23

A WAV file has the potential to hold "more" or "better" data than an mp3. WAVs employ no compression, no loss; they are as close to an exact replica as it is possible to get. An mp3 employs lossy compression to achieve the smaller data size. Lossy compression means that information is actually just thrown away if the algorithm decides no-one would be able ...


9

I think your main issue with understanding this is that you're looking at it from too simplified a view. Frequency response curves can only be so accurate, and even within a manufactured line of a particular mic by a particular company there may be slight variations. Polar response is not identical between two different model microphones (even though they'...


9

The standard ISO for a 31 band Eq is as follows HZ:20/25/31.5/40/50/63/80/100/125/160/200/250/315/400/500/630/800/1K/1.25K/1.6K/ 2K/ 2.5K/3.15K/4K/5K/6.3K/8K/10K/12.5K/16K/20K. But I think that this wasn’t the question, the question was how to calculate it, right? First, every octave doubles or divides per two a chosen frequency. Let’s take as a ...


8

Using a higher order filter will give you a greater roll-off slope in the filters stop-band. So a 1st order filter has a roll-off slope of -6db/octave, 2nd order filter has a roll-off slope of -12db/octave, 3rd order filter has a roll-off slope of -18db/octave, 4th order filter has a roll-off slope of -24db/octave, etc. This means the filter does not act ...


8

Very good question. I found nothing about it in the manuals. It seems I'm missing some knowledge that the creators of FM8 and Ableton's Operator share and use. So, my question is how do synth makers control the modulator amplitude and how does the interface of the synth and the controls from 0 to 100 change the modulator amplitude in the background? ...


8

No. When you convert a file from .mp3 to .wav, no new information is added: there is no way to regenerate the information that was lost when you created the mp3. All the extra data in the .wav file is redundant.


7

Yes, they double in frequency for each step. Seems like 10band eq's tend to start at 32hz and double through to 16k Like this: It's also nice to have a chart like this to put it in perspective: alt text http://www.beantownboogiedown.com/storage/Hertz-Chart.png?__SQUARESPACE_CACHEVERSION=1260244426517 If you can't make that out here is the link . So ...


6

If you zoom in to your waveform, you will see that it crosses the zero line twice per cycle (880 times per second). If you end your tone recording exactly at a "zero-crossing" then there will be nothing to create a "click" when played back. The "click" comes from ending the waveform somewhere above (or below) the zero-crossing. If the recording ends mid-...


6

There are three parameters of this filter that are described in the phrase "100 Hz 12 dB per octave low pass filter". I'll cover them in reverse order. Low pass filter - This means the filter does not change lower frequencies ("passes" those frequencies through) and blocks higher frequencies. Sometimes these filters are called "high cut filters", but that ...


5

-- edit -- this question piqued my curiousity enough that I ran a test for the tonebenders podcast. Check out the results here: http://www.tonebenders.net/tonebenders-episode-seventeen-questions-ozone-and-plural-eyes/ -- edi t-- I honestly think this is a good question that's worthy of a little thoughtfulness. IMO it is possible to eq one mic's ...


5

A 31-band EQ uses bands that are 1/3 of an octave apart. A 10-band EQ uses bands (frequencies) an octave apart. An octave is a 2:1 ratio between frequencies (doubling). This is a standardized form that was specified by ISO (International Standards Organisation).


5

I think you've misunderstood what Frequency is, with respect to audio. Whilst 'Frequency' typically is 'how frequently something occurs', in audio it's how many times a sine-wave oscillates in a second, rather than how many things you hear in a second. eg. A standard kick-drum track at 60 BPM means you'll hear a kick-drum sound once-per-second. That actual ...


5

MP3 is the 'colloquial' name for "MPEG 1 Layer 3" audio encoding. The purpose of mp3 encoding is to reduce the overall size of an audio data stream whilst maintaining an acceptable level of listening quality. It is implemented using a "codec", meaning that you need an "Encoding" function and a "Decoding" function in order to listen to the audio. The ...


4

There are definitely tendencies - and these mainly appear through the use of the same types of instruments in a genre, some more explicit than others, e.g. Drum and bass, Funk and A capella. Almost all modern electronic dance music uses a steady repeating kick drum and bass pattern - and as you note, this defines a good portion of the frequency ...


4

In order to make sound, your computer must drive the speaker with a time-varying voltage. In order to create the time-varying voltage, the computer must send a sequence of numbers to a Digital-to-Analog Converter (DAC). The simplest .wav file just contains a sequence of numbers that are ready to send to the DAC. An .mp3 file is a much more sophisticated ...


3

Those combined EQ + spectroscopes can seem a bit misleading. The curve on your EQ isn't always what's actually happening, and similarly metering is only an averaging of the signal because audio is a much faster rate than your monitor. With a regular eq it's really more about using your ears to find something that works. You might be better off using an FFT ...


3

It is true that the sensitivity of our ears varies based on frequency and that high pressure sound can be more damaging without being noticed, but if you are not listening too loud it shouldn't be a problem. You just need to be really careful that it isn't actually too loud. It is possible to damage your hearing without feeling any pain when you are using ...


3

You are applying a low pass filter - this tends to remove the middle frequencies in the spectrum a bit and the upper frequencies a lot. This is how I hear it anyway - try using a simple tone control to simulate the effect - probably just a fair amount of treble cut would do the same. If you want to be really precise use a graphic equalizer that has a bypass ...


3

The membrane is an analog device. It's operation varies continuously. Thus, you cannot change the overall recorded signal to sound like what another microphone would have picked up using static post-processing. Regarding noise, different microphones also have different noise prints. Different microphones also have different housings, which affects how the ...


3

FM8 mirrors the DX7 implementation which uses a 0-99 value to represent modulation index of 0 to 14. So, an output level of '85' corresponds to an index of 4.


3

Do you mean in livesound situations? If so, it depends on soo many factors that it is impossible to give a average frequency range. But something i noticed when working with a band that only use in-ears is that feedback through FoH started (if at all) around the lower midrange. I think this is common due to the way lower frequencies radiate. With the band i ...


3

Change is typically "disturbing". A sharp change of amplitude (a crash, a scream, an explosive sound, etc.) Or even sudden quiet after moderate but steady background noise will disturb some people. So a circuit that detects any sharp change (up or down) in sound amplitude. That would be my primary focus for detection of "disturbing sound". Of course, ...


3

It's a Logarithmic scale & is used where a linear scale wouldn't really make sense & would reduce the detail in the lower portions of the graph. As each octave doubles the frequency. The difference between, say, 50Hz & 100Hz is an octave, so is the difference between 5kHz & 10kHz. If that information were to be presented on a linear scale,...


3

By far most passive acoustic systems are linear and time invariant. As such, they don't create frequencies not present in the original signal. Passive systems that aren't linear are typically of the snaring/clacking variety. They either create overtones to existing frequencies, or they have their own strong resonance but are triggered by external signals. ...


3

Wavelength is the inverse of frequency (1/f) so all you need is to perform an FFT (Fast Fourier Transform) on your signal to get its spectrum (harmonic content). This can be done in many ways but from the way the question is formed ("Is there any way to extract wavelength ranges out of it? I only need to know the numbers."), I suspect Matlab or Scilab might ...


2

I wouldn't always say the winner is the expensive one, like a bad mic used well is always better than a good mic used poorly. Weeelllll sometimes : P But my understanding (and I possibly think this just to sleep better at night) was that the reason you can't truly just process sound from a cheap microphone is that it may have not captured those frequencies ...


2

I'm not an expert in psychoacoustics, but in trying out examples such as an 800Hz tone and another (which I varied from 802Hz to 1khz) they were all distinguishable instantly, with no effort. At very close frequencies (ie 0 to about 2 Hz difference) beats were the major audible component, but above that, two pure tones are heard. The only combinations which ...


2

. but that's just a guess, really.


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