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
There are two approaches in which you can learn it.
First one: use your ears! Experiment with what happens when you tweak the EQ settings for various kinds of audio, e.g. voice recordings, full mixed music tracks or white noise. You will soon get a feeling about what the highs/mids/lows are and which frequencies are prominent in which instruments. As a ...
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 ...
9
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 ...
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
A lot's been said about compression but nothing's been said about intelligibility. If you're producing VOs, trails, bumpers, idents or jingles with vocals over a bed, EQ is also your friend here. If you critically listen to professional productions, they often lowpass AND highpass music under speech in addition to heavily compressing and EQing the vocals. ...
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
5 ways a newbie can be more professional:
1) Don't mess with the levels while the recording is in process as this changes the signal to noise ratio enough to make the track sound wonky.
Set your levels before you punch the record button, I generally run as many takes as needed to set the peak load.
2) Don't expect everything is in tune.
Make double sure ...
7
There is no such thing as the audio quality of a MIDI recording, because MIDI contains no audio data. It's just a protocol with the information about when which key was pressed with what velocity. Transforming that data into an actual hearable audio signal requires some kind of MIDI synthesizer/sampler, which is then the only thing that affects the sound. ...
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
That's a well-known problem.
First of all, you shouldn't be using the microphone connector on the laptop. These inputs are usually not only particularly susceptible to buzzing noises, but also to all kinds of distortion and aliasing. Use an external audio interface, there are very affordable USB ones available.
Using an interface with XLR inputs, connected ...
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
The 'mic' input on your laptop is designed for a microphone's low level input. It has a preamp to boost the signal to 'line' level hence if you are feeding a line level signal into a 'mic' input it will be amplified into clipping aka distortion.
Assuming that your 3 channel mixer has "phono" level inputs and line level outputs, it would be good to get ...
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
Just add a compressor on your voice-over. Set the ratio to 2:1. Set a middle long release and a bit faster attack. (Or auto if you have it.) Then you lower the threshold until it gets better. Just play with the ratio and threshold until you get it right.
What the compressor does is that if the audio gets over the threshold in volume it will wait the number ...
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
In the studio I record my guitars directly in to Cubase or Guitar Rig with no distortion before the record chain. This way I can alter the effects precisely (I pop them in the effects loop) and is the way most studio work seems to go with other bands I know. If you do mic up your amps, you can get a very live sound, but it takes a lot of work to get it clean ...
6
At a basic level, you need an audio interface, AKA something to get the audio into your computer and, for acoustic guitar (or for recording the sound of an electric guitar amp/cabinet), you'll need a microphone of some kind. The "USB adapters" that you've seen are simple audio interfaces.
Many audio interfaces have USB connections and should work with ...
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
The important thing about such sounds is that the sources aren't coherent, neither the deviations in rythm and frequency nor the actual audio phases and amplitudes follow any predictable relationships. That's unlike the extra signals normal chorus/delay/phaser effects add, where the modulation is usually at best periodic and the relation very simple, only ...
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 ...
Only top voted, non community-wiki answers of a minimum length are eligible
