The extremely embarrasing basic question thread

[Ricky_Spanish]

If your hi-passing bass or kicks u should use the best quality eq u have, ie not from a daw sequencer there is a marked difference with the result, probably 'cos they are designed with cpu efficiency in mind.

I personally would recomend the sonnox eq for surgical tasks which also has a 36dbf slope. And for mastering the PSP neon which is super high quality.

[Turnipish_Thoughts]

Not really an embarrassing question for me, but just something I'd like to know... Is there a difference between say using a parametric EQ to highpass/lowpass something or using a filter to highpass/lowpass something..? Same effect is it not?

This will explain the difference between EQ's and filters to you (Source ~ Sound FX; Unlocking the creative potential of recording studio effects - By William Russo)

5.2.10 FILTERS
An important third option exists for reshaping the frequency content of a
signal: the fi lter. Engineers speak generally about fi ltering a signal whenever
they change its frequency response in any way. Under this loose defi nition,
all of the equalizers we have discussed so far are audio fi lters. But to be
more precise, a true fi lter must have one of the two shapes shown in Figure
5.5. A high-pass fi lter (Figure 5.5a) allows high frequencies through with no
change in amplitude, but attenuates lows. A low-pass fi lter does the
opposite. A low-pass fi lter (Figure 5.5b) allows low frequencies to pass
through the device without a change in amplitude, but high frequencies are
attenuated.



Because the sonic result can be rather similar to shelving fi lters cutting out
extreme high or low frequencies, there is the potential for some confusion
between them. Filters distinguish themselves from shelving equalizers in
two key ways. First, fi lters are cut-only devices. They never boost at any
frequency. Shelf EQ can cut or boost. Second, and this is important, fi lters
offer a never-ending amount of attenuation beyond the selected frequency.
They do not fl atten out like the shelf. They just keep cutting and cutting. If
there is some unwanted low-frequency air conditioner rumble on a track
that you never, ever want to hear, a fi lter can offer signifi cant attenuation.
A shelf equalizer will have a limit to the amount of attenuation it can
achieve, perhaps only 12 or 16 dB down. The weakness of using a shelving
equalizer in this case is easily revealed on every quiet passage whenever
that track is being played, as it might still be possible to hear the air
conditioner rumbling on faintly in the background.

So to answer your question, A filter offers possible complete amplitude attenuation whereas an EQ has a limit to the amount of attenuation/amplification it can achieve. An EQ is usually used to modify the spectral content of a sound through attenuation or amplification, for example to compensate for frequency masking of clashing content between different tracks/sounds. A Filter is usually, on the other hand , used to either completely remove spectral content or, for creative reasons, accentuate complimentary harmonics (comb filter with resonance turned up, for example). An EQ lacks a resonance parameter, but with an EQ you are allowed to set the amount of attenuation/amplification. A filter will keep cutting indefinitely based on the steepness curve of the filter pole (12, 24 e.t.c.).

The two components are quite similar in a lot of respects but the differences are important and both EQs and filters are therefor intended for different (although similar) sonic alterations. Learning when to use which would be a wise choice.

[Ldizzy]

a lot of eqs include lpf and hpf tho... am i right?

[gen_]

no, those are not filters in a strict sense as they do not mathematically filter the sound out, they musically filter the sound out in an attempt to sound like expensive analogue equipment. (See the post I made already) They are poor for removing frequencies you can't hear as expensive analogue equipment is pretty much the same (low frequencies were removed after mastering on vinyl as they were printed, as they had a tendency to knock the stylus and make it jump at very high levels).

That and the fact that if you've ever seen a Filter built into a EQ on a spectrum analyser you will realist that most don't completely reduce sounds below the frequency to your noise floor.

That's why the expensive mastering suites come with multiple EQ/Filters. For example, iZotope Ozone has a Digital EQ mode specifically for this and Waves make the Q10 series of parametric EQ's (as opposed to the REQ range which is musical)

[eldoogle]

Ok. So a filter is to completely remove frequencies, and and eq is for creative use to make instruments sound better or to duck out a frequency that conflicts with another, like the sub and the kick drum? Or is a filter good to use for the kick-sub issue?

[Ldizzy]

no, those are not filters in a strict sense as they do not mathematically filter the sound out, they musically filter the sound out in an attempt to sound like expensive analogue equipment. (See the post I made already) They are poor for removing frequencies you can't hear as expensive analogue equipment is pretty much the same (low frequencies were removed after mastering on vinyl as they were printed, as they had a tendency to knock the stylus and make it jump at very high levels).

That and the fact that if you've ever seen a Filter built into a EQ on a spectrum analyser you will realist that most don't completely reduce sounds below the frequency to your noise floor.

That's why the expensive mastering suites come with multiple EQ/Filters. For example, iZotope Ozone has a Digital EQ mode specifically for this and Waves make the Q10 series of parametric EQ's (as opposed to the REQ range which is musical)

got a good read on this somewhere!?!?!?!.. i feel like im missing out

thx btw

[Turnipish_Thoughts]

fuck the boat

[symmetricalsounds]

Don't use Abletons EQ8, its an EQ not a filter. As Ldizzy said, each EQ affects sound differently (EQ8 is pretty bad imo). Get your hands on a free EQ VST or use Abletons other basic EQ (the one with just the knobs). The problem with this is your trying to do something very scientific and most built in EQ's and FX are designed to be musical. they say their doing something, and it sounds like their doing that something but often their doing something else entirely.

For example, I noticed last week that hi-passing live hats at 10k using Logic's channel EQ has the effect of actually raising the volume of them despite having remove quite alot of signal below the 10k mark and having no resonance or Q.

if you really are experiencing problems with eq8 then try right clicking and going hiquality.

also have a read of this http://rhythminmind.net/1313/?p=361

[samdam1]

whats a music?

[gen_]

Don't use Abletons EQ8, its an EQ not a filter. As Ldizzy said, each EQ affects sound differently (EQ8 is pretty bad imo). Get your hands on a free EQ VST or use Abletons other basic EQ (the one with just the knobs). The problem with this is your trying to do something very scientific and most built in EQ's and FX are designed to be musical. they say their doing something, and it sounds like their doing that something but often their doing something else entirely.

For example, I noticed last week that hi-passing live hats at 10k using Logic's channel EQ has the effect of actually raising the volume of them despite having remove quite alot of signal below the 10k mark and having no resonance or Q.

if you really are experiencing problems with eq8 then try right clicking and going hiquality.

also have a read of this http://rhythminmind.net/1313/?p=361

It's not really a problem as such, its just its analogue model is not one I am very fond of... it doesn't really fit with my sound.


As for info on digital filters or more specifically FFT filters which is what you want.
http://www.qsl.net/dl4yhf/speclab/filters.htm

Notice that there is no bell curve, just a steep drop, as the filter converts the sound into a 'picture' (like you would see on those FFT heatmaps of your track in Audacity) and deletes the frequencies you select. This is your ideal tool, otherwise you next best bet is a linear phase EQ. Every other EQ does not follow a specific method to equalise and as a result they all sound different and do different things to your sound. The final option is Brainworx's free Cleansweep, which EQ's each channel (L&R) seperately so plays less with your sound.

[gen_]

Sorry for the double-post but I felt this needed a bump for detail.


Taken from here, a great explanation of FFT vs standard filtering
http://www.kvraudio.com/forum/viewtopic.php?p=3919987

your EQ is (usualy) a DSP algorithm which works with samples in the "time" domain
this means that you give it 1 sample of audio, it returns one sample of processed audio
the filters inside this EQ usualy work with feedback (IIR), and are "casual"
the goal is to change the amplitude of different frequencies of the audio - this is doable..

FFT is something completely different..
FFT "analyzes" (if i can call it that way) the input audio and "builds" new information about it in another form, called "frequency" domain
you can think of this like you've split the frequency range with a whole pile of band-pass filters and you're now recording the individual amplitudes of these individual frequencies
but it does that with no filters, it uses a special technique..

so, in short, it "Transforms" the audio signal from time-domain (sampled waveform) into frequency-domain (amplitude levels of individual frequency bands)

then, this frequency-domain data can be transformed back into time-domain (so you can hear it) .. tho, it won't sound exactly the same (this depends on how dense the FFT was)

now, there is a difference between using an EQ (a bunch of IIR-filters) and FFT
FFT needs to look into the "future" this is no problem for offline processing (like your Audition app)
thus, the real-time FFT algorithm has to "accumulate" some of the audio _before_ it can start to output any data, which means that it delays the audio..

so how does your FFT-EQ work?
it just transforms the audio into frequency domain, then you just manipulate the amplitudes of individual bands, and then it transforms it back to time-domain
because it's offline processing, the algorithm can analyze the _whole_ audio if it wants to, and it can process with the highest quality possible, since there is no time-limit, you'll wait for it ;]

[tavravlavish]

How big of a difference is plugging something into your 1/8 inch mic input on your comp compared to using a audio interface in?

[Mammoth]

How big of a difference is plugging something into your 1/8 inch mic input on your comp compared to using a audio interface in?

it depends what you are plugging in.

[gen_]

How big of a difference is plugging something into your 1/8 inch mic input on your comp compared to using a audio interface in?

it depends what you are plugging in.

+1

Quick and dirty answer is that Audio interfaces have preamps which 'clean' the sound with magic and witchcraft. Sound cards do too but they're shite.

You can pay as much as 3 grand on just a preamp unit on its own and it is totally worth it. My dream unit atm is a Neve 1073 and thats £1800. I had the pleasure of recording someone through my cheap £100 mike and that and I swear down it made her sound incredible.

[cyz1]

How does one get smooth LFO changes? Whenever i'm shifting the LFO rate in say a drop it's always really choppy and uncomfortable.

[wub]

How does one get smooth LFO changes? Whenever i'm shifting the LFO rate in say a drop it's always really choppy and uncomfortable.

Are you automating the change or adjusting it manually?

[cyz1]

How does one get smooth LFO changes? Whenever i'm shifting the LFO rate in say a drop it's always really choppy and uncomfortable.

Are you automating the change or adjusting it manually?

Adjusting it manually, I don't know what automating is. I'm on logic btw

[wub]

How does one get smooth LFO changes? Whenever i'm shifting the LFO rate in say a drop it's always really choppy and uncomfortable.

Are you automating the change or adjusting it manually?

Adjusting it manually, I don't know what automating is. I'm on logic btw

Automation is where you prerecord the change in parameter you want carried out, so that when the project plays back, it does it automatically as opposed to you doing it yourself. I'm not a Logic user myself, but this video should be able to help;

[cyz1]

I liked the video you found and it answered some questions but the biggie for me right now is automating the LFO which i don't think was covered in it

[wub]

I liked the video you found and it answered some questions but the biggie for me right now is automating the LFO which i don't think was covered in it

Ok, the principle behind automation is the same, no matter what parameter (in your case, the LFO) is being automated.