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-Obscurity-
09-20-2005, 11:21 AM
I am using the default mic input in my Dell computer to record into Cakewalk, and for some reason everything I record is unbalanced slightly to the left. If I switch to the line-in input instead of the mic input, everything is recorded slightly to the right. I have checked the balance on the volume control, nothing is panned anywhere on the mixer, computer, or software, and all my cables and adapters are working fine as well. After laying down a few tracks, it peaks on the side that is being recorded slightly louder because of the accumulative volume buildup. Could it be that my default sound recorder is wired slightly off balance, or is there something else I can do? It is strange how the mic input is off balance to one side and the line-in input is off balance to the other. I am using Windows XP if that matters. Like always, there will be rep to those who can help.

Lateksi
09-20-2005, 11:35 AM
Really weird.
I'd say that your soundcard is just broken, or just extremely bad quality, but if the soundcard is just an integrated one, that really doesn't surprise me.

-Obscurity-
09-20-2005, 11:39 AM
Really weird.
I'd say that your soundcard is just broken, or just extremely bad quality, but if the soundcard is just an integrated one, that really doesn't surprise me.

Yes, it is an integrated one, is that common for this to happen?

shayne_122
09-21-2005, 12:55 AM
Yes, it is an integrated one, is that common for this to happen?


Not as much common as it is more plausible than a high quality soundcard.

So I guess you'll just have to compensate for the panning on your soundcard with your mixer. (turn the pan knob to the right)

Seafroggys
09-21-2005, 01:45 AM
Or buy a soundblaster live. I think people still sell them at about $20-30. I got mine for $100 4 years ago, and it still kicks ***. Unlike gfx cards and processors, sound cards can age well.

-Obscurity-
09-21-2005, 10:47 AM
Not as much common as it is more plausible than a high quality soundcard.

So I guess you'll just have to compensate for the panning on your soundcard with your mixer. (turn the pan knob to the right)

Well, I assume there would still be some signal loss due to the algorithms used in processing the sound through the computer. Even though it is panned in the mixer, it would still be recieved a little "off" through the soundcard. It would be a weaker signal, and it would be impossible to get it centered with 100% accuracy. I think I am just going to buy an M-audio 24/96 card and throw this one out the window.

Lateksi
09-21-2005, 12:52 PM
Getting a semi-pro soundcard is really good idea if you want to make decent recordings.
I got M-Audio Delta Audiophile 2496 card couple of weeks ago, and I'm very satisfied with. I really recommend you to get it.

-Obscurity-
09-21-2005, 01:02 PM
Getting a semi-pro soundcard is really good idea if you want to make decent recordings.
I got M-Audio Delta Audiophile 2496 card couple of weeks ago, and I'm very satisfied with. I really recommend you to get it.

I always wondered how much of a difference it actually makes from the standard 1/8" input soundcard. Is the quality noticeably different?

Lateksi
09-21-2005, 02:16 PM
The quality is really noticeably different. First differency I noticed was the absence of noise while recording, and then, with further analysis bigger sample rate and bit-depth also make big difference. And of course minimal latency, I can use software effects to alter the guitar sound in real time. My V-amp2 sounds much better when I use eq and BBE Sonic Maximizer vst's on computer while playing with it.
Also AD- and DA-converters are so good, that even existing music files sound better with this card.

-Obscurity-
09-21-2005, 02:32 PM
The quality is really noticeably different. First differency I noticed was the absence of noise while recording, and then, with further analysis bigger sample rate and bit-depth also make big difference. And of course minimal latency, I can use software effects to alter the guitar sound in real time. My V-amp2 sounds much better when I use eq and BBE Sonic Maximizer vst's on computer while playing with it.
Also AD- and DA-converters are so good, that even existing music files sound better with this card.

Can you tell me anything I should know about sample rate and bit-depth? Does recording in 24 bit really help even though mixing for a cd requires 16 bit? What exactly does bit-rate effect, and what bit rate should I use?

Lateksi
09-22-2005, 11:51 AM
Well...
Bit-depth tells how many bits one sample has, that is how precisely the value of signal is saved. (16 bit is 65536 possible values and 24 bit is 16777216 possible values, but don't look at the numbers too closely, hearing the difference is much harder...). I'd record everything with 24 bits and mix the recording into one 24 bit-depth file and convert that to 16 bit file with some quality program that uses good dithering and such like. At least in theory, quality should be better than using 16 bit recordings.
About sample rate, converting one sample rate to another often degrades quality if you don't know what you are doing. CD's have sample rate of 44.1kHz, but recording with higher rate has some advantages (better sound quality and lower latency), but what sample rate to use? 96kHz is kind of a standard, but converting it to 44.1kHz with bad program could create some artifacts and distortion, that recording the whole thing with 44.1kHz in the beginning would have been better.
I have decided to use 88.2kHz sample rate on my recordings, because when converting it to 44.1kHz doesn't cause any artifacts to be created. Reason to this is that when you divide 88.2 with 44.1 you get 2, so every other sample is discarded, but dividing 96 with 44.1 you get 2.17687... which isn't an integer, so discarding samples is more complex and causes (extremely) mild distortion (of course oversampling prevents much of that, but still...).

Conclusion, this is too hi-fi'st to be practical, so use what you think is most comfortable for you, because 95% of people can't hear any difference with their normal speaker between 16bit/44.kHz and 32bit/88.2kHz recordings...
And really, fine tuning sample rates and bit-depths don't help anything if your other recording equipment aren't even bearable quality.





Ok... who understood anything of what I wrote? :D

-Obscurity-
09-22-2005, 11:58 AM
Well...
Bit-depth tells how many bits one sample has, that is how precisely the value of signal is saved. (16 bit is 65536 possible values and 24 bit is 16777216 possible values, but don't look at the numbers too closely, hearing the difference is much harder...). I'd record everything with 24 bits and mix the recording into one 24 bit-depth file and convert that to 16 bit file with some quality program that uses good dithering and such like. At least in theory, quality should be better than using 16 bit recordings.
About sample rate, converting one sample rate to another often degrades quality if you don't know what you are doing. CD's have sample rate of 44.1kHz, but recording with higher rate has some advantages (better sound quality and lower latency), but what sample rate to use? 96kHz is kind of a standard, but converting it to 44.1kHz with bad program could create some artifacts and distortion, that recording the whole thing with 44.1kHz in the beginning would have been better.
I have decided to use 88.2kHz sample rate on my recordings, because when converting it to 44.1kHz doesn't cause any artifacts to be created. Reason to this is that when you divide 88.2 with 44.1 you get 2, so every other sample is discarded, but dividing 96 with 44.1 you get 2.17687... which isn't an integer, so discarding samples is more complex and causes (extremely) mild distortion (of course oversampling prevents much of that, but still...).

Conclusion, this is too hi-fi'st to be practical, so use what you think is most comfortable for you, because 95% of people can't hear any difference with their normal speaker between 16bit/44.kHz and 32bit/88.2kHz recordings...
And really, fine tuning sample rates and bit-depths don't help anything if your other recording equipment aren't even bearable quality.





Ok... who understood anything of what I wrote? :D

I understood it, and repped you for your helpfulness. I had one question though. I have decided on sticking with 16-bit due to CPU limitations, but how much of a noticeable difference in quality will I gain from recording at 88.2 kHz vs. 44.1 kHz? Is it worth it?

Lateksi
09-22-2005, 12:13 PM
I'm not sure, you just have to try both and decide for yourself. Lower latency is bigger reason for me to use 88.2kHz than better sound quality.

-Obscurity-
09-22-2005, 12:55 PM
I'm not sure, you just have to try both and decide for yourself. Lower latency is bigger reason for me to use 88.2kHz than better sound quality.

What exactly is latency?

Lateksi
09-22-2005, 01:37 PM
Latency is the time between the moment when signal goes into the soundcards input and the moment when it comes out from its output.
Normal soundcards have zero monitoring latency, when signal that goes into the soundcard comes out with zero latency. Better soundcards, the ones with ASIO-driver support, can route the signal through software and the to output within only few milliseconds (with soundblasters it can take over one second). Advantage is nice, you can e.g. connect guitar to soundcard through di-box to get only clean signal to software, use some amp modeler (like Guitar Rig) and hear the modeled sound in real time.

-Obscurity-
09-22-2005, 05:18 PM
Latency is the time between the moment when signal goes into the soundcards input and the moment when it comes out from its output.
Normal soundcards have zero monitoring latency, when signal that goes into the soundcard comes out with zero latency. Better soundcards, the ones with ASIO-driver support, can route the signal through software and the to output within only few milliseconds (with soundblasters it can take over one second). Advantage is nice, you can e.g. connect guitar to soundcard through di-box to get only clean signal to software, use some amp modeler (like Guitar Rig) and hear the modeled sound in real time.

Ok, I think I get it. I don't think I have to worry too much about that, I do all my guitar processing either before recording, or later during the mixing process. So this really wouldn't effect me right?

Lateksi
09-22-2005, 05:53 PM
Right. But possibility for low latency is never a bad thing.

-Obscurity-
09-22-2005, 05:57 PM
Right. But possibility for low latency is never a bad thing.

True. Thanks for all your help, you really explained alot. I ordered the M-audio 24/96 card and a beta 58 from Musicians friend yesterday, so my recordings are about to get a ot better. :thumb:

shayne_122
09-24-2005, 02:18 PM
24/96 card and a beta 58

Uh oh, heh heh.

A 57 is better for recordings, but oh well.

Liam Gaughan
09-25-2005, 11:44 AM
try recording mono?