Even floating point numbers can clip and degrade.
If I'm mixing, I prefer 64bit floats if available, but I use mostly analogue gear now .. so the mixing happens outside of the digital domain.
I do have some software I wrote to automagically remix music.. the jazz-o-tron 1000. Internally it uses 80bit IEEE samples (YES, 80.. its not a typo).. but I only use ...
Line in is an audio input normally around 150mv line sensitivity and can be used for devices such as tape players,cd players,mp3 players etc.
It cannot be used as a microphone input as you would hardly hear it, a microphone input needs to see an input sensitivity of around -5 mv input,as cd players and mp3 players already have an output of around 100+mVolts ...
Warning: this is perhaps not the answer you are looking for :)
You have already got enough gear. What you really need is time, patience, perseverance.
Experiment with what you already have, whatever it is that you own, use your imagination and creativity. A typical style of music was not only 'invented' because of the sound or purpose of the gear; it was ...
The only thing that really stands out to me about your setup is the lack of FX. You can safely disregard any effects built in to your mixer. There are a ton of great rack units out there from the '90s that can be found for peanuts on eBay and your local version of Craigslist.
Start with Alesis - take a look at the Wedge, QuadraVerb, MidiVerb and Ineko/Akira....
Well, as you've noticed yourself, it will normally work. This is because both channel outputs should be expected to have the same output impedance, so wiring them parallel effectively creates an averaging circuit. However, this is not really an intended mode of operation.
The output impedance of this combined output will be half the individual impedance, ...
The absolute essentials for home studio recording are:
A good mic. You will never get a natural sound out of the piezo element of your guitar alone; even the best ones sound plastic-y. You don't need a Neumann, but that $15 Radio Shack desktop mic ain't gonna cut it either. I would start looking around the $100/mic range for a dynamic mic, with the SM57 ...
When using a DAW, wether for midi sequencing or audio recording, the fundamental timing of the DAW is managed by a high frequency clock.
Depending on your DAW and audio interface, this clock source can be the internal clock of your PC, a clock on your audio interface, or an external clock provided by a third party hardware. But all pieces of the system must ...
That should work, as long as you ensure that the PC is not running all sorts of irrelevant software and processes, and you make sure there is enough virtual memory allocated (about 3 Gb - it is probably 32Bit OS, so no point in allocating more - make sure you make a fixed allocation.. no "let windows handle things" here as you may get unnecessary disk ...
How do I input two mics simultaneously on a Laptop?
To be able to input two mics simultaneously on a laptop, you need a two channels audio interface with two mic inputs. Using one interface for both mics will guarantee that both signal are digitized by a common clock, which is a must for your application.
There are numerous options for such an interface, ...
Echo make reliable interfaces and I'm gutted they've killed off much of their lineup. I'm hoping there's an Echo8 in the works.. Apogee Duet seconded. MOTU's Microbook looks promising but I'm hearing from people having trouble.
I've had Kyma for a few months now. The sound quality is awesome. I have been using it for things like
Adding spacial stereo width on tracks in a song.
Adding chorus to a track in a mix.
Reverb is a breath of fresh air.
I've also added saturation to tracks in a song.
Just the simple use of Kyma has upped my production in my mixes. Yes, I've made ...
Simple answer is if you are feeding a stereo signal to a single input you need the signal to be summed so you do not lose any elements.
To do this you need 3 resistors in the XLR end, diagrams are available online however its not a simple solder job. If you shop around you can get stereo to mono cables but make sure they have resistors and are summed to one ...
After a lot of research I figured this out. It's simple once you know which letters go to which:
I connected them as follows and it worked:
BROWN: XLR 1 (ground) -> JACK S (sleeve)
RED: XLR 2 (hot right) -> JACK R (ring)
WHITE: XLR 3 (cold left) -> JACK T (tip)
G is the chassis ground and is not normally used.
I still don't know ...
XLR 2 (hot) should go to JACK TIP,
XLR 3 (cold) should go to JACK RING,
XLR 1 (ground) should go to JACK SLEEVE
The TN, RN and GN connections are used in the XLR combi connectors with extra switching contacts.
On the picture these points are empty, so no switching contacts here.
Best practice is to connect XLR pin 2 to TIP (T), pin 3 to RING (R). Pin 1 is always GROUND. The TN, RN, GN, refer to TIP NORMAL, RING NORMAL, GROUND NORMAL, which would have additional pinouts if they were available on the connector. A "Normal" circuit is one that passes the current UNLESS a plug is inserted to break the connection (break the "NORMAL.") ...
You may want to look at "Zone Mixers" or "Matrix Mixers".
An example would be here
(this is just an example, I have no affiliation with website nor product)
One thing to keep in mind with such equipment is that inputs, as well as outputs, are usually single (mono) channels, and therefore you need to use 2 inputs, and 2 outputs if you desire stereo.
The audio processing speed is dependent entirely on the speed of the CPU and RAM in the machine. Storing the DAW software itself on a slow hard drive will not have any effect at all on the processing speed.
What will be critical however is where you store the audio that you are working with. Most DAW's will buffer audio to a certain extent, mitigating the ...
You need a multimeter to measure the voltage and current.
I assume this XLR pin usage:
1 .. ground
2 .. positive
3 .. negative
If XLR pin usage is different in your setup you will need to switch pins with the pins I describe below accordingly.
Measure Interface and Cable
Do the first measurments directly at the interface XLR port and optionally a second ...
This is known as gain-staging.
In a pro environment this is done by watching the meters at each stage, setting levels to give the highest signal without ever going near distortion (leaving headroom). High signal gives a lower noise-floor, higher SNR (signal to noise ratio).
In a semi-pro or consumer environment with no accurate metering you can really only ...
It is called a normalled bay when in the context of a patch bay. See graphic explanation below.
In your case though, it sounds like you want a monitor controller or small mixer.
See: Mackie Big Knob as an option with a lot functionality.
Also, do note that headphone output and speaker outputs tend to be optimized differently.
By definition you are looking for a mixer. A mixer is a device that takes multiple inputs and adjusts the levels in to one output. I suppose in theory, you may be able to make a passive circuit that could variably attenuate the signal and combine them while preventing back feed, but that would still be a simple passive mixer and might well have sound ...
A mixer, also known as sound mixer or [mixing-]console is a device that takes multiple analogue† signals (often: a lot of microphones plus some line signals, but it can really be anything), and combines them into a single analogue signal. It comes with a couple of dials where you can set how loud each channel is, and at least a simple tone control for ...
The FC3 is only compatible with keyboards with half dampering
capability. We recommend the FC4 piano style pedal instead.
NOTE: The FC3 is only compatible with: S90ES/XS, MOTIFXS/XF,
P120,140,155,P250, P60, P70, P85/P95,NP30, All CP series digital
- Yamaha FAQ
From what I understand, the FC3 features "continuous sustain". This ...
I recommend paying attention to various MADI solutions.
For example, I highly recommend to see the RME's MADI solutions in conjunction with an Ferrofish's A32 AD/DA Converter
You can add and add new MADI expansion in the chain - up to 32x/64x/and more analog ports.
Many professional external audio interfaces can be stacked or extended to get more inputs/outputs: RME, MOTU, Avid (Pro Tools HD), SSL, Apogee...
Some work by daisy chaining firewire or thunderbolt interfaces, others use a PCIe card with one of more dedicated digital multichannel interfaces that connect to rack hardware containing the analog inputs/outputs.
Not aware of any single units that will provide that many analog outs (but happy to be wrong) Also, FYI Motu have updated their product range recently so something like 2 x 16A's or 24Ao units could fit the bill