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Dithering is a Mathematical Process - NOT a psychoacoustic process.

I've lost track of this thread... Regarding intersample amplitude, I have seen 3-6 dB over full-scale fairly often for full-scale, or near full-scale (IEEE testing is typically at -1 dBFS) input signals, and 6-12 dB or more inside the digital filter structures subject to how well the filter gain structure was optimized. Before I had the grad class, and again after I forgot and shot myself in the foot again, I could easily hit 20 dB "over" in the digital filters. Which is quite vexing when you do all the simulations using 64-bit floating-point math (where things are fine) and then dump the coefficients into a 16- or 32-bit DSP/MCU in the actual implementation. :facepalm:
And therefore this article:
https://techblog.izotope.com/2015/08/24/true-peak-detection/

Most threads related to mastering/mixing habit/procedure in this and other audio forums are somehow derailed by intersample over, loudness war, "true" peak, LUFS, DR meter and so on. The term "true" peak is misleading anyway, different hardware/software filters operate differently, different true peak meters show different true peaks values.

As long as the source is not clipped at fixed point intersample level, it is not clipped. ISP and codec clipping should be addressed during playback/broadcasting, not during mastering.
 
And therefore this article:
https://techblog.izotope.com/2015/08/24/true-peak-detection/

As long as the source is not clipped at fixed point intersample level, it is not clipped. ISP and codec clipping should be addressed during playback/broadcasting, not during mastering.

True if at the dac no upsampling is done. As soon as you interpolate/upsample new higher peak values can occur.
If these are not handled properly this could lead to real clipping. But I agree that every DAC should take care of it.
If this the case you can indeed say: "As long as the source is not clipped at fixed point intersample level, it is not clipped"
 
True if at the dac no upsampling is done. As soon as you interpolate/upsample new higher peak values can occur.
If these are not handled properly this could lead to real clipping. But I agree that every DAC should take care of it.
If this the case you can indeed say: "As long as the source is not clipped at fixed point intersample level, it is not clipped"
"As long as the source is not clipped at fixed point intersample level, it is not clipped" refers to those who are willing to use digital volume control/normalization during playback. I wrote a very long article about it:

https://archimago.blogspot.com/2019/06/guest-post-why-we-should-use-software.html
 
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The decoding bit-depth can be configured.
View attachment 49896

Algorithms in modern lossy codecs don't have a fixed bit-depth. In fact, if you insist to use fixed-point, there will be risks of clipping during decoding, even if the input file is 16-bit.
https://izotope-rx.livejournal.com/5760.html
https://forum.cockos.com/showthread.php?p=2001665#post2001665

Indeed. AAC may be able to do 128kb/s and sound good for a stereo pair, but don't try to calculate its bit rate by the levels used in a PCM file. The mapping is very different.
 
As long as the source is not clipped at fixed point intersample level, it is not clipped. ISP and codec clipping should be addressed during playback/broadcasting, not during mastering.

No please. do not insert ISP's. Many, many DAC's don't handle that right. Yes, I have measurements, including some very disturbing ones.
 
Interestingly enough, CoolEdit Pro shows the proper real bit depth of that f...ing mp3 file, which is 16-bit.

How many bits per sample are available with lossy coding? Even at (stereo) 320kbps, ~4.3 bits/sample.

Of course, even lossless entropy code would reduce the file size. And, "joint stereo" can be used for stereo.

Using the "Codec Preview" facility provided by iZotope Ozone, set to AAC @ 320kbps (I think it uses joint stereo for all bit-rates; no setting is available), the input and output looks in Bitter looks like this:

1581558861493.png


Where the source (16-bit music file from a CD) is on the left, the encoded/decoded output from iZotope is on the right. (Both are the left channel, actually, although the codec is being fed both left and right channels.) Indeed, it returns 32-bit float.

Conveniently, iZotope has a "Solo Codec Artifacts" feature. So, comparing the spectrum of the source and the "codec artifacts":

(Left channel only of source and "codec artifacts," BLUE=SOURCE, PURPLE="CODEC ARTIFACTS.")

1581559027430.png


Hmm, hardly "16 bit" performance, is it? (Although perceptually is another story...)

(N.B. The spectra were averaged over 10 seconds, so this doesn't properly reflect the dynamic allocation of resources (bits.)

By way of comparison, here is the source compared to the difference between the source and truncation to 4 bits:

1581561780132.png


And compared to the difference between the source and truncation to 6 bits:

1581561836882.png



BTW, the iZotope AAC codec (at 320kbps) can code a 1kHz sine down to an amplitude slightly under -160dBFS before "giving up":

(Output after coding/decoding, i.e. NOT the "codec artifacts" output option.)

1581559686927.png



Here is a multiband compressor that has a maximum bit-depth setting for its output of 24 bits (varous dither/noise-shaping modes possible, or if turned off then it will simply truncate):

1581560076466.png


Here is another multiband compressor that has a 16 bits output option:

1581560109647.png


As such, Bitter is unreliable, because no one knows what processing was used with the recording. I do not care about computed data that carry no useful new information.

Right tool for the right job.

Bitter is a bit meter. It ONLY reads the activity of the bits that it receives as input; or in other words, it is used to measure bit activity output by a process.

That is why I offered the example of a old EQ plug-in (32-bit float internally) vs. a newer EQ plug-in (64-bit float internally.) Both return 32-bit float to the host software, so nothing different to note in Bitter; but looking at the spectral plots, one is a mess and the other is clean (given truncation to 32-bit float from 64-bit float.)

IOW, Bitter won't tell you if your signal has been "screwed up," audibly or otherwise.

I can think of dozens of possible instances where being able to get a quick bit meter reading of what's being output from a process is very useful indeed. Obviously, it is only one tool in the box.

For example, I have not used foobar in a while, but I imagine that you could use Bitter after resampling to check whether the resampling process used outputs float or fixed-point.

As for your CoolEdit Pro result of "16 bits" for the "Actual Bit Depth," how is it coming up with this figure? Does it mean that the MP3 was decoded to 16-bit fixed point? (I don't have Adobe Audition installed, let alone CoolEdit Pro)
 
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https://books.google.com/books?id=a...wB3oECAgQAQ#v=onepage&q=13 db miracle&f=false

So a very low rate MP3 can still have 24 active bits, even if it only sends 1.3 bits/sample. Do not try to introduce perceptual coding into this, it's a very different issue.

Fact: MP3, AAC, and other codecs have much more than 24 bits of dynamic range as configured. There's a world of difference between instantaneous SNR and dynamic range.
 
No please. do not insert ISP's. Many, many DAC's don't handle that right. Yes, I have measurements, including some very disturbing ones.

Hence "dBTP" measurements have been standardised, promulgated and implemented so that this (hopefully) does NOT happen at mastering, or broadcast limiters being "blind" to ISP's, etc.

c.f. The old practice of "normalisation" (predicated on false ideas about "resolution.")

Care to share any of these stories of disturbing measurements? :)
 
So a very low rate MP3 can still have 24 active bits, even if it only sends 1.3 bits/sample. Do not try to introduce perceptual coding into this, it's a very different issue.

Fact: MP3, AAC, and other codecs have much more than 24 bits of dynamic range as configured. There's a world of difference between instantaneous SNR and dynamic range.

Thanks JJ, hole in one c.f. my incompetent waffling on. :)
 
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There's a world of difference between instantaneous SNR and dynamic range.

This is a tough one to explain to the general user. One could point out that for much music the entropy of the data allows about a 3x lossless reduction so you only have 8 bits per sample for full 24 bit audio.
 
This is a tough one to explain to the general user. One could point out that for much music the entropy of the data allows about a 3x lossless reduction so you only have 8 bits per sample for full 24 bit audio.

The old "NICAM" (Near-Instantaneous Companded Audio Multiplexing) system that was used to add digital audio to analogue TV broadcasts (in parts of the world) was an early lossy coding system. Easier to understand than perceptual coding...

1581564533356.png


[Source: https://www.etsi.org/deliver/etsi_en/300100_300199/300163/01.02.01_60/en_300163v010201p.pdf]


(EDIT: Hmm, unless you meant "entropy" rather than "perceptual entropy?" 3x would seem to be rather generous for lossless coding?)

(EDIT (again): D'oh, I suppose floating-point would be an even more straightforward example. Long day, my brain is toast...!)
 
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Care to share any of these stories of disturbing measurements? :)

There's a couple posted in the old thread about bad measurements. I posted some really ridiculous examples of both clipping and ISP's there. I cited it recently in one of these threads, it's old enough it took me 10 minutes to find it and I had to hit "older" I think twice in the search.
 
For an ideal ADC or DAC, SNR goes aas 6N dB (N = number of bits) and SFDR goes as ~9N dB, a big difference. And that does not account for dither, nor the ability of processing systems (radar, radio, our brain, etc.) to extract signals from below the noise floor. See e.g. https://www.audiosciencereview.com/...ital-audio-converters-dacs-fundamentals.1927/

I have to ask what you're measuring as SFDR. I'd say more like 6N+3 myself, but I'm not totally sure of what you're defining there.
 
(EDIT: Hmm, unless you meant "entropy" rather than "perceptual entropy?" 3x would seem to be rather generous for lossless coding?)

He meant entropy as in the part of the signal that's not redundant. That's how lossless codecs work. There are also "nearly lossless" codecs, see my previous post. :) There's a whole world of stuff out there in coding that people do not commonly grok.
 
He meant entropy as in the part of the signal that's not redundant. That's how lossless codecs work.

Indeed. :)

There are also "nearly lossless" codecs, see my previous post. :) There's a whole world of stuff out there in coding that people do not commonly grok.

Nice link. :) Albeit grossly unsuitable reading at this time of night! o_O

I remember fiddling around with ADPCM (IMA?) when I first acquired a sound card. 4:1 compression. It seemed to work pretty well, but then it was being converted to analogue by the questionable D/A converter and out via the extremely dodgy analogue section of some ancient SoundBlaster. ;-)
 
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No please. do not insert ISP's. Many, many DAC's don't handle that right. Yes, I have measurements, including some very disturbing ones.
Hence "dBTP" measurements have been standardised, promulgated and implemented so that this (hopefully) does NOT happen at mastering, or broadcast limiters being "blind" to ISP's, etc.
"standardised" but different "true" peak meters actually show different readings. Therefore confusing, unreilable and misleading. Here are examples:
https://forum.cockos.com/showthread.php?t=193559
https://forum.cockos.com/showpost.php?p=1843676&postcount=89
https://forums.cockos.com/showpost.php?p=1862980&postcount=25
https://techblog.izotope.com/2015/08/24/true-peak-detection/

https://www.masteringthemix.com/blogs/learn/inter-sample-and-true-peak-metering
It’s worth noting that not all true peak meters are created equal. This is because detecting true peaks requires the audio to be converted to a higher sample rate and then low pass filtered to get back an approximation of the original audio. According to the maths (as is ever the case in ‘Digital Signal Processing’) better approximations require more CPU cycles and so different true peak meters will have different accuracy/CPU usage trade-offs.
The irony is that a meter using "better math" has nothing to do with the math used in the target device (e.g. a DAC), so meaningless.

Remember that filters in DACs and SRC and other devices are not "standardised" either, and therefore potential clipping should be addressed during playback, by using digital volume control/management/normalization, not during mastering.
 
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