New answers tagged audio-recording
0
Problem solved. I bought me a preamplifier, with good gain control.
I also bought a condenser microphone but I suspect a dynamic microhone would have done just as well.
-1
If you are able to directly capture the digital feed and you record to a lossless format that matches the bit depth and sample rate of the original source, then no, you should not lose any quality. If it is going threw an analog hole, then yes, you will.
1
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 ...
1
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 ...
1
Success: 10ms
Just needed to disable the onboard soundcard.
(In computer sound preferences, not in asio4all)
0
Yes, you can use Csound as a VST using Cabbage.
http://thecabbagefoundation.org/
1
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 ...
1
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 ...
2
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 ...
3
Why you are using Asio4All?
Try using the driver from M-Audio.
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 ...
2
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 ...
0
The first thing to realize is that dB is a logarithmic scale. For every 10dB change, the signal level changes by an order of magnitude. 0db means the signal is unaltered when sampling or "full power" when dealing with a recording. -infinity is 0 signal strength and the amplitude of the signal can be anywhere in between.
To get a +dB level, ...
0
This depends on how high the quality of your recordings need to be. If you are looking to create a personal note library, or commentate on Multiplayer Gaming Videos, then I would just look at improving your recording technique. For example, moving the mic away from your mouth and speaking louder. You should be able to avoid plosives (pops) and your voice ...
0
It's difficult to know how significant these unwanted sounds are - can you post them so we can hear the track (or at least a section of the track)?
You have got to keep your wits about you when editing - I use wavelab and sometimes I put markers around the problem areas so I don't forget which part of the track I'm working on. Also getting a section of ...
2
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 ...
0
Definitely clipping and a poor quality mp3 - was that the mp3 taken from your phone?
Clipping is destructive - you lose information that is basically impossibly to recover. I tried a 5-band multiband compressor on the recording but it is so poor quality i gave up after a minute.
1
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.
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 ...
2
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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