Audio quality prediction – AudioVMAF?

tormento · 8th July 2024, 17:30

I was asking myself if something similar to PSNR, SSIM or VMAF exists for audio.

I saw some paper about AudioVMAF but no executable at all to try.

Has anybody some experience of it?

Emulgator · 8th July 2024, 23:09

SNR indeed comes from audio.
It is so simple: Subtraction, then relation.
Encode - Source = Error signal.
Relate this to FS level and there it is: SNR

To judge what these -46dB from a cassette tape or video encode might mean:
It is a 200th of FS signal. Just amplify that error signal and listen to it.
This tells more about signal degradation than any artificial unit.
All is there, ready to be analysed: kind of distortion, frequency components of distortion, temporal distortion, noise, modulation products...

SSIM, VMAF are rather concealing those clear and ugly numbers, like:
"How much dirt can we hide under the carpet until 90% of customers won't notice anymore ?"
Well, then it becomes subjective. Good for content distributors, bad for archival.

PSNR is still my first go at video quality, but this is engineer's side of things.

tormento · 9th July 2024, 14:21

Quote:

Originally Posted by Emulgator

Encode - Source = Error signal

Can you suggest me what program(s) do you use?

Emulgator · 9th July 2024, 18:34

Any audio editor will do.
CoolEdit, Audacity, SoundForge, Reaper, Audition, ProTools,..
Subtract, or Invert&Mix.

P.S. And you will want to work in 32bit float, so specify "open as 32-bit" (wording depends on Editor used)

j7n · 28th July 2024, 02:38

You need to carefully align the two files in time using a transient as a point of reference. Switch the time ruler to samples. Sometimes the latency is a fractional number of samples. There was a tool called DeltaWave that aimed at doing the alignment automatically. Most lossy codecs have some delay that is only compensated with the most popular formats. You can distil the difference to a single number by calculating the RMS power, possibly with Equal Loudness Contour in Sound Forge, so that high frequencies are deweighted.

8th July 2024, 17:30	#1 \| Link
tormento Acid fr0g Join Date: May 2002 Location: Italy Posts: 2,667	Audio quality prediction – AudioVMAF? I was asking myself if something similar to PSNR, SSIM or VMAF exists for audio. I saw some paper about AudioVMAF but no executable at all to try. Has anybody some experience of it? __________________ @turment on Telegram

8th July 2024, 23:09	#2 \| Link
Emulgator Big Bit Savings Now ! Join Date: Feb 2007 Location: close to the wall Posts: 1,657	SNR indeed comes from audio. It is so simple: Subtraction, then relation. Encode - Source = Error signal. Relate this to FS level and there it is: SNR To judge what these -46dB from a cassette tape or video encode might mean: It is a 200th of FS signal. Just amplify that error signal and listen to it. This tells more about signal degradation than any artificial unit. All is there, ready to be analysed: kind of distortion, frequency components of distortion, temporal distortion, noise, modulation products... SSIM, VMAF are rather concealing those clear and ugly numbers, like: "How much dirt can we hide under the carpet until 90% of customers won't notice anymore ?" Well, then it becomes subjective. Good for content distributors, bad for archival. PSNR is still my first go at video quality, but this is engineer's side of things. __________________ "To bypass shortcuts and find suffering...is called QUALity" (Die toten Augen von Friedrichshain) "Data reduction ? Yep, Sir. We're that issue working on. Synce invntoin uf lingöage..." Last edited by Emulgator; 9th July 2024 at 18:40.

9th July 2024, 18:34	#4 \| Link
Emulgator Big Bit Savings Now ! Join Date: Feb 2007 Location: close to the wall Posts: 1,657	Any audio editor will do. CoolEdit, Audacity, SoundForge, Reaper, Audition, ProTools,.. Subtract, or Invert&Mix. P.S. And you will want to work in 32bit float, so specify "open as 32-bit" (wording depends on Editor used) __________________ "To bypass shortcuts and find suffering...is called QUALity" (Die toten Augen von Friedrichshain) "Data reduction ? Yep, Sir. We're that issue working on. Synce invntoin uf lingöage..." Last edited by Emulgator; 9th July 2024 at 18:38.

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28th July 2024, 02:38	#5 \| Link
j7n Registered User Join Date: Apr 2006 Posts: 148	You need to carefully align the two files in time using a transient as a point of reference. Switch the time ruler to samples. Sometimes the latency is a fractional number of samples. There was a tool called DeltaWave that aimed at doing the alignment automatically. Most lossy codecs have some delay that is only compensated with the most popular formats. You can distil the difference to a single number by calculating the RMS power, possibly with Equal Loudness Contour in Sound Forge, so that high frequencies are deweighted.