Clipping without maximum audio level?

[HendrikBoom3]

I'm transcribing old reel-to-reel audio tapes from the 60's, recording them as WAV files on a ZOOM H1 digital recorder, then editing them on audacity.

On one transcription yesterday I got a loud crackling sound that my son, who has experience with other editing software, identified as clipping. So I retranscribed the tape at a lower recording level and the noise went away.

So then I thought -- If there's clipping, it must mean that the waveform reached the limit that can be represented digitally. It should be easy to have a computer program read the recorded file and test whether the waveform ever gets near the limit, so I can tell whether there is likely to have been clipping without me having to listen intensely to the whole thing.

Then I thought -- Doesn't Audacity already read the whole file and show me local-peak intensity in a graph? So go look at the file with Audacity and discover that the intensity in nicely centred about zero, and never even goes near plus or minus 0.5. In fact, it looks as if it maxes out at about 0.25. There are no vertical red lines at all, and yes, I did remember to click on View->Show Clipping. And yes, it was the file with clipping noise I viewed. The file without clipping noise has *much* smaller peaks, staying within plus-or-minus 0.1 on the Audacity graph. And yes, I'm looking at the dark blue part of the graph, not the light blue part.

I'm using Audacity 2.0.3 as installed from the Debian jessie package.

What is likely to be going on here?

When does clipping really occur?

If the limits aren't plus or minus 1.0 on the waveform graph, what are they?

-- hendrik

[steve]

Clipping can occur anywhere along the signal chain.
The "signal chain" is the path from the original sound through to the final recording.

If you try recording a Saturn V rocket taking off with a normal microphone, the microphone will be overloaded and no matter how low you turn down the recording volume the recording will be distorted. The same can occur at any part of the signal chain. It is essential for a good recording that the signal level is within the capability of the equipment at every stage of the recording.

I have a Zoom H2 and it is terrific, except that the recording volume control is misleading, The H2 has a 3 position Hi,Medium,Low recording level switch. It also has an Up/Down digital recording level control. The switch is great - if the sound is loud, set it to low sensitivity - if the sound is quiet set it to high sensitivity. The digital recording level is useless - it just scales the recorded signal. Setting the digital recording level low does not prevent clipping, it just makes clipped audio quieter because the clipping has occurred before the signal got to the digital recording level control. (Does that make sense?)

[HendrikBoom3]

Yes, that makes sense. It hadn't occurred to me that a device as good as the Zoom H1 would have such a misleading recording level indicator or an input level control that operates only after the A-to-D stage.

It doesn't quite match my experience that the clipping disappeared when I turned down the input level control on the H1. But maybe I misremember and did something else at the same time. I'll have to experiment to test your very plausible theory.

Assuming you are correct, and I suspect you are, I'd have to adjust the input level at the source, in this case on my reel-to-reel tape player. Which I can do.

So my next step is to use trial and error to set the digital recording level on the H1 so the clipping occurs just when the digital level indicator indicates it might. Then I'll have the feedback I need to set the output level of the tape recorder.

It does go against the convention of setting the analog output levels high and the input levels low so that any hum picked up en route will be attenuated at the destination.

Thanks. I'll report back.

-- hendrik

[steve]

I've got the H2. The H1 may be different. The H2 has a low/mid/high switch which works on the analogue stage. I normally leave the digital level control set at 100 and just use the switch to control the input level (aiming for too low rather than too high), then use the Amplify effect in Audacity if necessary. The Amplify effect essentially does the same as using the digital level control, but safely.

[kozikowski]

As above, clipping at 0.5 is a signpost that you are recording high level sound on a microphone level connection. This happens to computer soundcards all the time. Newer Windows laptops are missing the High Level input prompting people to try and use Mic-In. Bad move, normally. They get what you have.

My H4 has provision for switching between different levels and inputs. Consult your instructions.

Koz

[HendrikBoom3]

@steve:

I examined carefully the file I had recorded with the input level set very low, and it too exhibits clipping. It's not as noticeable, because the volume of everything is very low, making it about as hard to hear as the music.

Attenuating the analog signal outside the H1 does help. I set the H1's input level to 100, and reduced the volume on my tape deck so as to get the input level display into the -6 to -12 range, and there was no audible clipping. The background hum was loud, though.

It looks as if you are right that the so-called "input level" switches, which the manual says are used to set recording level, do indeed affect the waveform *after* the A-to-D conversion, and therefore have no effect on clipping.

@kozikowski:

There's no documentation about line vs mike input, except the label by the socket on the H1 bearing a picture of a mike, a slash, and the words "LINE IN". Nothing about it in the manual. I've been wondering how it can tell the difference. Maybe some kind of impedance measurement? Or maybe not at all?

@Next things to try:

The device's specs are to do 16-bit sampling, I believe. Audacity tells me 32-bit floating-point. I'm assuming this means that samples are recorded in the file as 32-bit floating-point, there is ample room to have 16-bit precision even after the so-called "input level" setting reduces everything be a factor of a hundred. But of course the H1 could botch this arithmetic, too. Still, assuming I'm deaf enough not to hear the difference between 10-but and 16-bit audio, things could work even if I'm wrong on this. I might be able to tell the difference by looking at the WAV file bit by bitter bit.

So I should set the analog level externally to the highest level that avoids clipping, then set the "input level" low enough to get the resulting audio into a reasonable range. That's probably my best chance to reduce hum. Aside from getting better-shielded cables, of course.

-- hendrik

[steve]

Perhaps a daft question, but I presume that you have the "Auto Level" control turned off?

[HendrikBoom3]

Yes, the auto level control is turned off!!

And Lo Cut is also off.

-- hendrik

[HendrikBoom3]

When I opened a stereo WAV file in Audacity, to the left of the intensity graphs it said "stereo, 44100Hz 32-bit float"

Does that mean that the samples in the file are recorded as 32-bit floating-point numbers? And does that mean that they can be out of normal range without being clipped? (assuming the signal that was stored as 32-bit float wasn't already preclipped, of course.)

I'm wondering, because I've actually played one such file through audacity without hearing any clipping artifacts, even though audacity was showing a lot of red vertical bars on the graph.

Of course, converting this to another format might clip it, not my concern at the moment, though.

-- hendrik

[steve]

When I opened a stereo WAV file in Audacity, to the left of the intensity graphs it said "stereo, 44100Hz 32-bit float"
Does that mean that the samples in the file are recorded as 32-bit floating-point numbers?

By default Audacity converts WAV files to 32 bit float on import. It does not mean that the file was 32 bit before it was imported.
Yes 32 bit float can go beyond 0 dB.
The red "Clipping" lines indicate that the waveform reaches 0 dB or more. If the waveform "just touches" 0 dB it may show a red line which indicates that it is at the absolute maximum for "valid" audio signals.