"Change Tempo" is also increasing volume

[steve]

the input waveform had no clipping whatsoever

I disagree. The input waveform has much clipping, but it is intentional clipping - an important part of the "over-driven" electric guitar sound. This is a close-up of the input waveform that you provided, and the "clipping" (flattened peaks) is very clearly visible:

[eheilner]

This is a very late response, but I replicated the behavior reported by CraigG58 that "Change Tempo" is increasing my volume. In my case I was decreasing the tempo by 6.1% and my peak volume increased roughly 2.5 DB - which was causing clipping (my original 24 bit 88.2K recording was mixed very "hot").

The good news is that I also replicated the behavior reported by DVDdoug - in that I could amplify by -3 DB and the clipping disappeared - my nice "rounded" waveform was restored. Those extra 8 bits are quite handy.

[eheilner]

My previous post was overly optimistic. The increased amplitude caused by change tempo occurs mostly on peaks - so if I amplify the final result by -3 db the total sound is significantly lower.

Apart from the obvious such as doing some peak limiting, does anyone have a better idea?

[eheilner]

Well, my previous post was overly optimistic. It turns out that most of increasing volume - at least on my project - occurred on peaks. I.e., existing peaks in the original were accentuated. So simply reducing the gain by -3 DB has the effect of noticeably reducing the overall volume. Not good.

Apart from the obvious - such as some peak limiting - does anyone have a better idea?

To answer the obvious question - the instrumentation is Viola & Acoustic Piano

[steve]

Well, my previous post was overly optimistic. It turns out that most of increasing volume - at least on my project - occurred on peaks. I.e., existing peaks in the original were accentuated. So simply reducing the gain by -3 DB has the effect of noticeably reducing the overall volume.

The clue is here:
"my original ... recording was mixed very "hot""

The only way to get a really loud ("hot") recording, is to heavily limit the peaks.
If you process audio that is heavily limited, with any effect that has a variable phase delay, then the peak level will increase.
The only way to restore the original loudness while avoiding clipping, is to limit the peaks again.

In short, if you wish to apply an effect that has variable phase delay (such as SBSMS time stretching), to a very "hot" (high level and heavily compressed / limited) audio track, then the way to do it is:

1. Ensure that the track is in 32-bit float format.
2. Apply the effect (in this example "Change Tempo" with "high quality" enabled).
3. Apply a limiter to limit peaks to 0 dB or a tiny bit lower.

[eheilner]

Apologies if this response comes off as a bit hostile, but I don't understand your response. Here was my question:

Apart from the obvious - such as some peak limiting - does anyone have a better idea?

So clearly I did not need any instructions on how to implement peak limiting - and in fact that is exactly what I have done. But perhaps you were attempting to speak to a wider audience - perhaps there are folks who are not familiar with this technology. Or maybe you just wanted to emphasize that this is the only possible approach. If that is the case then I apologize for any perceived hostility.

All that aside, I have not made myself clear. As the OP (and I) have complained - a well designed audio effect algorithm should only do what it is advertised to do - and nothing more. Imagine (if you will) that a Peak Limiter unknowingly boosted certain frequencies by 3 dB. That would be highly undesirable. Yes, you could insert a filter after the fact to undo that - but unless you knew ahead of time that this was going to happen you would be hard pressed to figure out why your recording did not sound like it should.

Now it may be the case that the increased volume is an inherent property of SBSMS, but I could not find this documented anywhere. If you are aware of any such documentation, I would appreciate you pointing me in the right direction.

Getting back to my situation, the fact that my recording was mixed very hot was actually a good thing - since the peaks alerted me to the fact that that something else was going on besides the time stretching. If my recording had been mixed at a lower level I likely would not have noticed anything until mastering - at which point it would have been a mystery where the additional peaks came from.

[DVDdoug]

a well designed audio effect algorithm should only do what it is advertised to do - and nothing more.

Changing tempo without changing pitch (and vice-versa) is a very complex process that's imperfect and it requires trade-offs. And, it's not perfectly-reversible.

[steve]

Or maybe you just wanted to emphasize that this is the only possible approach.

It's the only approach if you want to use SBSMS time stretch (the "high quality" algorithm) with audio that is heavily limited and you want to maintain the same loudness. It's the combination of all three things that make it necessary:

1. an effect that has a variable phase delay
2. heavily limited audio (or any "flat top" audio, such as square waves)
3. maximum loudness

Conversely:

1. If maximum loudness isn't a condition, then no need to limit, just normalize below 0 dB.
2. If the audio isn't heavily compressed, then the problem is unlikely to occur.
3. If you use the "Sound Touch" time stretch algorithm ("high quality" check box not selected), then the problem is unlikely to occur.

[steve]

Now it may be the case that the increased volume is an inherent property of SBSMS

SBSMS does not "increase volume". The "loudness" (ie "volume") of the output is pretty close to the loudness of the input. The "limitation" is due to variable "phase delay". This limitation is common to many effects, notably most IIR filters (such as Audacity's "Low Pass" and "High Pass" filters.

As I illustrated in this post, if there is a frequency dependent phase shift, then although the "sound" may be the same, the "shape" of the waveform will change.

In the special case of "flat top" waveforms, the phase of different frequencies are aligned to produce minimum "amplitude", so if there is a frequency dependent phase shift, then the peak amplitude will increase.

only do what it is advertised to do - and nothing more.

I agree, and that is why the Change Pitch / Tempo effects do NOT apply limiting after SBSMS time stretch. If you want the processed audio to be peak limited, then you can do that manually after applying the time stretch, but Change Pitch/Tempo does not assume that you want that.

[eheilner]

Steve - Thank you for your timely response. I understand and agree with almost everything you said.

That said, I still haven't made myself clear. My issue is with the documentation. If an effect that is labeled to alter X (the time domain) also significantly alters Y (peak level), then that should be documented.

I have put in a request for a documentation enhancement to the SoundTouch libraries. One additional sentence in the description of the SBSMS algorithm would have saved me several hours of time!