alone. If it isn't working, then you need to figure out what is wrong and what tool you need to use to fix it. If your ...


3

could you maybe use Soundflower to route the system audio into a vst compatible program? just an idea.


3

Okay - the one essential our core techy (my lead singer) has bought is a wide range of compressors. Maybe 12 high quality rack compressors now, and he swears by them - they make life a hell of a lot easier when recording, and they are independent of your DAW, which software compressors aren't. I wouldn't worry about effects until you either find one you '...


3

I would say Gross Beat falls under the time manipulation category, it just creates stutters by jumbling the audio around a bit, and changes the speed or volume level on certain tracks. I imagine that most of the sound effects created by gross beat could be achieved quite simply by cutting up and automating property's of audio clips. Hope this helps


3

Since your interface has an instrument input and the stomp boxes are designed to work with an instrument line, I would hazard that the best quality results will likely result from going in to the instrument input. The trick will be to get the output from the audio interface down to the correct impedance for the pedals which may require a significant amount ...


Only top voted, non community-wiki answers of a minimum length are eligible