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
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 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
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
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 ...
5
The problem is not really going to be the mic in your headset, although obviously a cheap mic is likely to be lower quality than a good one.
The problem is that recording in to a normal PC is at a very low voltage, so the difference between noise (mains hum, interference etc) and the signal you want is not that great, so when you amplify the signal you also ...
5
Honestly, my answer to this question, in which you infer that you plan to rely primarily on audio, is "do it any other way than you are planning".
Here's why; I work for an alarm monitoring company. In our line of work, where every signal from an alarm panel could be snot-nosed employees playing around or a matter of life and death, it's video, not audio, ...
5
Two things that can easily ruin quality, which you should avoid.
First is room acoustics. Even the best microphone won't get you a good sound if it's placed in a room with strong echo resonances. If you can't record in a professionally treated studio, you should do it in a room with as neutral sound as possible – avoid exposed parallel walls. Record the ...
5
shorts and a t shirt, to minimize clothing noise when doing a foot pass. clothing moves can be added separately. pants swishing noise can build up when doing multiple passes of feet.
5
A lot of Foley artists use sweat pants, soft clothes that don't make sound when rubbing together. Also take off any jewely, belts, etc.
5
Drum tuning is largely dependent on the style of music being played. A jazz kit will usually be tuned to exact pitches in a scale, such as the snare being tuned to the root, one tom tuned to the fifth, one to the third, one to the root an octave lower, etc. A jazz kit will also usually be tuned such that the top and bottom heads are in tune with each other....
5
Virtually all modern audio interfaces for computer use have clock-synchronized sampling across all inputs. This is a prime requirement for most ordinary multi-track recording. It would be difficult to find a modern, multi-track (>2) audio interface that did NOT meet this requirement.
5
Lower bit depths are not easily available simply because they are not looked for commercially, not because of some intrinsic technical difficulty. Audio interfaces or other capturing devices have their ADCs (Analog do Digital converters) based on integrated chips optimized for the features that are most requested commercially. In principle it would not be ...
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