Audacity Forum

Some may remember I was asking a load of questions a while back trying to get Audacity set up for vinyl ripping.

Nearly 40gb later I can report that I am delighted with the quality; it is really very, very good indeed. But it is in further post processing I am getting poor results, using the same workflow I successfully used on my first ~150gb of rips 2001-2008. I wonder if anybody can help me to pin down where I am going wrong?

What I am doing is removing clicks in Audacity and saving 16bit .wav files ripped via an Asus Xonar D2 soundcard.

I am then importing these into a program called Steinberg Clean, that allows you to apply various noise removal filters, EQs etc, and save the output. I realise others use different tools for vinyl clean-up, and am always open to suggestions, but Clean suits me well as it is simple and effective and has always served me well.

Except now.

What I am getting is heavily distorted bass, using approximately the same EQs I have always used.

I believe what is occurring is that my new Asus soundcard is much more sensitive and capturing lower frequencies compared to my old Soundblaster card, and that Clean is somehow distorting these into the audible spectrum, when I tweak the low frequency EQ (even applying negative gain appears to produce distortion).

I suspect the solution may be to apply an EQ cut off in Audacity (hopefully via a chain) but would like to quantify settings.

Hoping somebody can help me to sort this out. . .

Do the files have distorted bass when you have first recorded them, or only after processing with "Clean"?

Steve,

Many thanks for your reply.

I have been on holiday but having researched my problem further had started to believe there were in any case too many variables to be able to pin down a solution; I really appreciate you trying to help me out.

To answer your question yes, on the most heavily affected tracks there is in fact some distortion prior to processing in Clean, which then accentuates this.

It also seems clear from having looked at a number of frequency distributions that the issue almost certainly arises in my Pro-ject pre amp instead of the Asus soundcard (I had previously been using a Vestax PMC 40).

Attached is a sample from one such track (I can also post an audio clip if that helps)

Does this give you any ideas?

Shaky wrote:Does this give you any ideas?

Not really, though you could try applying a 20 or 30 Hz high pass filter as the first step in your processing. Sounds below 20 Hz cannot be played by most audio equipment.

That’s one of the things I am a little confused about. For example, digging out the spec on my old but still very good Vestax PMC 40 mixer, I see it supposedly has a frequency response of 20-20,000Hz. But looking at the spectrum of .wav file recorded with this in Audacity I see there is plenty going on under 20Hz, also right down to sub 3Hz as per the spectrum using my current setup posted above. Where does this come (through?) from?

Furthermore, many of my old .wav files also have a hump in the spectrum around 12-15Hz, which I suspect is the area causing me problems, but these max out at around -48db. As with frequency is there rule of thumb regarding what level of db is audible?

FWIW I have been trying to trim this stuff with the Audacity EQ and have used two presets depending on the magnitude of that hump. I have also tried to experiment with the high pass filter but am having trouble configuring the parameters, especially the roll off. Do you have any suggestions?

And coming back full circle to the business about audio equipment frequency response under 20Hz, I nevertheless believe - admittedly subjectively - that many of my old vinyl rips sound better than CDs. I am therefore reluctant to cut off more than necessary of the low end frequencies, if that is the reason. . .

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-- Wikipedia --
IEC RIAA curve
An improved version of the replay curve (but not the recording curve) was proposed to the International Electrotechnical Commission with an extra high-pass filter at 20 Hz (7950 µs). The justification was that DC coupling was becoming more common, which meant that turntable rumble would become a greater problem. However, the proposal did not achieve traction, as manufacturers considered that turntables, arm and cartridge combinations should be of sufficient quality for the problem not to arise.

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20 Hz is jet engines, earthquakes, or that one low organ note that you can't hear so much as feel. Chances of those musical notes making it onto and off the vinyl record are zero. There were reports of people trying and producing vinyl records that nobody could track without putting a penny on the headshell. The stylus would literally leave the groove from the abuse.

So you probably have award-winning motor rumble and should get rid of it. As in the article, Audacity will cheerfully "edit" battery DC voltages and yes, it causes no end of problems. High pass filter at 30Hz and let it droop to the left, probably either 12dB or 24dB per octave.

Another real world note. You can very nearly not hear a 6dB change. It's very gentle. That's from 1 to 0.5 on the Audacity default timeline.

Koz

"DC" stories abound. When I lived back east (US), one of the engineers fascinated with his new Crown DC-300 power amplifier decided to do a real-world test. He connected a 1.5v penlight battery to the input of the dc-coupled amplifier and amplified it to 6v making it the most expensive lantern battery on earth.

http://en.wikipedia.org/wiki/Lantern_battery

Another is the thought experiment of what would happen if you actually had a sound system that would go down to DC. You wouldn't need a fan or a vacuum because DC sound is wind. You could use your speakers to blow-dry your hair.

That gives you a colorful idea of what kind of trouble you can get into with a sound system that goes down too low.

Put that kind of audio into a powerful conventional sound system without warning everybody and you'll be picking up the woofer cones from the other side of the room -- and you won't be able to hear for a while.

Koz

Thanks very much Koz, lot’s of very useful background there. I have one question on this: in a big club PA what is the low frequency that makes your chest almost vibrate/resonate?

As regards the high pass filter, many thanks for the tip on the cut off and roll off. Armed with this I went back and tried to apply that to one of my distorted files, but immediately realised why I originally abandoned that approach.

It should be apparent that I have little formal understanding of audio engineering, but I do have a little experience of digital signal processing from another field; it is rightly said that a little experience is a dangerous thing, but what concerns me applying the high pass filter is that it seems to raise some of the peaks in the waveform.

What I mean by this is that looking at say a 5 minute track in a compressed window with a flatish ceiling, following the application of the high pass filter this will become visibly jagged with the new peaks exceeding the old ones.

As I understand it a band pass filter should only subtract and never add to a signal, therefore I can not explain this behaviour and am distrustful of the result (in contrast the EQ works as I would expect).

Is there something I am missing such as the right filter quality q setting (BTW how should that be determined???) or is there something else going on I am missing?

Again many thanks to everybody for the wonderful help.

Shaky wrote: in a big club PA what is the low frequency that makes your chest almost vibrate/resonate?

Big club PA systems use (powerful) "bass bins" to produce frequencies down to about 30 Hz, or perhaps a little lower.
For example, the (big) JBL SRX728S dual 18" speaker is specified: -10 dB at 27 Hz.

The low/high pass filters work in a similar way to (hardware) analogue filters.

An attempt at an over-simplified explanation:
As the corner frequency is approached there is a gradual phase shift. The result of this is that the peak of a particular frequency components my be in a slightly different place from where it was originally. This is the normal behaviour for this type of filter. Depending on where the peaks of specific frequency components occur, the combined amplitude of all frequencies at a particular point in time may be higher or lower than the original level.
The Equalizer effect uses a different type of filter (an FFT filter) that does not shift the phase, so the peak level will remain as expected.
Within the audio range, both types of filters produce sonically good results, (though FFT filters are unable to produce steep filter slopes at very low frequencies without an impractically large FFT size).