Hot answers tagged

7

This is often caused by accidentally sending a FX Return channel back into the Bus that is feeding the same FX processor that return is coming from, creating a loop. Select your FX Return channels and view the HOME screen, which shows you all of the 16 Bus sends for that channel on the right. Make sure that for each FX return, there is no send level to its ...


7

I believe "frame" is the term you're looking for - the set of samples from every channel for a given point in time.


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 ...


5

A good digital signal should either get there or not. Minor differences in cable quality are not going to have any impact at all other than latency. Latency is simply the delay in how long it takes to get to point a to point b and won't impact the sound quality significantly. USB is a digital signal, so a "premium" USB cable is a load of crap. There is ...


5

The DAW is not that important (FL is ok to start). Since you are getting started and you will find difficult to design the sounds, you need a good set of VSTi plugins that emulate the video game consoles' soundchips. The good news is that there are loads of these plugins to make chiptunes (most of them free)... just take a look... Chipmusic Plugins 9 of the ...


4

The S/PDIF I/O on the Saffire Pro 24 is stereo. The Optical is an 8-track ADAT input. The use case for using the S/PDIF I/O would be for using a digital stereo source or destination. You could use put an outboard digital effects processor in that loop, or say, get audio from a separate digital recorder into your computer, or bounce a stereo mix out to a ...


4

Mostly because of preference. Most people prefer to be able to choose an amp they like with monitors they like. Keeping the parts separate also makes maintenance easier. There may also be concerns about the power of the amp interfering with the speaker some or vice versa, though I'm not sure if this would really make a significant difference. The long ...


4

I think the answer is that they're looking to have the cleanest signal path. The introduction of a DAC is just one more link in the chain, and when the chain is only as strong as the weakest link, the fewer links, the better. Also, high-end DACs are an industry unto themselves. High-end monitor companies focus on the mechanical properties of the speaker,...


4

Analogue medium do technically have a higher dynamical range than digital due to the nature of being analogue ("atom"-level wave description, although you have material limitations of the material vinyl itself, but still). If the human ear can hear the difference of a 24-bit digital version of it, or even a 16-bit version, is of course open for debate. If ...


4

It's comb filtering. This happens because the two sounds are played not exactly at the same time, but with a tiny delay between, yet unlike in the real world the sound events are exact copies, not just very similar. As a result, significant portions of the sound spectrum will cancel in both copies, while other frequencies always add up; this uneven (but ...


4

Texture, AKA Timbre, comes from overtones, AKA harmonics. When you play a note on any instrument (including your voice), the note has more than one frequency. The lowest frequency is called the fundamental frequency. It is the one we hear most and the one that determines what note it actually is. The higher frequencies are what we call the overtone series....


3

As a primarily live audio engineer I can tell you that this is called a digital snake and they rock! As you know playing with old heavy analog snakes esp. with high channel counts is LAME! Digital snakes can carry hundreds of channels of sound over one ethernet cable using several different protocols and are of such good quality now, that even the seasoned ...


3

See also AES67, a layer 3 protocol standard that brings together Dante, RAVENNA, Q-LAN and Livewire. https://en.wikipedia.org/wiki/AES67


3

I find such arguments generally invalid. The whole point of digital is that the signal is transmitted as-is, without loss or coloration. Now, one might make an argument from the standpoint of interference (e.g., if you're running your USB cable alongside a thinly insulated power cable with a 10kw load), but such arguments are silly as well. If someone out ...


3

The M-Audio KeyStudio is a MIDI controller keyboard - it doesn't make any sound, rather, it sends MIDI messages to instruct something else to make sound. Since you're running it into FL Studio, I presume you intend to control some software instrument, so you will need to load that instrument, route the MIDI messages to it, and enable monitoring. All of that ...


3

Digital: +Easier to design +Low cost +Infinite replication (in software) +Total recall +Undo +Often easier and faster to use +More bandwidth/channels in a smaller form factor +Better S/N performance in hardware +Does stuff that cannot be (at least reasonably or cheaply, or at all) done with analog electronics +Usually implements visual feedback of the ...


3

First, lets start by clarifying that "volume" isn't a particularly technical concept. Most generally "volume" refers to SPL or sound pressure level which is the amount of pressure being exerted by sound waves and depends on the distance from a sound source. Rather what we are talking about here is the impact of signal level. Within an analog system, the ...


3

To make chip music sounding 'nostalgic' you will also need to recreate the techniques used in the old days where sound channel is limited. Like you can have only 2 square waves channel. (that was NES's limitation, 2 Squares , 1 Triangle , 1 Noise and (delta modulation) sample channel) So, in order to form a chord you would use a very fast arpeggio to create ...


3

Simply... the 'texture' or complexity of the wave. The addition & subtraction of a myriad simple sine waves of different frequencies & phases, overlaying each other. Without going into any detail, I'll leave that to the guy who gets the 'tick' for his answer ;-) ... a drawing of an apple & a photograph of an apple are both recognisably an apple,...


3

According to theory, all complex sound waves can be decomposed into a set of sine waves, which are the most basic of wave shapes. Two sine waves of the same frequency will be identical except possibly for amplitude. That is, one may be louder than the other, but they're otherwise the same and can't be distinguished.


3

A perfect sine wave can only be one shape as it is derived from geometry: So it always sounds the same. But yes, a wave can take any shape and this affects the timbre. An interesting point however, is that these movements all need to be reproduced by the speaker, which is possibly the weakest link in the chain. Consider how a speaker reproduces a perfect ...


3

After having a decent hardware system, your bottleneck of latency will be the signal vector size and the IO-buffers of your software. The smaller you can set them, the less latency you will experience and the more cpu stress will be the result. If your media is not able to deliver in time, you will also experience signal drop-outs, but that's why you want to ...


3

Yes, any analog transition or transmission has quality loss. How noticeable that loss is depends on a variety of factors including the quality of the DAC (digital to analog converter), the quality of connections, the quality of cables, the quality of any amplifiers, etc. That said, it isn't likely to practically matter particularly much. All digital audio ...


3

0dBFS 1kHz sine defines a peak value. Alignment of audio levels is usually done at an RMS level of -18dBFS or -20dBFS depending on the standard you are applying. The line level alignment level that the amplifier will use depends entirely on the manufacturer. For the sake of argument, let us consider that the alignment level is -18dBFS. -18 dBFS = 0 dBu = ...


2

tl;dr: digital has no headroom. 0dB is the max. this is the European kind. USA digital is 2dB hotter. give it a bit of time, it will start to make sense. from here


2

I rely on a lot of digital tools when making sound. For instance, I often put an EQ on every channel, have a lot reverbs running and some compressors too. All automated, so I can get the most out of them (to my abilities) all of the time. I run a large number of tracks and busses. And I must say the noise reduction technology of today is way better than it ...


2

I highly recommend reading up some literature upon the post sound process, especially John Purcell's Dialogue book and Dave Yewdall's as well. I say all of this because I took at look at the questions you have asked on SSD, and all 3 seem to be considerably related/reformulated and all are quite rudimentary 'bread-and-butter' topics of post sound. This ...


2

The best quick, comprehensive, practical introduction to the film pipeline that I know of is the DV Rebel's Guide by Stu Maschwitz. It doesn't cover everything in depth, but it will give you enough terminology to know what else to search for. Film and video have very different workflows, but based on your question, it sounds like that may be part of what ...


Only top voted, non community-wiki answers of a minimum length are eligible