The audio interface would be used instead of your sound card.
The laptop itself shouldn't play a role in the quality of the sound. (Though it can happen... see @left's comment below)
The sound quality will be affected by the signal chain. I.e., the original audio quality, the settings of the software you use to edit the audio, the interface you ...
I've been very happy with the Stienberg UR28M. I love having a big volume knob as well as separate volume knobs for 2 sets of headphones, and it's nice to have dedicated mute, dim and mono buttons.
It doesn't have a ton of inputs/outputs, but I never record more than a couple of live sources at once so I don't miss them.
It's got balanced outputs for your ...
I think it is important to mention, although you probably already know, is that the quality of the instrument and the performance are hugely important factors. Something you can consider though, or maybe purchasing an inexpensive preamp like those by ART, and some other companies like the presonus blue tube. As far as the volume discrepancies, that is ...
The biggest problem with combining your mixer and interface is that most of the cheaper options have sub-par pre-amps, which would definitely affect your quality. I would personally recommend investing in a quality audio interface (a great option would be focusrite) with solid pre-amps, and then later you could invest in a cheaper mixer.
First – obviously, sound starts at the instruments. Rich, classical sound pretty much requires at least a decent baby grand. If you only have an ordinary upright piano, no miking can really get you there. Similarly, the room acoustics have an enourmous influence: any square room has nasty resonances, and even if these are hardly noticeable while you're ...
(In my experience) Scarlett products can have some issues automatically switching sample rates, so either Audacity was recording a 48kHz signal at 96kHz or the Scarlett was playing back a 96kHz signal at 48kHz (which makes more sense if it was really choppy).
You can try Focusrite's beta drivers if you still have issues, they've improved the automatic ...
According to the Steinberg UR22 operation manual, you need to setup the device in Cubase:
When the “ASIO Driver Setup” window appears while the Cubase series
program is launching, confirm that the device is selected, then click
“OK.” (page 11)
If that dialog does not appear, you need to go through the settings manually:
In top menu go to Devices
There does not seem to be a shortcut to jump from a pattern on the playlist to its corresponding mixer channel. I assume this is because it is possible to combine multiple channel rack instruments within a single pattern.
I recommend keeping instrument, pattern and mixer insert names consistent so you can visually scan for them. Additionally, you could ...
Did you check the manual, see snippet below. A SM58 is a dynamic microphone and needs to be connected at microphone level.
You will need a XLR-XLR cable.
Each analog source input channel provides combination XLR/TRS on the rear panel. These balanced/unbalanced analog audio inputs sup- port the following input levels:
• Mic (microphone) for ...
It depends on the driver.
I used to have a Behringer UMC404HD and that showed up as 2 devices, it had 2 channels combined into each device (channel 1+2 and channel 3+4).
My TASCAM interface for example shows up as a single 16 channel device.
I don't know about your other two products. You could contact the manufacturer.
One side note: Behringer is doing a ...
This isn't a task I normally do using System tools, but I think you ought to be able to do it by creating a Multi-Output Device in Audio Midi Setup [Apps/Utilities].
Click the + button then check which devices you want to use as simultaneous outputs. Select the Multi as your output device in each desired App. If an app has no direct routing capability, set ...
I was first going to write a long answer with all permutations on how you could hook everything up, in multiple configurations. However, this would be a very long answer, and would probably complex the matter in regards to the choices you need to make.
The positive thing is that you have a good audio interface, which will do most of what you want. The ...
What you need is any interface with an XLR input that provides phantom power. Either that or buy a preamp that provides it.
So that's all you need with your interface:
XLR in (also called Canon)
I'm sure the focusrite is fine but there are hundreds of choices.
The NT1-A is a good mic, but bear in mind as it's not dynamic, it will pick up ...
Your question is a bit confusing, but I will answer by showing you an example.
CP300 Piano - Line Outs = 2 x XLR / 2 x 1⁄4 inch(Phone) Jack
Silent Violin - Line Out = 1 x mono 1/4 inch TRS(balanced phone) Jack (a guess?)
Microphone - Mic Level = 1 x XLR
So let's look at a scarlet interface:
Scarlett 2i2 (User Guide - 2nd Gen) - (...
Every USB-MIDI connection adds at least one or two milliseconds of latency.
A single USB connection typcially is not noticeable.
Every DIN-MIDI connection also adds latency; at 31250 baud, a three-byte message requires about one millisecond to be transmitted.
A MIDI Through connection does not add latency, but the MIDI input circuitry slightly distorts the ...
I have successfully used the original Firepod (one before FP10) on El capitan. Do not use the Presonus drivers. They are not necessary. Just plug and play. Presonus drivers will likely cause a problem. The hardware however is working fine, with no drop outs or issues. My computer is a 2008 Mac pro 8 core system. This may be more of a new Mac hardware ...
I retired my FirePod for years because Presonus discontinued this product and stopped updating driver to support new versions of OSX.
I gave up when I upgraded OSX to v10.8 Mountain Lion. Presonus states clearly in this page that FirePod/FP10 doesn't support OSX10.8.
I'd go with two of the same microphone type (large-diaphragm type), one close to the piano and one further away to capture some ambient sound and reflections in the room. This way you can adjust the level of directness/"wet"-ness afterwards (moreover the ambient source may be a better choice if you afterwards want to apply artificial reverb, should your ...
I haven't used the NT1a in a very long time, but the issue I remember having with it was being pretty cold and not having a firm core to the sound. The NTG-2, on the other hand, sounds awful. I've never actually used my own even though I got it for free from a friend (I did give it the benefit of a doubt when I first got it in the studio alright to see if it ...
Have a look at Usine, it's designed to be used for touch interfaces on windows and is a fair bit friendlier than Max for building with. Don't think it'll run under WinRT, but full Win8 on a tablet shouldn't be a problem.
Your best bet is probably the Open Sound Control protocol. But you'll have to look around since I'm unfamiliar with Windows 8 applications. If it's backwards compatible with Windows 7 stuff you may be in luck.
If super fast response times is not my first priority (as would be the case for live on stage performances), the easiest, most flexible solution that works well for me is a simple VNC remote desktop app running on any tablet that is connected to my main audio pc or laptop. It shows either a custom interface I built with Max/MSP (which runs on the main audio ...
I find, certainly with short notification sounds or even power-upesque sounds, that creating a a small arpeggio works. Put some delay on this, then pitch it up so that it affects the speed also, until you find the sweet spot.