Distorted audio recording from line in

[AusIV]

Sorry to double post this in multiple boards, but nobody answered in the recording board so I thought I might have more luck in the OS specific board.

I'm trying to record through my line in, and I get a high pitched ringing and an incomplete range of sounds. The audio sounds fine if I unmute the line in, so I know a good signal is coming in. Additionally, I can use MythTV to record from that Line in instead of the one associated with my tuner, and the sound recrods fine associated with a TV show, so I know it can be recorded properly.

I've tried a variety of settings, but I haven't gotten anything to sound anywhere near right.

What settings are recommended for getting high quality sound?

[richardash1981]

What project rate (sample rate) are you recording at in audacity?
What you want is to use the native sample rate of your hardware, which is most likely to be 44.1 or 48k in consumer hardware.

[prost]

I use Mandriva & have a problem recording from line-in, so I use Mandriva's basic sound recorder. maybe we have to wait for v1.3 stable to come out - it'll be alsa not oss.

[terryxela]

For the last 2 day I have been trying to solve the same problem. I got a nice player to play vinyl and digitalized the tracks. The problem was that ringing sound in the background. I tried other sound sources including amorok internet radio and the same problem.
I removed the tv cards that I use for myth following a user's tip and no result.
I used Audacity and also the krecord with same results= high pitch background sound. Then I tried the gnome recording application and everything recorded fine so I realized the problem was in my end.
Well as usually the solution was simple and it was to change the sampling rate. I was using the standard 44.1 and I went to 48 and everything is OK. Because 44.1 is a common used sampling rate I did not think it was a problem.
I hope it help you.

-=terry(Denver)=-

[richardash1981]

So in your case it was a lousy re-sampling algorithm in ALSA causing the problem. Given that you are the first person to to actually define what you mean by "distortion" (and it turns out not to be anything to do with either Total Harmonic Distortion or Inter-modulation Distortion), this may or may nor apply to any of the other posters.

It's possible to configure ALSA to use libsamplerate for sample rate conversion, which should fix the issue, but I can't now locate the link describing how.

[terryxela]

Well I though I posted this message but I do not see it so I will try to rephrase it

Richard my knowledge of sound and distortion is very limited. My approach was to change one variable at the time and see the results. Kind of brute force approach.

>It's possible to configure ALSA to use libsamplerate for sample rate conversion, which should fix the issue, but I can't now locate the
> link describing how.

I am not sure what do you mean. Do you say that you can convert the "corrupted" files? I do not need it because I rm them. Do you say that you can fix Alsa so I can use the 44.1 without problems?. Well I will look for that thread.

Thxs for you reply.

-=terrry(Denver)=-

[richardash1981]

Distortion normally has a very specific meaning to do with the gain of a circuit depending on the instantaneous amplitude of the signal. This is what is going on with a Distortion effect on a guitar, the terrible sound you hear if the recording level is set too high, or the distortion figure quoted for a power amplifier. The artifacts produced by poor resampling are different, and involve the shifting in time of high-frequency sounds (ringing) and creation of tones not in the original (artifacts). The causes of these are completely different, so to just say "it's distorted" to mean it doesn't sound as good as the original is misleading. My main point is that although this fixed the issue for you, it's wouldn't help someone with the gain set too high and causing clipping one little bit, so saying that changing the sample rate to 48kHz solves the problem is overstating the case somewhat. It's likely to help people with cheap, nasty on-board sound, most of which is locked to 48kHz, but not anyone with an older SoundBlaster card (because they are locked at 44.1kHz).

And yes, I mean you can configure ALSA so that recording at 44.1kHz sounds much better than it does by default if you change the re sampling algorithm.

[terryxela]

Richard,

Thank you for all the info (BTW I save it). I installed libsamplerate. I will work with it sometime this week.

Regards,

-=terry=-

[terryxela]

Richard,

I installed libsample rate and now I can record in 44100 with perfect sound. Just for my own info which lib this libsamplerate replace? I read all the doc that are in the dev package and I kind of get it but not all the way.
Thxs again for your great help and have a Happy Thanksgiving.

-=terry=-

[john82]

Richard my knowledge of sound and distortion is very limited. My approach was to change one variable at the time and see the results. Kind of brute force approach.

-=terrry(Denver)=-

actually i find that this is the best approach to solve any kind of problem, nothing brute about it.
it also works %99 of the time, that is if the problem is included in one of the variables which you intend to change.
Johnny