Hot answers tagged audio-recording
5
Once a signal is digitized it is treated as a number (as you quite rightly point out) and for 16bits the range of numbers are -32768 to +32767. The numbers are created by an analogue to digital converter.
The analogue to digital converter (ADC) will have a maximum input range from -X volts to +X volts (i.e. real signals that you could measure inside your ...
4
Super simple explanation of the invert button. Imagine a simple sine wave. Thats the solid line in this graph. If you had the invert phase button on a console or use the plugin in your DAW does the same thing, you get the sine wave illustrated with the dotted line in the graph.
Why do you care? If you were to sum these two sine waves you'd end up with ...
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Phase is one of those things that is a lot more complex in reality than it first seems from the theory. Sound waves are transmitted through oscillating air pressure. Electronically, it is captured as a signal that maps the pressure changes to changes in electrical voltage. If you had a fixed frequency wave, it would look like a sine wave going up and down ...
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There is no direct or absolute correspondence between 'original' amplitude and sample value. The concept of dB itself is one of relative amplitude -- unless you specify a reference value, as in dBa or dBm, all you know about a decibel is that it represents a (logarithmic) ratio. There is no 'zero dB' as such.
In asking about original amplitude do you mean ...
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I'm surprised nobody else has mentioned this, but the unit in the digital domain is dBFS, where FS stands for Full Scale. 0 dBFS represents the maximum digital signal level. A signal would clip in the digital domain above 0 dBFS. Note, however that just restraining a signal to just below 0 dBFS might not be enough to avoid clipping: Depending on the method ...
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The tool you linked can widen the stereo image of the music and it does this by inverting the signal on one channel relative to the other and adding a portion of the right channel to the left and vice versa. Thus if a particular sound were a little bit "left" it would become "more" left as you increase the stereo enhancement.
The meter you mention gives ...
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Analogue Mixer Desks use resistors to mix two (or more) signals.
If your individual signals (left and right) have a recognized output impedance (50 ohm, 75 ohm, 100 ohm, 150 ohm, 300 ohm, 600 ohm, 100 ohm etc) then, the two signals will "mix" to form a composite signal.
If the two signals are from low impedance sources such as amplifiers then you need to ...
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Hmm, that's a rather complicated setup that could have problems. Normally combining a left and right channel is simple, simply wire the signal and ground lines together and the signals will combine to form a combined mono signal from the stereo signals.
The problem is that since you are splitting the signal, you need one output to maintain the stereo ...
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Play around with the ASIO buffer size, assuming you're on Windows. If you're on OSX, it's Core Audio instead of ASIO, but same idea.
Generally, ASIO4ALL (and actually, any low-latency audio driver) will expose a parameter for how large the ASIO buffer is. The buffer is a buffered set of samples between your sound hardware and your computer. The idea is that ...
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Your mic is ok. What you hear is called Audio Clipping. Clipping limits a signal once it exceeds a certain threshold of signal values representation.
You can also find some explanation here.
Unfortunately, when clipping introduced - the original signal is not reconstructable.
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Everything is a matter of drivers. Basically, the drivers must be installed with a ProTools 8.0.1 or less, and they WON'T WORK ON A X64 OS !
If you use the regular Digidesign drivers, they won't work on a 64-bits OS. The best way to make the MBOX work on 64-Bits OS is to download (they cost $60, but better take a TORRENT file) the PloyTec USB ASIO Drivers ...
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