5

less tracks per project. less plugins opened. if you use your factory soundcard (the one on your motherboard) you'll probably suffer from latency and a bad signal/noise ratio. But you still can definitely work with it (music, mixing, and even small movie project(s)). FYI : when a computer is too limitated for your work, don't forget you can always render ...


3

As detailed in the online manual In general these days, the power-off switch on digitally controlled pro audio equipment initiates a controlled shut down procedure instead of just killing the power supply, so you need to worry less about “popping” the system. That being said, you should probably still ensure that all amps & powered speakers are tuned ...


3

You really just need the best of everything you can afford - as an extreme example, marrying a dual 6-core Xeon to 2GB RAM & a 64GB 4200 rpm drive would be rather a waste. Very very broadly... CPU clock speed will give you raw power. CPU core count will give you multi-threading; higher plugin count. RAM will give you 'room to breathe' for everything ...


2

It all depends on particular case, but that setup of yours should do nicely. It did whole EDM production on PC FAR worse than your laptop, but it reuired me to get really creative. But it could be done easily. Just remember two things: don't go crazy with modern VST plugins, more often than not stock plugins in your DAW of choice are far better optimised, ...


2

For straight recording, I'd say you'd probably manage. If your audio card has it's own processor, then it should be fine. You may have to cope with a large latency setting. When it comes to effects, I think it will NOT cope. I have a laptop that does great from 2006, but I've upgraded its insides, it's now got a 2GHz Dual Core T2500 (2MB L2 w/667MHz FSB) ...


2

In short, no. Your DAW will have the ability to route signals to and from the interface as is. Each channel strip in the DAW can have sends and receives, same for the monitor channel. For example if your DAW has a monitor channel that automatically receives solos, you route that monitor channel to a hardware output on the USB interface. The exact way of ...


2

The "Tape" input is effectively identical to "Aux". It is almost certainly an ordinary line-level input. It should be trivial to connect the speaker or headphone or line output from your computer to your stereo box. It would be enormously helpful to show us a photo of the BACK of the amplifier unit.


2

It depends on the digital mixer. Many digital mixers run on firmware rather than software and such firmware is much simpler than a modern operating system, so sudden loss of power is not a concern due to there not being any critical state data that could get lost. The board will start up clean from a fixed state every boot regardless of what you do. There ...


1

Solved- I got a Mackie Big Knob passive($60) and it solved the issue..


1

The "fix" is to get an interface that doesn't do this, or never let the machine sleep. I have a similar problem with dynAudio BM6a's in that when the interface switches, the 400w bi-amps really don't like the bang. Fortunately mine don't whine or scream at me, just the bang, but it doesn't take many repetitions of ear-damagingly loud recurrence to ...


1

This should be a Memory Problem, the pops and clicks etc. are memory overruns. Turning up your Buffer Size/ Sample Rate or deactivating some Costy effects while producing because when you render/export your audio file these clicks and pops will not be happening anymore. If ur PC has a very low ram to use (4 or maybe 8 gigs) having 30+ effects is mostly not ...


1

Link them up exactly as you would with a guitar amp - guitar > pedals > amp/Scarlett. Analog effects pedals don't have latency. Only once you go through the computer will you see any latency at all. If the Scarlett has a 'through' setting that avoids going through the computer before it can send to headphones/monitors, then you can avoid that too.


1

There are two types of condensor mic - Electret condensor and true condensor. Both require a positive DC charge in order to function. Condensor mics are based on capacitative plates which physically move in the presence of sound waves. This movement alters the static charge on the plates and thus the voltage between the plates changes. This voltage can be ...


1

You need a cable that has four connections. Ground, L, R and mic. This will need to split to two connectors - one stereo connector and one connector for the mic as usually, on PC's the mic connector is seperate. IF you can, try and find a way of using the bluetooth interface for everything - that will give you a better chance of making it work.


1

An audio interface is definately what you need... Really this is very basic stuff, so just go on Google and type "Usb audio interfaces" and just find one that suits your pocket and your needs, it'll instantly make a world of difference and actually set you on the right tracks. If you are any close to being serious about your art/recording/music or w/e ...


1

You should look if there is a jack called "AUX", it will take an external source as an input for your amplifier, otherwise no easy way to do it.


1

I would suggest also some GNU/Linux distribution. You need to learn some skills about it, but it is not hard. I have recently installed Debian 7.8 on my VAIO from 2005. I can easily do video editing and that includes effects, without a separate video card! So I would recommend: Debian 7.8 (not 8.1) with XFCE desktop ...and important: use only stable ...


1

It's difficult to record several people talking with just one microphone, because using just one microphone, it will have to be omnidirectional (i.e. capture sound equally from all directions) and be placed at an intermediate distance between all participants and this can cause three problems: Lack of separation of ambient noise from useful sound. It's ...


1

Nope, the best case scenario - you'll only get one good sounding mono channel. Stereo cords and adapters have two rings on their jack while mono only has one


1

The "three prong plug" you are talking about is an XLR connector. It is pretty much standard with when working with professional audio gear (it is rugged and less susceptible to noise). XLR connectors look like these two (female and male): The GXL2200 microphone has the following specifications - notice the P48 phantom power requirements: Operating ...


1

You need an audio interface, and one with phantom power (that microphone needs phantom power). An interface will connect to the computer through USB or Firewire (typically), and handle the input and output of audio (converting analogue to digital and vice versa). Connecting your microphone(s), instrument(s) and speakers/headphones to the interface is just ...


1

According to the datasheet for the microphone (located here), this microphone requires a 48 Vdc phantom power supply. You can fabricate or purchase a suitable phantom power supply with a balanced-to-unbalanced adapter so that the mic can be connected properly to your laptop computer. Alternatively, you can purchase an inexpensive mixer that has the proper ...


1

I have a "recording" folder on my HomeComputer and a "recording" folder on my StudioComputer, and I wanted these working folders to stay in sync, similar to dropbox or google drive. The folders contain my Logic 9 projects, sample libraries, and small video projects. The solution I found is synchronization software called GoodSync, and it's working pretty ...


1

Transmitting digital audio is not just about sending 1s and 0s at the right speed. Whenever you transmit a digital signal, it's actually analog. You have to modulate the digital signal into an analog one and then demodulate it back into a digital one on the receiving end. (This is where the term modem comes from - modulator/demodulator) So in addition to ...


1

Im adding to this, except the room / ambient noise that stuff may produce (cars passing , cpu fans etc) i think you are refering to digital hashing noise, what you said as electronics noise lowering the windows volume will not fix this, this is a largely known issue and there are some ways to only try beating it. Some people never beat it and it has to do ...


1

...and turn down the microphone gain to reduce background noise... That's patently counterproductive. The gain on a solid-state mic preamp should always be set as high as possible without clipping; that's the whole point of the preamp. In particular, the lower you set the gain, the higher the effective noise level will be, because any boost later on will ...


1

Is an audio interface a requirement? It seems to me you could internally route these signals with a software solution. I am not a mac user, but on the windows platform there is a piece of software called Virtual Audio Cable that allows you to create additional audio channels and patch them how you see fit. I suspect a similar solution exists for Macs. I'...


1

Each channel in your DAW is limited to a single thread for all signal processing that occurs on that channel. This is because splitting the channels threads would introduce latency in playback. More than one channel can share a thread. So multi-threading is a bit of a lark. If your running a lot of channels, threads can be beneficial because your pool of ...


1

There are some issues to solve, but it is doable. I wrote a Java theremin. Available to try out if you have Java 8 running. Consider this a beta test version--please write back if you have suggestions! Contact info is on website. http://adonax.com/main/musicapps.htm Main issues that I've come across: latency smoothness -- easy for there to be "zippering" ...


1

https://www.eurekalert.org/pub_releases/2017-03/qmuo-qpo031317.php This reference "describes for the first time the use of the Filter Diagonalisation Method (FDM) in music signal processing. The technique has origins in quantum physics and is employed to study molecular dynamics and nuclear magnetic resonance." I begin to imagine how quantum phenomena could ...


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