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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....


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 ...


6

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'...


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 ...


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

There is a VERY simple way to do this, as I have needed the exact same effect before. Click in an empty plugin window, go to Audio Units > Apple > AUpitch this will allow you many many semitones of pitch shift with the ability to automate smooth changes. 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 ...


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

I feel your choice of plugins might get you near the sound you want to hear in your song (heck, they might even get you somewhere better!) but as for recreating the distinct sound of AM radio (specifically AM TALK radio)... well, I'll just say that your way seems like WAY more work than it needs to be and start with the explanation. Before we start ...


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

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


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

Normally mixing is done with a single voice on each channel because it is impossible to completely separate one audio signal from another after they are combined. Sound is additive and there isn't a way to identify one from the other to pull apart cleanly. That said, what you can do is enhance parts of the sound that will make one voice more clear than the ...


3

Another point to consider is the harmonic content of the instrument that you are EQ'ing. Above (and sometimes below) the fundemental note and frequency of the instrument there will often be harmonic and distortion that make up other portions of the sound. This is the main reason that you can't just slash away everything but the fundemental. Sure you would ...


3

You could either: 1, manually automate the level like a square wave 2, use a tremolo plugin set to square wave and automate the mix control 3, insert a copy of the sample into a sampler and punch in 16/32nd notes with slight gaps (turn release down to 0) 4, chop small sections out of the clip. There's a few alternatives, but they are usually plugins ...


3

Instead of simply listing practical details about how you're making your sounds (eg. types of mics, foley techniques and so forth), you might consider making more theory-based speculations about why certain sounds, or qualities of sound, are appropriate to convey the particular messages that you're trying to convey; how they create meaning for the listener ...


3

Definitely. "Guitar" effects have almost never been used only for guitar. Keyboard and bass players in particular have long used effect pedals. Effects pedals have also been used live and in studios to process all kinds of sources, including vocals. Many times processing some sources like vocals require adapters and level and impedance matching to make the ...


3

Yes, it is simply a compressor - typically you'd use one with a relatively low threshold and ratio. And some may store metadata for the whole track once it has been compressed/normalised once. I know my car stereo could do that - would zip through new tracks to identify peaks and normalise against them. You don't need predictive normalisation though - ...


3

It's 'simply a compressor'... however it's a very specialised type of compressor. There are probably others in this field, but the go-to name for radio compression is Optimod by Orban Optimod is, to over-simplify, a multi-band compressor specifically made for radio transmission & includes specific timing & frequency compensation for the way radio ...


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