i have a Tascam US-122, a Behringer UB2442FX mixer, and of course an SP-303.
i was wonderin' how i could record into my computer with good sound quality that's not midi, MIDI is midi, so i got that sorted out (messin' with limiters on ableton) but for some reason i thought doing INSTRUMENT-MIXER-TASCAM BOX-USB-COMPUTER would make the sound quality good, but all it does is gives it delay in recording it and i feel like i've wasted lots of money buying the mixer.
any suggestions on what i should do?
also what should i do with my mixer O:?
Making good quality noises!
Moderator: TGN
Re: Making good quality noises!
hi
i don't know the Tascam interface. i use an M-audio fast track pro, a low/mid level interface. what i can say is that the difference between low/mid level and mid level interfaces (like for ex.motu, tc electronic etc) is not many on the quality of converters (yes, of course there is a little difference but not that big), but the main difference stands on the preamps. the preamps of my M-audio are not that good. i mean, if you rise the signal to "14h o'clock" situation is still ok, but if you rise more, then some hiss noise begins...
i have also a yamaha mg102c mixer to record more than 2 instrmennts simultaneosly, so i'm pretty like your recording situation. this mixer is ok, but when you rise few knobs, you have to "equilibrate" well the all the knobs levels + the master knob in order not to have some hiss noise sort out...so, in my case, i try to make a sort of balance between M-audio FTP preamps/Yamaha mg102 volume. once i record without hiss, i can always rework the tracks on a software in order to establish better volume's levels.
in reality, the best thing should be to have an audio interface with more than 3 stereo ins, like this you (and i) could recorder straight into the audio interface: in an "audio chain" is better to have less passages before get into the pc. the sound will be like this more natural.
other solution:
you can apply a preamp before put the sounds into the audio interface: there are also preamps that have digital out but i don't know if your Tascam interface got digital in...
i don't know the Tascam interface. i use an M-audio fast track pro, a low/mid level interface. what i can say is that the difference between low/mid level and mid level interfaces (like for ex.motu, tc electronic etc) is not many on the quality of converters (yes, of course there is a little difference but not that big), but the main difference stands on the preamps. the preamps of my M-audio are not that good. i mean, if you rise the signal to "14h o'clock" situation is still ok, but if you rise more, then some hiss noise begins...
i have also a yamaha mg102c mixer to record more than 2 instrmennts simultaneosly, so i'm pretty like your recording situation. this mixer is ok, but when you rise few knobs, you have to "equilibrate" well the all the knobs levels + the master knob in order not to have some hiss noise sort out...so, in my case, i try to make a sort of balance between M-audio FTP preamps/Yamaha mg102 volume. once i record without hiss, i can always rework the tracks on a software in order to establish better volume's levels.
in reality, the best thing should be to have an audio interface with more than 3 stereo ins, like this you (and i) could recorder straight into the audio interface: in an "audio chain" is better to have less passages before get into the pc. the sound will be like this more natural.
other solution:
you can apply a preamp before put the sounds into the audio interface: there are also preamps that have digital out but i don't know if your Tascam interface got digital in...
Re: Making good quality noises!
For your delay problems.. the mixer should not be the issue unless behringer have used a crappy DPS for the effects and in order to keep it in sync with non-effected signals have created a delay on the entire input.. but this should not be the case.. you should be able to measure if there is a difference by trying with and without the mixer.
Most likely it is the latency buffer size set for your sound card (you should be able to set it in any audio software / DAW).. A mid-quality audio interface should be looking at an input latency of 5-7ms when using say less than 2 stereo inputs.. If you're using windows you could try using ASIO4All drivers. When setting your latency you need to put it as low as it will go without creating artifacts in the sound (clicks/pops/glitches)
Also dont forget that there will be another latency on the output of your audio interface.. usually about the same length of time. the total latency from input to you hearing it will be the sum of the input latency + the output latency, but when you make a recording on the computer, it will only be delayed by the input latency.
A total in+out latency of less than 20-25ms should really not be noticeable unless you're playing complex piano etc. - even 35-40ms latency is barely an issue for most things.
As for getting a good sound - the mixer won't change it alot, unless you use the EQ (there is something nice about analogue eq)
The trick with this is to use the pre-gains up the top of each channel strip.. turn them up to about 3 oclock. this keeps the noise level much lower than if you use the gain controls at the bottom for each channel to boost it.
Most likely it is the latency buffer size set for your sound card (you should be able to set it in any audio software / DAW).. A mid-quality audio interface should be looking at an input latency of 5-7ms when using say less than 2 stereo inputs.. If you're using windows you could try using ASIO4All drivers. When setting your latency you need to put it as low as it will go without creating artifacts in the sound (clicks/pops/glitches)
Also dont forget that there will be another latency on the output of your audio interface.. usually about the same length of time. the total latency from input to you hearing it will be the sum of the input latency + the output latency, but when you make a recording on the computer, it will only be delayed by the input latency.
A total in+out latency of less than 20-25ms should really not be noticeable unless you're playing complex piano etc. - even 35-40ms latency is barely an issue for most things.
As for getting a good sound - the mixer won't change it alot, unless you use the EQ (there is something nice about analogue eq)
i have also a yamaha mg102c mixer to record more than 2 instrmennts simultaneosly, so i'm pretty like your recording situation. this mixer is ok, but when you rise few knobs, you have to "equilibrate" well the all the knobs levels + the master knob in order not to have some hiss noise sort out..
The trick with this is to use the pre-gains up the top of each channel strip.. turn them up to about 3 oclock. this keeps the noise level much lower than if you use the gain controls at the bottom for each channel to boost it.
Samples.
Re: Making good quality noises!
cartesia wrote:i have also a yamaha mg102c mixer to record more than 2 instrmennts simultaneosly, so i'm pretty like your recording situation. this mixer is ok, but when you rise few knobs, you have to "equilibrate" well the all the knobs levels + the master knob in order not to have some hiss noise sort out..
The trick with this is to use the pre-gains up the top of each channel strip.. turn them up to about 3 oclock. this keeps the noise level much lower than if you use the gain controls at the bottom for each channel to boost it.
thanx man
but normally i tend to don't turn them more than 2 or 3 o'clock...
anyway, until i record without many hiss, for me it's ok.
but, retourning to the audio interfaces, what i want to point here is that if you have a sound card like mine (not bad for its market level) and you want a good audio interface, my advice is not to buy the next level product (like motu 828 series or tc electronics etc, wich are good , don't get me wrong..) but to buy a product from the hi level because the difference between the 2 first levels are not too big.
anyway what audio interface do you use Cartesia?
Re: Making good quality noises!
The mixer definetely doesn't introduce any delay, it's analogue, means it operates at lightspeed (literally). Even if you use its effects, those are just mixed into the original signal additionally.
The delay you desccribe can only come from the audio interface settings. What software do you use for recording? If possible, use one which supports the ASIO drivers of your audio interface. ASIO, unlike DirectX or standard windows MME drivers, use a way around the various software levels in windows, and "talk" much more directly to the interface, allowing MUCH lower latencies. On a reasonably fast PC (Core2Duo upwards) a well made ASIO driver allows latencies below 5 milliseconds. For example, I'm using a buffer size of 64 here, which at 48kHz sampling rate equals 1.3 milliseconds latency. Works like a charm. Also, you should always turn on the automatic latency compensation in the software you're recording with, if possible.
For the leveling problem, this article has some valuable insights:
http://www.bluebearsound.com/articles/levels.htm
And cartesia, I have to disagree: an uncorrected latency of 20-30ms destroys any tight rhythm you might have made. Or at least seriously alters it. Things will just sound off and wrong. If everything you record has that delay (usually the case), the time correllation between different tracks will stay intact, but the whole will have a gap of silence in the beginning. Sucks balls if you're recording loops into Ableton Live, for instance! Again, automatic latency compensation helps.
The delay you desccribe can only come from the audio interface settings. What software do you use for recording? If possible, use one which supports the ASIO drivers of your audio interface. ASIO, unlike DirectX or standard windows MME drivers, use a way around the various software levels in windows, and "talk" much more directly to the interface, allowing MUCH lower latencies. On a reasonably fast PC (Core2Duo upwards) a well made ASIO driver allows latencies below 5 milliseconds. For example, I'm using a buffer size of 64 here, which at 48kHz sampling rate equals 1.3 milliseconds latency. Works like a charm. Also, you should always turn on the automatic latency compensation in the software you're recording with, if possible.
For the leveling problem, this article has some valuable insights:
http://www.bluebearsound.com/articles/levels.htm
And cartesia, I have to disagree: an uncorrected latency of 20-30ms destroys any tight rhythm you might have made. Or at least seriously alters it. Things will just sound off and wrong. If everything you record has that delay (usually the case), the time correllation between different tracks will stay intact, but the whole will have a gap of silence in the beginning. Sucks balls if you're recording loops into Ableton Live, for instance! Again, automatic latency compensation helps.
2x Vestax PDX 2000 + Denon DN-S1000 + Numark 5000FX
Re: Making good quality noises!
LoneStar wrote:The mixer definetely doesn't introduce any delay, it's analogue, means it operates at lightspeed (literally). Even if you use its effects, those are just mixed into the original signal additionally.
The delay you desccribe can only come from the audio interface settings.
exact.
LoneStar wrote:
For the leveling problem, this article has some valuable insights:
http://www.bluebearsound.com/articles/levels.htm
thanx for this, i have to read it.
Re: Making good quality noises!
ellaguru wrote:anyway what audio interface do you use Cartesia?
I used a presonus firebox (dont use computer at all for music at the moment)
anyone looking to buy an interface I'd recommend it.. high quality build, good latency, good sound - although at this point in time it might be best to wait for USB 3 because firewire is on its deathbed - but still.... you can pick them up really cheap now (about $160 on ebay) - I paid $500 for mine when I got it 2 years ago.
Samples.