18
I don't know where you read about USB interfaces needing to "compress the sound", but it's nonsense. For your application two channels will do; at 96 kHz and 24 Bit that's less than 5 MBit/s: even USB-1.0 can handle that without compression. and USB-2.0 is already more than ten times faster, so even in multichannel applications it's often plenty good enough.
...
11
If you're not using any aftermarket audio hardware, and just recording from the line-in on the computer, you already have the bare minimum to record two channels at once.
That line-in is likely stereo, i.e., a left channel and a right channel. Or, more appropriately for your purposes, a 2-channel input.
Microphones are all mono (save a few specialty ...
10
Well, it depends! The first important consideration: what style do you play / wish to record? For a classical performance, you need quite a different sound than for a folky dance tune, a jazzy improvisation or even a rock or metal lead role.
The main part of the differences in sound is the room component. For a classical recording, you want a good-sounding ...
10
Yes, it is possible, but it isn't easy.
There are a number of free tools for editing audio in the frequency domain. I haven't had much luck isolating specific sounds with them, but I have been able to do some sonic manipulation that wasn't possible with traiditional audio editors.
Tapestrea, from the Princeton Sound Lab.
Spectro-edit
SPEAR
The latest ...
9
Your analogy with colour works in a way, but remember that a picture is millions of pixels. One HSV triple gives you a colour, but it doesn't give you textures. With, say, 16 pixels you could approximate a texture. The more pixels you have, the closer you get to being able to reproduce 'any' texture.
When recording audio as a digital signal, we measure the ...
8
As sound waves travel through bone as well as air, of course you will sound different to a recording. When you play back a recording you just won't get any of those sounds transmitted through your skull.
You will be able to approximate the sound by using an equaliser and boosting our cutting frequency ranges - trial and error is your best bet here, as no-...
8
I think your client wants the original wav-file, instead of the mp3 you sent, for the final product. I only sent mp3's when sharing demo's. If the client is happy, I sent the wav files, because you don't want the final product to have another conversion to mp3.
7
The purpose of a direct box is simply to take a quarter inch guitar line signal and convert it to a balanced signal that won't degrade over the longer run to the sound system. A DI doesn't do anything that can't be done with a decent recording interface since you aren't going a long distance with the signal.
You are best off to get an audio interface that ...
7
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 ...
7
In simplest terms, the gain is used to adjust the strength (i.e. voltage level) of the signal, whether that be within the electrical components of the mixing board, within the software DAW, signal sent to magnetic tape, etc. Volume, on the other hand, is used to adjust the loudness of the signal as you perceive it, whether that is through speakers, or ...
7
There are 5 stages/phases in a Album production: Recording > Editing > Mixing > Mastering > Printing.
It's worth naming "Pre-production", which is before the 5 steps and can be very important to make things go better. Pre-production can be a wide variety of things. All from making the music arrangements, defining artist identity, vision, and intention; to ...
7
The response curves are for each of the three selectable roll-off switch positions...
Flat
–6 dB/octave below 100 Hz
–18 dB/octave below 80 Hz
From the spec sheet -
Low Frequency Response Switch Positions
Flat; –6 dB/octave below 100 Hz; –18 dB/octave below 80 Hz
6
The job of a foam screen is to attenuate the velocity of wind before it hits the microphone. If you don't reduce the wind, then the mic will be overdriven, and you will get a badly distorted signal. You can't just filter that out after the fact.
This is also why an analog compressor or a digital compressor with a lot of headroom is a good idea between your ...
6
The only "natural" (i.e. "using only the data in the sample") way to change the pitch of a recorded sample is to change the speed of its playback. So a sample that sounds an octave lower should take twice as long to play back.
But you have a tool that somehow manages to play that sample back octave lower, while somehow keeping the playback time the same. ...
6
They will be able to reduce echo in their complete state, but don't expect them to help too much, as there are some caveats:
Just piling them up in a corner won't be very effective - they will absorb many frequencies better than a solid wall, but you still have flat reflective surfaces, and they will each have a resonant frequency. As long as the resonant ...
6
It's true that LDCMs are more sensitive, but that's in practise not such a big concern – good SDCMs already have plenty of headroom, usually the noise floor is well below any ambient noise even for chamber music room applications. Of course, the diaphragm mustn't be too small, cheap mics with less than a centimetre usually won't do. But they still beat ...
6
I wouldn't record using FX unless I was very sure about the end result I wanted and how it would sit in the mix, just to leave my options open since you can't go back and remove the FX. On the other hand, if the FX are an integral part of the performance then it might be a good idea to record after the FX. Sometimes the performer might hit a 'sweet spot'that'...
5
First, Audacity does support ASIO, it just can't include it in binary releases because the Steinberg license is GPL-incompatible. But if you build Audacity and the ASIO SDK from source, that should work fine.
Alternatively, you can use a different DAW. Reaper is very flexible, supports ASIO and a bunch of other non-free standards out of the box, and is ...
5
In general, if you unplug anything from a power amp, you could amplify small currents into a pretty loud pop...which could be enough to damage speakers or amplification circuits.
With phantom power, the problem is that when you unplug, the power to circuits is dropped quickly so you get current surges from inductors and capacitors, and these can be much ...
5
The quick and dirty explanation.
Dynamic:
Usually but not always cheaper
more rugged and less likely to be damaged so a lot of stage work is done with dynamics
Used for high sound pressure work like on snare drum or instrument amp
Sometimes used in studio work though usually vocals and acoustic instruments will need condensers
typical example Shure SM58
...
5
Never expect to get usable quality when recording through laptop built-in sound hardware, be it the microphone or just the preamp and AD.
A microphone that you can operate at all on your sound card's mic input is almost certainly not suited for piano recording; virtually all such microphones require 48V phantom power.
But you say this problem doesn't occur ...
5
The specifics of different implementations of these visualisations vary a lot. This answer may therefore not accurately describe your tool, but should give you an idea of what's happening nonetheless.
These visualisations plot frequency against phase. An audio signal can be represented by the sum of several sine waves of different frequencies, amplitudes ...
5
The main difference is Sensitivity (and Noise accordingly).
An LDC(Large Diaphragm Condenser) is more sensitive than an SDC(Small Diaphragm Condenser), and also tends to generate a higher output voltage, given the same input SPL.
Why the LDC is more sensitive?
Remember that a condenser mic is made of a conductive diaphragm next to a conductive backplate. ...
5
The most obvious answer is using compression on the input signal. The sound volume will be controlled and will stay below chosen threshold without loosing the character or sounding distant.
Not using a too sensitive microphone will help as well. The best choice here will be a good dynamic mic like Shure SM58 or SM57, but of course any other one will ...
5
This is an extremely basic live board that isn't really designed for recording. What are you doing with monitors and headphones? Do they actually have independent feeds? If not, I would suggest simply using the headphone output on your EMU and feed in in to a headphone distribution amplifier to send the output to multiple outputs.
You are getting noise ...
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