Audio Interface questions....

ChaosToo

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So - my 'home studio' is taking shape!

So far, I have the Behringer C-1, Wharfedale 502 USB Audio Interface, a little Akai keyboard and I'm currently running Magix Music Maker 17.

The PC is an HP - Intel Core Duo2 E8400 3Ghz, 4GB RAM, Windows 7 Home Premium.

Latency seems to be under control, although the ASIO4All drivers are annoying in that I can't get playback when I'm using them - but that's something I can figure out in time (unless I'm missing something).

Quick question though (for now) - is the USB I/O 'all that', or would there be much to gain/lose if I used a twin mono 1/4" to stereo mini jack to connect to my PC via the rear soundcard 'line in' socket? (onboard sound is the Soundmax Integrated HD Audio).

So far, messing about with a single TRS to mini jack lead, I can't see much difference, but I'm very much a beginner and I want to get into good habits......

Any hints/tips/links on setups would be much appreciated ......

:D
 
Hmm, I see the Wharfedale is a "budget" interface but I say there is still a good chance it is much better than the soundcard audio in terms of noise. Did it not come with an ASIO driver? Most interfaces do and should be better than ASIO4ALL although it has come in handy at times as well.

When you say you can't get playback, how are you monitoring? You probably should have your speakers or headphones connected to the interface, again assuming you have an ASIO driver for it.
 
The Wharefdale is every bit a copy of the Behringer I had on order but which always seemed to be out of stock! It's a cracking unit and it has true 48v phantom power, unlike the Behringer! (which I need for the C-1)

The playback issue is when I add tracks - if I use ASIO, then I get no playback from the already recorded track(s) when I'm recording subsequent track(s) - using the Wave driver, I can monitor the playback track(s) from my PC's speaker socket whilst recording the other track(s), which I obviously need to do. If I monitor from the mixer, all I get is the track I'm currently recording (again - unless I'm missing something? The instructions are woeful!)

I'm sure given a bit more time I can work it out - but my first issue is whether USB I/O is better or worse than a jack I/O solution ........
 
USB vs audio card quality in a computer can have a lot to do with how well the computer motherboard and it's components are shielded for electrical noise. Some USB setups will generate electrical noise that will make it's way into the recording, while the sound card through the AUX/Line input on that same computer may not. You just have to try each way and see which sounds cleaner.

I have one old Intel P4 motherboard that is electrically loud. USB audio devices pick up the electrical hum off of this board and it ends up as trash in the recording. While the built-in sound card using the Line in jack is extremely quiet. So I used the sound card.

It your situation I would use the default Windows audio driver rather than the ASIO4ALL. The performance difference isn't going to be great enough to cripple yourself with not being able to play things back in other programs.

One point of home recording that is often overlooked is treating the recording space. Read up on acoustic treatments and how treat a room (and DO NOT use egg cartons or moving blankets). A properly treated room will greatly improve the quality of your recordings regardless of what equipment you use. As they say, garbage in equals garbage out.

People quickly get focused on recording gear. And while good recording gear makes a difference, remember that all that fancy equipment has to pass through some type of converter to get into your equipment. If you were to upgrade any single piece of equipment first - upgrade the USB audio interface. A good quality A/D converter from a company like RME or Apogee (if you own a Mac) will allow more detail and depth to be heard in your recordings than a $100 A/D converter. But...

It's all about steps and learning. Learn to treat your space. Learn how to use the equipment you have. Learn to listen to the recordings and edit them. Then if/when you upgrade your equipment you'll have a better understanding of what will best suit your need at that particular time.
 
That's great advice! I'll certainly get round to sorting the room out once I've got the basics sorted.

It's quite a small space, with laminate flooring, but there's enough angles in the room to break up the sound. For what I want to achieve, the current interface is just fine (and the mic is superb!). My lack of talent will be a drawback, but hey!......

Re. the noise levels, the USB is VERY quiet, so that's not the problem, it was more from an inherent latency issue that I was looking at USB Vs Jack/line - but, again, that might be just me not using the drivers properly (buffers and whatnot are a black art!).

Thanks for the input guys - keep it coming!
 
Re. the noise levels, the USB is VERY quiet, so that's not the problem, it was more from an inherent latency issue that I was looking at USB Vs Jack/line - but, again, that might be just me not using the drivers properly (buffers and whatnot are a black art!).

Everything passing into the computer and coming back out has latency. For a solo player laying down one track at a time it shouldn't really be a big issue. One tip is to only have one ear covered with headphones. Then you hear yourself live with one ear, while having music/singing playing into the other ear, and recording.

Buffers are easy. All buffers are is a staging area set aside in memory so that the audio has a place to "wait" until the recording software can write it to the file. When you are recording a single track at a time (a single voice, or a single instrument) buffers aren't really a big issue because the amount of data going to the recording software is minimal (relatively speaking).

A larger buffer will cause you to wait a bit more, but it can result in a more stable setup because it can hold more audio giving the computer time to process the signal.

A lower buffer results in less waiting and faster recording to disk, but it can introduce artifacts such as dropouts, stutters, etc.

You adjust the buffer until you find a setting that seems stable, is fast enough for you, and still has good audio quality.

I record with a buffer setting of either 512 or 1024. I'm using a quad Core i7 with 16 GB of RAM and record a single channel at a time. I could decrease the buffer size, but with my workflow I don't really gain anything substantial.
 
I've had the same issue before (quite a while ago now) where the active recording track was the only thing being monitored. I think it was a software or driver setting (signals properly routed). I'm curious to know what it was once you get it resolved. I recently got Pro tools MP9 and a M-audio interface and I have to say that I've had way less problems getting everything set up and working than any other interface/software combo I've had. I had a Tascam interface that I got so frustrated with that I left it on the sidewalk in front of the music store I was working at. I really should have destroyed it so it couldn't ruin anyone else's life........
 
OK, so I've been messing about and it seems the USB side of things does cause FAR more noise than even a hashed up mini jack to mini cable.

On my mixer, I have L and R balanced output (1/4") and I've just been using a single mini jack lead, with a 1/4" adapter on one end, plugged into one of the two MAIN outs and into the rear line input on my PC.

Obviously this is makeshift, but I presume the 'right' connection would be two mono 1/4" jacks to a stereo mini jack?

I also can't find a way to make the ASIO drivers playback, so I'm just using Wave drivers and they're working just fine with no latency issues that I can notice (it was more my timing previously I think! What can I say - I'm a drummer!)
 
OK, so I've been messing about and it seems the USB side of things does cause FAR more noise than even a hashed up mini jack to mini cable.

On my mixer, I have L and R balanced output (1/4") and I've just been using a single mini jack lead, with a 1/4" adapter on one end, plugged into one of the two MAIN outs and into the rear line input on my PC.

Obviously this is makeshift, but I presume the 'right' connection would be two mono 1/4" jacks to a stereo mini jack?

I also can't find a way to make the ASIO drivers playback, so I'm just using Wave drivers and they're working just fine with no latency issues that I can notice (it was more my timing previously I think! What can I say - I'm a drummer!)

Remember that signals from a microphone, and instrument, are always a mono signal. If you are recording with one mic, then you don't need stereo going to the computer. You just need the mono channel the mic is on. You can get away with a cable that is 1/4" mono at the mixer to a 1/8" mono to the Line-in on the computer. Lay each instrument track down as a separate mono track then convert each track to stereo in the recording software at the final stage.

If you need to record two channels at the same time, two mono 1/4 jacks (or RCA jacks) out of the mixer to a Y stereo mini jack would work. A company called "Comprehensive" makes some very reasonably priced cables that still sound good. Not sure if you can get the brand in the UK.

Generally using the Line-in at the computer is the safest choice. It helps to avoid creating multiple gain stages in the signal chain, which can introduce noise. The mic input on the PC will have an amplifier in it and it will introduce noise if the signal has already passed through preamps before reaching the computer.

With digital recording, recording tracks in mono, and then converting them to stereo in the software isn't really a problem as long as you are laying down each instrument as a separate track. You can work with the pan feature in the recording software to "place" the various instrument tracks where you want them on the virtual music stage in the final mix.

On a lot of consumer grade equipment the mono signal defaults to the left channel. With a mixer you can use the pan knob to determine if the mono will be left or right channel. Then in your recording software you select the appropriate input channel to be the main mono channel.

With higher end gear you are often able to split channels to be individual mono channels independent of each other.

Good for your for finding which input method is the quietest! Please excuse me if the stuff below is something you already know. I'm also posting it because these threads get read by other folks who are trying to learn.

More on USB noise below, but first... You are now beginning to work to address something called the "noise floor". In recording the term noise floor describes the level of noise, in decibels, that your recording setup (equipment), and space (the room) generates and hears when all the input levels are turned up to the proper recording level and no music, signing, etc. is being played.

How low a noise floor you need depends on what your are recording. Very quiet and delicate acoustic playing, and single spoken voices, require a low noise floor to have all the nuances of the playing and speaking sound great. Blaring high decibel metal rock with distortion doesn't require a low noise floor. Proper room treatment goes a long way to reducing noise floors.

Mics have something called "self noise", this is the amount of electrical noise that the microphone itself will introduce into the signal path. This appears in the recording as hiss. The lower the self noise of a mic in decibels (dB) the less noise/hiss the mic introduces into the recording. Condenser mics in the $100 - $200 range will often have a self noise around -20 dB. Mics in the $1,000 range can have a self noise around -12 dB or less. These numbers are important because the less self noise a mic has, the more actual signal you get into the recording from your instruments and vocals.

The noise with USB tends to go away as the quality of your gear improves (better attention to design and specs). When building a PC used for recording taking the time to research which motherboards are well shielded and grounded is worth the extra time.

All recording chains can pick up something something called a "ground loop hum". This is a hum in the signal chain that is due to an issue with electrical grounding somewhere in the entire electrical/wiring setup (it's a nightmare to try and fix in a home). It is present with equipment using three prong plugs and AC current. If your computer uses a two prong plug and has a DC power supply (laptops, Mac mini, etc.) then usually ground loop hums aren't an issue with the computer, but any equipment using AC can still pick them up (studio monitors, mic preamps, mixers, etc.) and introduce them into some part of the signal chain.

A company called Ebtech makes several solutions for eliminating ground loop hums. I use their Ebtech HumX three prong plug filters. They work great, but each one is limited to 6 amps max. I have one on the power chord of each of my studio monitors, and one plugged into my mic preamp. When I was using a Mackie mixer I also had one on the mixer's AC plug.

Lastly, mic preamps and mixers often have a "Low Cut" filter. This is a button that will reduce the signal below a certain frequency. Typically the frequency is either 80 or 75 Hz - low end rumble. Engaging this will help cut down rumble from HVAC units, electrical hum, etc. and lower your noise floor a bit. However some instruments (bass guitar and bass drum) may need those frequencies so you have to begin to understand what frequencies you want captured by what mic at what time.

Enjoy.
 
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Wow - that's a fantastic reply Spots - cheers for taking so much time over it!

Just as an aside, I think it's actually the software (Magix Music Maker) that is causing the issues. I've tried Audacity and it played right off the bat with far less USB induced noise. I'm also trying Samplitude Silver and Reaper on my next few days off.

Hopefully I can start playing and recording a few simple songs for the Seasons and then go from there ........ :D
 
Wow - that's a fantastic reply Spots - cheers for taking so much time over it!

Just as an aside, I think it's actually the software (Magix Music Maker) that is causing the issues. I've tried Audacity and it played right off the bat with far less USB induced noise. I'm also trying Samplitude Silver and Reaper on my next few days off.

Hopefully I can start playing and recording a few simple songs for the Seasons and then go from there ........ :D


Reaper is a good program, and the price is nice.

A nice free program is Presonus Studio One 2.

Enjoy.
 
Thanks for the heads up on Studio One - I hadn't seen that one before!

So far, I've tried Audacity, Reaper, Samplitude Silver and obviously MM 17.

I really like the look and feel of Reaper - but I'm finding the playback volume to be REALLY low (a problem many have, going by a Google search). Same goes for Samplitude - which made me think it's a settings issue, but Audacity and MM 17 don't have that issue, which has confused me!

I think some more time spent trying out a few of these packages, ironing out my setup issues and generally getting a better feel for things is the next step!

It's not going as easily as I'd hoped, but it's a fun learning experience. :D
 
Like Jim, I've never had an issue with the versions of Reaper that I've tried playing back quietly. Make sure you check the Windows volume settings for the recording and playback devices you have selected (Control Panel -> Sound -> etc).

One thing to keep in mind when you record. Digital recording should not be done at high levels. If you record and your signal is coming in to the recording software at a level of -15 to -12 dB, that is a good level. Don't try to increase it. Once the track is laid down, you can increase the volume to -2 to -3 dB on the peaks. This process of increasing the volume in the final audio is part of the process called "mastering". It is the process that brings the volume up to the levels you are used to hearing on CDs (remember those), etc.

Many people today record their digital signals way too hot. They try to keep the peaks around -3 to -6 dB. When you do this you actually loose detail and depth in the recording. Recording at those high/hot levels is a hold over from analog days when audio had to overcome the noise level of tape hiss, etc. 24-bit digital recording doesn't have that issue to deal with.

If you want more information on this type of stuff Google terms about recording and mastering digital audio.
 
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Digital recording should not be done at high levels...Many people today record their digital signals way too hot. They try to keep the peaks around -3 to -6 dB...Recording at those high/hot levels is a hold over from analog days when audio had to overcome the noise level of tape hiss, etc. 24-bit digital recording doesn't have that issue to deal with..

This presumes you're recording in 24-bit with a clean signal path. There are still plenty of budget interfaces that are 16-bit and budget mics may not give as clean a signal. Under those conditions, "record it hot" may still be entirely appropriate.

But in general, I agree that it is easy to try to do too much too early in the process. Mixing is a craft as nuanced as playing and takes time to develop and hone. True "mastering" is really outside the scope of the home studio although i contend you can get 90-95% there with modest equipment and oftentimes that is good enough.
 
I've been careful to keep the signal useable in the recording phase, as I kind of understand the whole clipping thing (which is just plain nasty in digital form). It's probably just a setting thing that I haven't figured out yet, but obviously I don't want to increase the playback volume unnecessarily because it'll just bring out all the background noise too.

Like I say - it's all a learning curve at the moment and I'm sure it's more me missing something in the setup ...... :D
 
Just a quick note to say thanks for the heads up on Studio One! It's fantastic! Nice and simple, but LOTS of possibilities to expand and mess with stuff, once my basic knowledge increases.

I've been running two of my electro-acoustic ukes straight into the mixer and have managed to get a decent playback volume - although I seem to have to have about 75% gain and level to reach -2db on the peaks (as suggested).

I'll be trying some mic'd up amp stuff soon (I know my condenser isn't best suited for it, but it'll do for now) to see how the levels are and I might finally get around to recording a track soon ...... :D
 
Just a quick note to say thanks for the heads up on Studio One! It's fantastic! Nice and simple, but LOTS of possibilities to expand and mess with stuff, once my basic knowledge increases.

Glad you are finding it helpful! It's an amazing program given it's price.

...although I seem to have to have about 75% gain and level to reach -2db on the peaks (as suggested).

I must have been unclear in my previous post. You do not want to record the mic or instrument so the peaks are at -2 dB. Record so the volume is initially lower (-12 to -9 dB on peaks), and then after you are done recording, use the program to raise the gain of the whole selection during the editing process.

Recording at a lower volume into the DAW, and then raising the volume later inside the DAW during editing, will give you audio that is richer, fuller, with more depth to the sound.
 
Ebtech humx is one of the best investments I've ever made...................without a doubt.
 
Ahhh - I get it now Spots - thanks for that.

Johhny - I'm lucky in that I haven't noticed any ground loop noise (so far) - but it's a useful thing to think about. Sadly, that product has the US type plug socket ........ :D
 
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