24bit vs 16bit, the myth exploded!

Mar 12, 2021 at 3:15 PM Post #6,091 of 7,175
Is DSD a horrible outdated meme format, considering it only has 2 quantization levels and 6dB of dynamic range? People claim DSD64 is garbage and noisy and that DSD is only good at 128fs+, but I couldn't tell a difference.
I guess you missed the information theory class then - apart from R2R DACs, pretty much all DACs use a delta-sigma approach, which is to say that they take the PCM and convert it to what is essentially the underpinnings of DSD - an averaging of the 1s and 0s to produce the required amplitude in the signal. Oversampling is what it is.

Now I'm no fan of DSD, as I think it's been marketed as something it's not. However as a digital archive medium it's bloody wonderful. I could program something to pepper a DSD file with 10% of random flipped bits and you probably wouldn't notice the corruption in playback. Do that to a PCM file and it would be unlistenable.

There's nothing really wrong with DSD, it's just that what it is isn't really intended for playback despite what the purveyors of DSD would have you believe.
 
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Mar 13, 2021 at 10:14 AM Post #6,092 of 7,175
I guess you missed the information theory class then - apart from R2R DACs, pretty much all DACs use a delta-sigma approach, which is to say that they take the PCM and convert it to what is essentially the underpinnings of DSD - an averaging of the 1s and 0s to produce the required amplitude in the signal. Oversampling is what it is.

Now I'm no fan of DSD, as I think it's been marketed as something it's not. However as a digital archive medium it's bloody wonderful. I could program something to pepper a DSD file with 10% of random flipped bits and you probably wouldn't notice the corruption in playback. Do that to a PCM file and it would be unlistenable.

There's nothing really wrong with DSD, it's just that what it is isn't really intended for playback despite what the purveyors of DSD would have you believe.
I actually knew the delta-sigma convertion and that all DAC's are basically DSD DAC's which is why I converted my SACDs to DSD and also upsample my PCM to DSD over the air so the DAC doesn't do it.

I heart DSD in SACDs was originally meant to be 4bits at 5.6MHz. If that happened, perhaps it would have been better.
 
Mar 13, 2021 at 6:21 PM Post #6,093 of 7,175
There is no audible difference in sound quality between CDs and SACDs other than the fact that SACDs support multichannel. If you only listen to stereo music DSD is as pointless as teats on a bull hog. It is more complicated and takes up more space for no audible benefit at all.

There is no "better" than audibly transparent. You can pack more frequencies and ultra low volume detail into the sound, but your ears won't be able to hear it. Once you achieve transparency, you can quit. Nothing else you can do will improve the sound, except for listening to better music.

Today, formats have become irrelevant. CD, HD Audio, SACD, DSD, high bitrate MP3 or AAC... It all sounds exactly the same.
 
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Mar 14, 2021 at 7:46 AM Post #6,094 of 7,175
Today, formats have become irrelevant.
That's why I am a fan of good music*, good production, good mixing and good mastering. The relevant stuff...

* Meaning music I personally like and enjoy, but might be crappy music for someone else.
 
Mar 14, 2021 at 5:19 PM Post #6,095 of 7,175
There is no "better" than audibly transparent. You can pack more frequencies and ultra low volume detail into the sound, but your ears won't be able to hear it.

There may be some extreme instances where you can hear the difference between 16 and 24-bit, with some training. Amir of Audio Science Review was apparently able to foil some of the 24-bit vs. 16-bit ABX tests by turning up the volume and carefully scrutinizing some of the quieter musical passages, for example. He explains how in this recent video...

 
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Mar 14, 2021 at 5:32 PM Post #6,096 of 7,175
Another instance where I think it could possibly make a difference is when making adjustments to the content's dynamic range with digital controls. With the volume control on a computer, or using a software-based digital EQ, for example. Increasing the bit depth on my audio device from 16 to 24-bit seems to help with the resampling of the levels in these cases. YMMV though.

If you are not making any digital alterations to the content, and simply listening to it "as is", then there may be no benefit to this.

If you are listening to 24-bit content though, then you probably want to maintain it at that bit-depth, rather than downsampling to 16-bit, which could be potentially lossy.
 
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Mar 14, 2021 at 6:03 PM Post #6,097 of 7,175
Someone was also asking earlier about whether the sample rates made any difference in the high frequency detail that you can hear. And I think the general consensus on that was that most recorded content does not generally go above 20k.

Some humans (ie not me) can hear above that frequency though. And maybe that's why video went for the somewhat wider range offered by 48 kHz.

Under ideal laboratory conditions, humans can hear sound as low as 12 Hz[11] and as high as 28 kHz, though the threshold increases sharply at 15 kHz in adults, corresponding to the last auditory channel of the cochlea.[12]

https://en.wikipedia.org/wiki/Hearing_range#Humans

The highest note that has supposedly ever been sung is G10 at 25.1 kHz. Most music won't go quite that high though. :) For comparison, a typical soprano part rarely goes above "soprano C", which is C6 at a mere 1.0 kHz! Falsetto and coloratura singers can go higher than this though.

Most of the info above a certain frequency in the treble range will generally be overtones and timbral info, I believe, as opposed to actual "notes".
 
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Mar 14, 2021 at 7:07 PM Post #6,098 of 7,175
I don't believe that training can allow you to hear frequencies above 20kHz nor noise floors below -90dB. "Training" is just a way to try to get people to not trust their own ears and take your word for it. It's a power play that has nothing to do with perceptual ability.

There may be some extreme instances where you can hear the difference between 16 and 24-bit, with some training. Amir of Audio Science Review was apparently able to foil some of the 24-bit vs. 16-bit ABX tests by turning up the volume and carefully scrutinizing some of the quieter musical passages, for example. He explains how in this recent video...

I try not to comment on this guy, but when he made the claim to me that he could hear the difference between 16 and 24 bit, I called him on goosing the volume and he refused to admit to me that he was doing gain riding. Cranking the volume on quiet parts isn't some unique skill that takes training. Sure you can hear something down below -90dB if you pump up the volume way up on a fade out. That doesn't take training. That is just plain cheating. No one listens to music riding the volume level to pull up all the quiet parts. Parlor tricks.

Some humans (ie not me) can hear above that frequency though. And maybe that's why video went for the somewhat wider range offered by 48 kHz.

I believe the person who could hear the highest was a child who could hear up to 23-24kHz. That isn't even one whole note on the musical scale above 20kHz- a tiny sliver of sound. By the time the kid was a teenager, she couldn't hear that high any more. No one can hear frequencies as musical pitch much above 10kHz or so. It becomes an undifferentiated squeal and then you just feel sound pressure. If you crank a 30kHz tone loud enough, you can feel it. That isn't hearing.

All this stuff about super audible frequencies is bologna. 16/44.1 is already overkill. The only reason higher data rates are needed is for mixing and mastering. None of the exceptions you mention have any relevance to listening to music in the home. The difference is inaudible in the real world outside the heads of audio "experts".

Video chose 48kHz because it divided evenly with the frame rate of film. It made film to video conversions less difficult.
 
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Mar 14, 2021 at 8:26 PM Post #6,099 of 7,175
I don't believe that training can allow you to hear frequencies above 20kHz nor noise floors below -90dB.

I tend to agree that this is probably the case in normal listening conditions. It would be interesting to see if there has been any testing on this though.

"Training" is just a way to try to get people to not trust their own ears and take your word for it. It's a power play that has nothing to do with perceptual ability.

Perhaps not surprisingly, I would disagree with this. At least when it comes to detecting things like audible noise and distortion, and some types of compression, resampling or aliasing artifacts. I can't give you any demonstrable proof of this though, beyond my own opinion. So my assertion is basically meaningless from a scientific standpoint. :L3000:

I try not to comment on this guy, but when he made the claim to me that he could hear the difference between 16 and 24 bit, I called him on goosing the volume and he refused to admit to me that he was doing gain riding. Cranking the volume on quiet parts isn't some unique skill that takes training. Sure you can hear something down below -90dB if you pump up the volume way up on a fade out. That doesn't take training. That is just plain cheating. No one listens to music riding the volume level to pull up all the quiet parts. Parlor tricks.

Fwiw, he acknowledges much of this in the above video. And it's why I gave it the label "extreme instances". Because this is certainly not the way most people listen to music.

If he were arguing the point with you though, then I think he might point out that "totally transparent in all listening conditions" and "transparent under normal listening conditions" may be two different things. And if you want the former rather than the latter, then maybe some of these smaller details may be worth paying attention to. It's just bits after all, so it's not like a few more is gonna hurt anything. :) And maybe, in some rare or extreme circumstances, it might actually help a little.

If Amir can pick out the differences in quieter recordings at a higher volume, then maybe... just maybe, it's also a possibility for others to detect some noise or quantization errors in some lower volume 16-bit recordings as well. (?)

I believe the person who could hear the highest was a child who could hear up to 23kHz. That isn't even one whole note on the musical scale above 20kHz.

...Which is why I mentioned that sound in that range is mostly overtones and timbral in nature. I certainly can't hear that high though. It would be interesting to see if there have been more studies on this as well though. Perhaps there are certain demographic groups with slightly wider or higher ranges of hearing than others, for example.

Some also claim that they can "feel" the higher frequencies as well. Which is certainly the case with frequencies below 20 Hz. I dunno about the higher frequencies though.

All this stuff about super audible frequencies is bologna. 16/44.1 is already overkill. The only reason higher data rates are needed is for mixing and mastering. None of the exceptions you mention have any relevance to listening to music in the home. The difference is inaudible in the real world outside the heads of audio "experts".

I disagree somewhat with you on this as well. Mainly because of the proliferation of higher-quality tools, digital tools that is, for manipulating and altering audio content more to preference on the user's end. In days of yore, this kind of thing was done mostly in the analog domain.

If higher bit depths and sample rates are beneficial for mixing and mastering content. Then why not also for making similar kinds of adjustments in the digital domain on the user's end? That is my only point (I think).

Video chose 48kHz because it divided evenly with the frame rate of film. It made film to video conversions less difficult.

Yes! Now that you mention it. I seem to recall reading that somewhere as well. Thank you for the correction on that. :thumbsup:
 
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Mar 14, 2021 at 9:29 PM Post #6,100 of 7,175
I try not to comment on this guy, but when he made the claim to me that he could hear the difference between 16 and 24 bit, I called him on it and he refused to admit to me that he was doing gain riding. Cranking the volume on quiet parts isn't some unique skill. Sure you can hear something down below -90dB if you pump up the volume way up on a fade out. That doesn't take training. That is just plain cheating. No one listens to music riding the volume level to pull up all the quiet parts. Parlor tricks.
In his video (around the 19:00-19:40 mark), he said that's exactly what he was doing to pass the test so I don't know why he wouldn't admit that to you.

Another instance where I think it could possibly make a difference is when making adjustments to the content's dynamic range with digital controls. With the volume control on a computer, or using a software-based digital EQ, for example. Increasing the bit depth on my audio device from 16 to 24-bit seems to help with the resampling of the levels in these cases. YMMV though.
Only extremely naive implementations of DSP would make the signal lose bits (unless that was the job of the DSP in the first place). I'm not sure what's the deal with EQ APO but I don't think receiving 24bit samples instead of 16bit samples should help it.

Let's say all the processing that is done to a signal is first reducing the gain by 6dB and then increasing it right back by 6dB.
Reducing the gain by 6dB corresponds to dividing the samples by 2 (12dB would mean dividing by 4 and so on). Dividing binary numbers by 2 is very easy. You just have to shift all the 1-s and 0-s to the "right" (towards the least significant bit).

So for example dividing some 8 bit number by two looks like this: 00110110 (54) divided by 2 is 00011011 (27). Everything just got shifted.
Multiplying is also easy, except the direction of the shift changes: 00110110 multplied by 2 is 01101100 (108).
With that in mind it can be seen how a big gain reduction could make the signal lose bits as the last bits are getting shifted out.

One solution could be just tacking on some zeros to the original number, for example if the input is a bunch of 16bit numbers just add eight 0s to the end of it and that will ensure that the bits won't get shifted out, they just get shifted into the last couple of zeros after the gain reduction. An extra 8bit could allow for 48dB reduction before the signal starts to lose bits. While that might sound good enough, I'm pretty sure most plugins do their calculations on either 32bit float or 64bit floating point numbers which work quite a bit differently but let's just say they'll have enough dynamic range for any kind of sensible processing. If EQ APO doesn't use floats then I can wrap my head around why 24bit might be better for it than 16bits.
 
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Mar 14, 2021 at 9:39 PM Post #6,101 of 7,175
If higher bit depths and sample rates are beneficial for mixing and mastering content. Then why not also for making similar kinds of adjustments in the digital domain on the users end? That is my only point (I think).
Yes it can be beneficial to use more bits during digital signal processing. But for most processing that users would want to do it would be no problem to start with a 16 bit signal, and do the processing in more bits. (That way accumulation of rounding errors that could faul up a few of the least significant bits won't touch the 16 most significant bits.)
[Edit: I didn't see @VNandor 's last post, but he basically says the same in other words.]
 
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Mar 14, 2021 at 10:21 PM Post #6,102 of 7,175
In his video (around the 19:00-19:40 mark), he said that's exactly what he was doing to pass the test so I don't know why he wouldn't admit that to you.


Only extremely naive implementations of DSP would make the signal lose bits (unless that was the job of the DSP in the first place). I'm not sure what's the deal with EQ APO but I don't think receiving 24bit samples instead of 16bit samples should help it.

Let's say all the processing that is done to a signal is first reducing the gain by 6dB and then increasing it right back by 6dB.
Reducing the gain by 6dB corresponds to dividing the samples by 2 (12dB would mean dividing by 4 and so on). Dividing binary numbers by 2 is very easy. You just have to shift all the 1-s and 0-s to the "right" (towards the least significant bit).

So for example dividing some 8 bit number by two looks like this: 00110110 (54) divided by 2 is 00011011 (27). Everything just got shifted.
Multiplying is also easy, except the direction of the shift changes: 00110110 multplied by 2 is 01101100 (108).
With that in mind it can be seen how a big gain reduction could make the signal lose bits as the last bits are getting shifted out.

One solution could be just tacking on some zeros to the original number, for example if the input is a bunch of 16bit numbers just add eight 0s to the end of it and that will ensure that the bits won't get shifted out, they just get shifted into the last couple of zeros after the gain reduction. An extra 8bit could allow for 48dB reduction before the signal starts to lose bits. While that might sound good enough, I'm pretty sure most plugins do their calculations on either 32bit float or 64bit floating point numbers which work quite a bit differently but let's just say they'll have enough dynamic range for any kind of sensible processing. If EQ APO doesn't use floats then I can wrap my head around why 24bit might be better for it than 16bits.

I think I followed some of that, VNandor. :wink:

As far as EQ APO is concerned, some of it (maybe most of it) may be due more to user error on my part. Because I didn't seem to have my device configured to the same bit/sample rate as the content to begin with. And was likely down-converting everything from 24/48 to 16/44.1, as well as making the volume adjustments on top of that. So there may have been a combination of factors that made the difference between the audio device settings more noticeable.

I'm doing this on a somewhat older PC as well, that may not have the most up-to-date algorithms and what have you as well. So that could also be a factor.

There did seem to be a slight (but noticeable imho) difference though in the clarity and smoothness of the audio after switching from 16/44.1 to 24/48. Maybe more than slight, in fact.

I don't know whether EQ APO is using floating or fixed point integers for resampling. But I believe the coding is open source, if you want to have a look. I am not a programmer. But if I were coding the app, I think I might use whichever method had the least latency, which would probably be the latter.
 
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Mar 14, 2021 at 10:32 PM Post #6,103 of 7,175
Yes it can be beneficial to use more bits during digital signal processing. But for most processing that users would want to do it would be no problem to start with a 16 bit signal, and do the processing in more bits. (That way accumulation of rounding errors that could faul up a few of the least significant bits won't touch the 16 most significant bits.)
[Edit: I didn't see @VNandor 's last post, but he basically says the same in other words.]

So basically as long as you're going from 16 to 24, and not the other way round, you should be ok then?

I wonder also about PC utilities for expanding dynamic range by undoing the compression used to boost loudness. And whether those could also produce some noticeable artifacts when going from 16 to 16 bits. If you're starting with content that has been brick-walled into an extremely narrow dynamic range, and expanding/resampling that out into a much wider dynamic range, then it would seem as though more bits might be beneficial for something like. I don't really know though.

Although I have not actually tried it, EQ APO also includes a filter for doing this kind of thing (as mentioned recently in another topic here)...

 
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Mar 14, 2021 at 11:08 PM Post #6,104 of 7,175
So basically as long as you're going from 16 to 24, and not the other way round, you should be ok then?
It also wouldn't be a problem to downconvert the end result back to 16 bits as a last step of the processing.
 
Mar 15, 2021 at 1:24 AM Post #6,105 of 7,175
In his video (around the 19:00-19:40 mark), he said that's exactly what he was doing to pass the test so I don't know why he wouldn't admit that to you.

You tell me. He was arguing here that because of his training, he could hear the difference between 16 and 24. He posted Foobar results to prove it. I asked him if he had changed the volume levels as he listened and he talked around my question, so I got more pointed and asked him if he gain rode the fade outs. He dodged it over and over and then got mad and went back to his own forum where he could delete the pesky questions like mine.

I think he was trying to impress us with his superhuman ears and when I pointed out that a noise floor below -90dB wasn't a noise floor under -90dB if you turn the volume up by +50dB, he didn't want to be challenged on it and just refused to answer. That guy has an agenda.

ADUHF, I said that the only purpose for 24 bit is to provide a deeper noise floor for sound processing in mixing and mastering. When you sit in your living room listening to Beethoven on a commercially recorded CD, there is absolutely no reason for it. Consumer sound processing doesn't get anywhere near making any difference either. 12 bit is sufficient for listening to music in the home. 16 bit is overkill. There is plenty of headroom there.

The only way to understand what is important and what isn't is to take some music tracks and run them through different kinds of degradation and see the effect in real world applications. Ethan Winer does that in the videos in my sig file. He takes a horrible buzzing noise and mixes it into music and drops it -10dB at a time. Take a guess where you can't hear it any more under the music... I think you will be very surprised. You can download his files and listen to them for yourself.

Without actually listening, specs are just abstract numbers on a page. Better numbers are better sound, right? Not always. To understand them with perspective, you need to translate those numbers to actual sound in a real world application. Then you know what -1dB sounds like as opposed to -10dB or -100dB. Numbers represent sound, but not always in an intuitive way. More is not always better. There is such a thing as good enough for human ears.

Dynamic expansion doesn't produce artifacts because of bit depth changes. It creates artifacts because there are many ways for the sound engineer who mixed the track to compress music. You have multiple variables, different ways to compress and different elements in the mix that can be compressed individually. Uncompressing it is like using a key to unlock a lock. If you don't know the exact kind of compression that was applied in the exact amount on the exact track at the exact point in the timeline, you can never uncompress it properly. You can only take a stab at it in one dimension across the whole track. The more you expand, the more artifacts you are going to get.
 
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