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43

In The Old Days... Prior to the invention of electronic calculators, multiplying and dividing long numbers was painful. Yet that was something engineers (including audio ones) had to do often. Logarithms were invented in the 17th century to simplify calculations. We have these fundamental rules of logarithms: log(X * Y) = log(X) + log(Y) And log(X / Y) = ...


10

What is 0 dB in digital audio? 0dB does actually represent the largest signal that a digital system can produce. Once a signal is in the digital domain it takes on a life of its own and 0dB remains full-scale. If you took a digital audio file (maybe from a CD or wav file) and loaded it onto another machine, the new machine would see the same binary numbers ...


6

Since the dB is a ratio as you say, "0 dB" just means "no change". In the context of digital sampling, I'd consider 'full scale' to be the reference point -- i.e. 255 in an 8-bit system -- making anything less a negative dB value. But unless it's specified, (as in dBa, dBm etc) it's just speculation and open to interpretation. There may be a DWS standard or ...


5

I'd say your basic tenet is sound, in so much as if it's too loud it will feed back - however, if there is a clearly-identifiable frequency at which the feedback occurs [assuming by that time you have no option of moving mic or speakers etc], then the next step would be to cut that frequency, allowing you to retain the overall apparent volume. Another thing ...


5

It is the amount of change. 0dB is full signal, negative dBs are a reduction in signal (down to -infinity which is fully off) and +dB are when a signal has a gain applied. 0dB is also sometimes called Unity. This is consistent across digital mixing and analog mixing where anything below 0dB is reductive (the source signal is reduced) and anything above ...


4

Most Condensor Microphones have a "pad" switch (usually -15 or -20 db). That will decrease your input volume, allowing for you to turn your gain up and have greater ability to fine-tune. Also, most mixers have a "pad" switch or button as well, and this will do the same thing. However, I have never encountered a condenser microphone producing such a ...


4

If I understand correctly, you are currently using your mixer's gain knob very close to it's minimum gain position, and you'd prefer to be in a middle position. This means that the output level of your microphones is close to the max input level of your mixer's inputs. Possible workaround (depending on your specific equipment) : There might be an ...


3

0dBFS 1kHz sine defines a peak value. Alignment of audio levels is usually done at an RMS level of -18dBFS or -20dBFS depending on the standard you are applying. The line level alignment level that the amplifier will use depends entirely on the manufacturer. For the sake of argument, let us consider that the alignment level is -18dBFS. -18 dBFS = 0 dBu = ...


3

Yeah, that happens eventually. I've been working with SM57s in live and studio setups for 15 years. In particular the ones used in live setups, which are used both indoor and outdoor, wears faster. I experience reduced gain, loss of low end and "dynamics". This is due to the membrane and edge hardening and dirt/corrosion in the coil hole. There is not much ...


2

The importance of gain is often overlooked. Many recording interfaces have some sort of clip meter Indicator (often in the mid price a single red light). Manufacturers often recommend setting gain until the light only occasionally lights up to get a desirable input level. However it is not just the highest loudness without clipping of the input signal which ...


2

tl;dr: digital has no headroom. 0dB is the max. this is the European kind. USA digital is 2dB hotter. give it a bit of time, it will start to make sense. from here


2

I've no experience with this type of gear, but functionality depends on the use-case. This device (and there are several others available with similar function) is mostly useful on ribbon and dynamic mics. But in most cases, as a sound designer or sound engineer, those types of mics are used on loud objects (drums, guns, explosions, cars). If you want to ...


2

Lots of lengthy answers here when all that is needed is a simple answer. The 0dB that you refer to is actually 0dBFS which stands for dB with reference to "Full Scale". 0dBFS is the highest peak digital sample level. Anything below this is normal signal, therefore shown as a negative number. -20dBFS is 20dB Below Full Scale.


2

Going to next level… Sound pressure level (SPL), in dB, uses the threshold of human hearing as the reference level (denominator): A fluctuating pressure of 2x10^-5 Pascals. I’d have to dig for the citation, but that reference was determined by averaging the response of healthy individuals (ears) back in the 1930’s(???). So, 0 dB is the threshold of human ...


2

Many people confuse two similar controls: GAIN control adjusts the amount of amplification in the first stage of the microphone preamp. It is used to "normalize" the amount of signal coming from various kinds of microphones. Some mics are very sensitive and need lower gain, and some are less sensitive and need higher gain. Gain controls typically do NOT go ...


2

You are talking about perceived loudness. This is quite unlike peak level. What you can do is measure the perceived loudness for all your songs and then adjust the gain accordingly. The best way currently available to measure perceived loudness is using the R128 standard. I hope you're on a Mac, because then you can use this free commandline tool: https://...


2

The process is identical, but one is tested, the other is not. Gain will not test for overs, it will just clip if you add too much. Normalisation will scan the entire track & only allow gain to be increased until the single loudest sample is at 0dBFS [or very slightly less if you want to avoid clipping on cheaper equipment. I've always used -0.3 as a '...


1

It may make a difference. Your sound card (and whatever else comes after your pre-amp) will inevitably add a certain level of noise to the signal itself, as will the pre-amp itself. If your pre-amp provides a very low level of noise, then increasing the gain will result in a better signal to noise ratio later on in the chain if there is noise being added (...


1

Can you give a bit more description as to what your asking. I have used plenty of microphones in live environments and what they need gain wise is always dependent on the specific make and models. You can't just grab a shure and an audix whether both condensed or not and expect the same performance. Condensers are typically more sensitive (Less gain needed)...


1

In theory there shouldn't really be any difference: Lets say you add some gain at the channel input, and reduce it equally by turning down the fader. Then there should be no difference. But preamps have their limitations: the more amplification, the greater the chance for clipping and non linear behaviour. Unless that is the exact effect you're going after (...


1

That combination should work. Make sure the Avid has phantom power on & is set for MIC (not "LINE") input. If that doesn't fix it, swap out the cables to make sure one of them isn't bad.


1

Your question is very unclear, however, unless your undefined 'USB soundcard' has an AU-compatible front-end or its own input gain stage, then you'd have to use something like Audio Hijack. You can't easily use VSTs on the input without an ASIO/VST controller & even then, they would normally be strapped to the output... which you would then need ...


1

The basic tenet is sound, the difference is experience. Feedback occurs when a particularly frequency of a sound is loud enough out of a sound source (such as a monitor) that it is picked up by a microphone that is feeding that sound source. This results in a loop that increases that frequency and turns it in to a constant tone. To prevent this, we must ...


1

MP3 metadata has a sort of dynamics compression hack where an individual track can say to the player “play me at 110% volume” and another track can say “play me at 90% volume” and the result is that those 2 MP3’s seem to be at the same perceived volume. With lossless audio, you don’t have this hack. You have to use actual dynamics compression. So the way ...


1

Replaygain is a metadata based system - it analyses the absolute peak value of your audio, then writes a scaling value which the player reads and uses to amplify the audio data when you play it back. The wave format doesn't natively support the metadata used for it so you have to either alter the actual audio data to normalise it, or use a hack to the ...


1

Yes, compressing twice is normally fine. In fact, as a tendency: the more compressing stages, the better – or rather: the less gain-reduction each compression stage applies, the better. One hard compressor will more likely sound unpleasant than two gentler ones. And in particular, compressing individual components before mixing results in a more ...


1

Use a compressor effect in any free tool such as audacity and set the threshold to be just above where the quiet audio peaks, and have a fairly high ratio and if it offers it, a faster attack. There should also generally be a setting for makeup gain or output gain, you can use that to bring the rest of the track up to where you want it to be Unless I'm ...


1

I have had this issue as well. I built my own microphones using condensor mics. The circuit was not my own and with it´s original design values I found that it was overdriving my mixer, A soundcraft Signature 16 channel.I could not go above the minimum gain setting at all. I eventually had to modify the circuit values so as to attenuate the output. End of ...


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