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Topic: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter? (Read 6658 times) previous topic - next topic
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SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Hi, the title is self explanatory, but for more context I have 2 versions of an album which I would like to encode to lossy (AAC): The CD version with 44.1KHz-16bit, and the 2 channel SACD version with 88.2kHz-24 bit.
So, does a lossy encoding sound better/have better fidelity if encoded from the SACD source? I don't notice any difference from my tests, but if I plug it on a (future) Hi-Fi system, could it be more noticeable? Or the extra "resolution" from the SACD source gets thrown away when encoding to lossy?

I tested 3 lossy formats for this, and at least one of them actually preserved the 88.2kHz sampling rate:

 - AAC    48kHz  (still more than the 44.1kHz from the CD version)
 - OGG    88.2kHz
 - OPUS  48kHz / 88.2kHz, depending on the program that analyzes it  :p
  (couldn't get the bit depth for any of them)

Since OGG preserves the high sampling rate, does that matter in the end result, or it's the same as if I did the encoding from the normal CD source, with only 44.1kHz?


And for another question, someone has borrow me another album, this one in 5.1, also with 88.2kHz-24bit but obviously with 6 channels. How can I choose the bitrate?  I saw here a post talking about this, but I didn't understand it very well and honestly it was from 2006 or something, things may have changed radically meanwhile.

I know that AAC, OGG and OPUS can encode 6 channels, so, if I want to encode that album with also lossy, and assuming that I like my 2-channel lossy files at ~192kbps, what bitrate should I choose for a 6 channel lossy file?

 Trying to answer this I made another test by encoding a 5.1 song into AAC, OGG and OPUS, choosing for each a bitrate of ~192kbps, and the AAC got 529kbps while the OGG got 827kbps, so it seems to add the bitrate to every channel, but it isn't a simple sum or the bitrate would be even bigger, so I imagine it lowers the bitrate when that channel isn't producing sound at the moment.
 Weirdly the OPUS got only a bitrate of 187, with a file size close enough to what a 2 CHANNEL 192kbps file would have. But it has 6 channels, I see it in the info and I see the 6 sound waves in Foobar. So, did it divide the 192 kbps by 6 in OPUS?
 What bitrate should I choose for at least AAC?

Thanks in advance for any advice you can give me.
I think this is the right board, but if it isn't, feel free to move it.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #1
Psychoacoustic lossy formats have no bit depth per se - they store data differently and in floating point format internally.
Also they shall filter high frequency content above 16-22kHz, so, no - 88.2/24 source shouldn't give you any perceivable better fidelity.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #2
Hi, the title is self explanatory, but for more context I have 2 versions of an album which I would like to encode to lossy (AAC): The CD version with 44.1KHz-16bit, and the 2 channel SACD version with 88.2kHz-24 bit.
So, does a lossy encoding sound better/have better fidelity if encoded from the SACD source?
Unlikely. And although there are "better reasons SACD should be worse", that is not likely either.

Lossy encoders discard what you cannot hear - or more cleverly: what is expensive to encode and has small chances of contributing to fidelity. And that in particular goes for ultrasonics.
Moreover, SACD has extra ultrasonic noise, so encoding the ultrasonics would be extra costly for something that is even unwanted. Look at this https://hydrogenaud.io/index.php/topic,122580.msg1012188.html#msg1012188 . The noise rising from the 20 kHz frequency is not loud, but you do not want a lossy encoder to spend bits on it.
A lossy encoder that spends its bits on ultrasonics, would put fidelity at risk. And if you tried to encode that way - without the application overriding you - it would be worse with SACD. Of course if the encoder already has enough bits to spend on the music, it wouldn't be audible.


Or the extra "resolution" from the SACD source gets thrown away when encoding to lossy?
The extra "resolution" and the extra unwanted noise from SACD should both be thrown away. That is not to say that you will never hear the difference, if you fine-tune bitrate to the level where you with careful listening can tell one of them from the original - but then it is also likely to be due to differences in mix/mastering (i.e. the actual music).

- OPUS  48kHz / 88.2kHz, depending on the program that analyzes it  :p
Opus works in 48 internally, but it stores the sampling rate from the source.
There are reasons for that. One is: If the sampling rate is lower, say 8 kHz, then Opus will tell you that even though it uses 48 internally, it isn't any better than 8. 

I don't know about Ogg. It is deprecated in favour of Opus anyway.

And for another question, someone has borrow me another album, this one in 5.1, also with 88.2kHz-24bit but obviously with 6 channels. How can I choose the bitrate?  I saw here a post talking about this, but I didn't understand it very well and honestly it was from 2006 or something, things may have changed radically meanwhile.

This from 2023: https://hydrogenaud.io/index.php/topic,124479.msg1030864/topicseen.html
With a link to https://hydrogenaud.io/index.php/topic,120007.msg997612.html#msg997612 which sets forth a nice rule of thumb. In particular, if you want to encode 5.1, double the bitrate you would use for stereo.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #3
Psychoacoustic lossy formats have no bit depth per se - they store data differently and in floating point format internally.
Also they shall filter high frequency content above 16-22kHz, so, no - 88.2/24 source shouldn't give you any perceivable better fidelity.
Thanks, I didn't know about lossy not having bit depth.
As for the frequency, is that really it? I mean, lossy encoding shedding away the frequencies above 20kHz is different to the frequency of the sampling rate. The previous is how much you can hear a high pitch sound and discarding what you can't hear, while the latter is how 'close to analogue' you can make a digitally sampled wave. And since OGG seems to preserve that original high resolution sampling rate, does it mean the sound has more "warmth, fidelity, dynamics" and all that stuff that people swear they have with analogue (sometimes overselling it a bit)?
  Or do those high sample rate, high bit depth sound only matter at the mastering level?

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #4
Lossy encoders discard what you cannot hear - or more cleverly: what is expensive to encode and has small chances of contributing to fidelity. And that in particular goes for ultrasonics.
Moreover, SACD has extra ultrasonic noise, so encoding the ultrasonics would be extra costly for something that is even unwanted. Look at this https://hydrogenaud.io/index.php/topic,122580.msg1012188.html#msg1012188 . The noise rising from the 20 kHz frequency is not loud, but you do not want a lossy encoder to spend bits on it.
A lossy encoder that spends its bits on ultrasonics, would put fidelity at risk. And if you tried to encode that way - without the application overriding you - it would be worse with SACD. Of course if the encoder already has enough bits to spend on the music, it wouldn't be audible.
(...)
The extra "resolution" and the extra unwanted noise from SACD should both be thrown away. That is not to say that you will never hear the difference, if you fine-tune bitrate to the level where you with careful listening can tell one of them from the original - but then it is also likely to be due to differences in mix/mastering (i.e. the actual music).
I'll check that link later, but thanks.
Yes I know the ultrasonics are a waste for listening purposes (nor for mastering purposes I bet), but I thought it could be the case that lossy codecs had more to work with to improve the encoded sound, if sourced from those high sampled high bit depth files.
So what you're saying is that all that "resolution" will be cut away in the lossy process, so it doesn't matter if I choose either CD or SACD as the source?

This from 2023: https://hydrogenaud.io/index.php/topic,124479.msg1030864/topicseen.html
With a link to https://hydrogenaud.io/index.php/topic,120007.msg997612.html#msg997612 which sets forth a nice rule of thumb. In particular, if you want to encode 5.1, double the bitrate you would use for stereo.

Thanks, I will check those links too. But that double the bitrate rule seems to apply more to the Opus format, since opus applies our chosen bitrate in general and not for each channel (192kbps, 6 minute song, 8.36MB file, what you would have with a 2 channel file, never with a 6 channel one), while AAC & OGG when choosing 192 kbps ended up with bitrates generally associated with lossless files, of many hundreds of kbps, I doubt those need even more bitrate.

Thanks for the answers so far.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #5
Yes I know the ultrasonics are a waste for listening purposes (nor for mastering purposes I bet), but I thought it could be the case that lossy codecs had more to work with to improve the encoded sound, if sourced from those high sampled high bit depth files.
Theoretically it wouldn't be unthinkable, say if what happens at 36 kHz informs the encoder on whether some 18 kHz content is an instrument or noise. (I would actually not be surprised it could be useful if and when someone put an artificial intelligence bot at work with voices at much lower lowpass.)

But is anything like that even attempted for music?

Anyway, ...
you are going to keep the originals, right? Not throw them away ... ? You don't want to do a rip job more than once.
Then consider lossies as disposable.
No need to start too high - if it sounds bad, you can ditch them and re-encode.
And if you did start high, no problem until your mobile device runs out of space - then again you can ditch them and re-encode.

What I do, is to encode to a format where I don't have much from other sources. Lossies are copied, lossless are converted: If my phone has an MP3 file, it is copied, not transcoded, but if my phone has a Musepack file, then I know it is converted from something and thus, this is not an "original" file.


Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #7
while the latter is how 'close to analogue' you can make a digitally sampled wave.
It doesn't work like that.

Consider circles. You can reconstruct a circle just from 3 points. Having 100 points doesn't improve anything. Similarly for signals. You can reconstruct a band-limited signal from samples at sampling rate just higher than 2x the bandwidth of the signal. Having sampling rate significantly higher than 2x doesn't improve anything.

while AAC & OGG when choosing 192 kbps ended up with bitrates generally associated with lossless files, of many hundreds of kbps
That's not my experience, at least with vorbis. Here "st.wav" is 2.0 file with pink noise and "mch.wav" is 5.0 file with pink noise, both 30 seconds long:
Code: [Select]
]$ oggenc -b256 "st.wav"
Done encoding file "st.ogg"
    Average bitrate: 253.2 kb/s

]$ oggenc -b256 "mch.wav"
Done encoding file "mch.ogg"
    Average bitrate: 268.5 kb/s

]$ ls -lh *.ogg
... 987K ... mch.ogg
... 932K ... st.ogg

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #8

you are going to keep the originals, right? Not throw them away ... ? You don't want to do a rip job more than once.
Then consider lossies as disposable.
I don't plan to get rid of the originals for now. If I need a new lossy reencode, the Flacs will be the "masters" for that.

What I do, is to encode to a format where I don't have much from other sources. Lossies are copied, lossless are converted: If my phone has an MP3 file, it is copied, not transcoded, but if my phone has a Musepack file, then I know it is converted from something and thus, this is not an "original" file.
I'm not sure I follow you here, you mean to do the encodes in more rare, exotic formats so one knows it can only be their own reencodes?

And yes, musepack gave me errors trying to encode 6 channel audio, otherwise it would have been included in my tests.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #9
Consider circles. You can reconstruct a circle just from 3 points. Having 100 points doesn't improve anything. Similarly for signals. You can reconstruct a band-limited signal from samples at sampling rate just higher than 2x the bandwidth of the signal. Having sampling rate significantly higher than 2x doesn't improve anything.

You're talking about the Nyquist frequency right? Yes, I guess that's why the 44.1kHz sampling rate is a little more than double the 20kHz human hearing limit. But what I meant is, what if between each sample in 44.1kHz, those intervals with silence (no sampling) actually had some higher frequency, that granted we wouldn't hear, but if repeated throughout the music, in the long run we could actually "feel" in a non quantified way?
I know this is a somewhat controversial topic with lots of placebo and wishful thinking in it, I for one can only barely and RARELY feel the difference between 128kbps and 192, but it was more like "since I have these lossless sources, can I make a lossy encode superior to another one from a simple CD source?"

while AAC & OGG when choosing 192 kbps ended up with bitrates generally associated with lossless files, of many hundreds of kbps
That's not my experience, at least with vorbis. Here "st.wav" is 2.0 file with pink noise and "mch.wav" is 5.0 file with pink noise, both 30 seconds long:
Code: [Select]
]$ oggenc -b256 "st.wav"
Done encoding file "st.ogg"
    Average bitrate: 253.2 kb/s

]$ oggenc -b256 "mch.wav"
Done encoding file "mch.ogg"
    Average bitrate: 268.5 kb/s

]$ ls -lh *.ogg
... 987K ... mch.ogg
... 932K ... st.ogg

I don't know what to say, that's not the result I have here, and I repeated the process 3 times, 2 with thefoobar convert tool, and one directly with the command line, both using the oggenc.exe. And see the attached pictures with the file and audio data of the resultant ogg file in my foobar , the foobar (pay no attention to my "experimental" interface lol), the encode foobar setting, and my version of oggenc.exe.
In the command line I used
 oggenc "filename.flac" -q 6
, which is the equivalent for 192kbps more or less. For some reason I couldn't make the --bitrate parameter to work.
In foobar, the 1264 kbps in the ogg file is a bitrate peak, the average is 800-and-something kbps

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #10
You're talking about the Nyquist frequency right?
Yes.

But what I meant is, what if between each sample in 44.1kHz, those intervals with silence (no sampling) actually had some higher frequency, that granted we wouldn't hear, but if repeated throughout the music, in the long run we could actually "feel" in a non quantified way?
Then writing about "how 'close to analogue' you can make a digitally sampled wave" was a really strange way of expressing it.

If you want to believe that perceiving (not necessarily hearing) supersonic frequencies is possible then yes, you need to keep those frequencies and thus use higher sampling rate.

In the command line I used
 oggenc "filename.flac" -q 6
, which is the equivalent for 192kbps more or less. For some reason I couldn't make the --bitrate parameter to work.
Well, there you go. "-q" is not about bitrate but about quality. It does automatically what the article that was linked describes how to do manually. It chooses a bitrate to keep some specified "quality". It shouldn't be a surprise that to keep the same quality with more channels, it needs to allocate more overall bitrate.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #11
Psychoacoustic lossy formats have no bit depth per se - they store data differently and in floating point format internally.
Also they shall filter high frequency content above 16-22kHz, so, no - 88.2/24 source shouldn't give you any perceivable better fidelity.
Thanks, I didn't know about lossy not having bit depth.
As for the frequency, is that really it? I mean, lossy encoding shedding away the frequencies above 20kHz is different to the frequency of the sampling rate. The previous is how much you can hear a high pitch sound and discarding what you can't hear, while the latter is how 'close to analogue' you can make a digitally sampled wave. And since OGG seems to preserve that original high resolution sampling rate, does it mean the sound has more "warmth, fidelity, dynamics" and all that stuff that people swear they have with analogue (sometimes overselling it a bit)?
  Or do those high sample rate, high bit depth sound only matter at the mastering level?
Many encoders apply bitrate dependent lowpass filter no matter the sampling rate.
Even lossless high bit depths and sample rates are practically impossible to double blind test against 44.1/16, so preserving such content in lossy encoding is considered useless - yes, their use is mostly for mastering or other special purposes.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #12

But what I meant is, what if between each sample in 44.1kHz, those intervals with silence (no sampling) actually had some higher frequency, that granted we wouldn't hear, but if repeated throughout the music, in the long run we could actually "feel" in a non quantified way?
Then writing about "how 'close to analogue' you can make a digitally sampled wave" was a really strange way of expressing it.

You understood what I said  : )


Well, there you go. "-q" is not about bitrate but about quality. It does automatically what the article that was linked describes how to do manually. It chooses a bitrate to keep some specified "quality". It shouldn't be a surprise that to keep the same quality with more channels, it needs to allocate more overall bitrate.
Aah, but in Foobar I encoded by choosing the bitrate, more specifically, moving the bitrate slider. And both files encoded by F2k and command line ended up with very similar bitrate and file size. So it's not just because of the encoder parameter 'q' I used, OGG and AAC apply the bitrate chosen to each channel, which compounds to a bigger general bitrate (minus similarities between channels that get subtracted) and file size.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #13
Many encoders apply bitrate dependent lowpass filter no matter the sampling rate.
Even lossless high bit depths and sample rates are practically impossible to double blind test against 44.1/16, so preserving such content in lossy encoding is considered useless - yes, their use is mostly for mastering or other special purposes.

In other words, any source (CD, SACD) will do fine when converted to lossy, and practically impossible to ABX the differences, no matter the bit depth and sampling rate?

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #14
Yes. If you can't ABX CD from SACD (assuming that mastering is the same), it's even less likely that you'll ABX lossy encodes of them. There's even a possibility of SACD encodes being inferior due to some unnecessary bits spent on ultrasonics in expense of what can actually be heard.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #15
Yes. If you can't ABX CD from SACD (assuming that mastering is the same), it's even less likely that you'll ABX lossy encodes of them. There's even a possibility of SACD encodes being inferior due to some unnecessary bits spent on ultrasonics in expense of what can actually be heard.

I see. More like a gimmick and "buy all this stuff again because theoretically is better".
 But as for your 2nd sentence the lossy process, though depending on the codec and bitrate, filters out everything above 20kHz, so those ultrasonics shouldn't be there in the 1st place.
 What I thought was if those ultrasonics, despite being cutt off when encoding lossly, would in a way "temper" the audible samples in the lossy encoding to make it better and closer to the original in some way, as in, ultrasonics that gave more accurate information to the codec about what samples to dedicate more bitrate, now that they have more resolution to work with. Like the "warmth" of the original. But apparently not. I guess all that extra data goes away. A pity, but since many people cant ABX the SACDs and CDs lossless sources in the 1st place, I guess not much is lost, at least perceivable.
 But it begs the question: Why does then ogg preserve the original sampling rate at 88.2kHz in the final lossy file?

By the way I use terms like "feel" and warmth", because I'm kind of a noob at this level of audio knowledge, and not because I think vinyl is better "just because vinyl", since as long as the masters are good and the same between it and e.g a CD, in many instances the "warmth" of vinyl derives from it's amplifier, that "colours" the sound.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #16
Filtering out something at 20kHz doesn't mean that precious bits won't be used for pretty useless 19kHz content.
The way data is stored in such lossy codecs makes it quite easy to resynthesize the sound at any bit depth and sample rate. Vorbis uses original sample rate to avoid resampling which may have its own share of problems.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #17
Filtering out something at 20kHz doesn't mean that precious bits won't be used for pretty useless 19kHz content.
The way data is stored in such lossy codecs makes it quite easy to resynthesize the sound at any bit depth and sample rate. Vorbis uses original sample rate to avoid resampling which may have its own share of problems.

1) Good point.
2) I didn't know that, thanks.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #18
Quote
But what I meant is, what if between each sample in 44.1kHz, those intervals with silence (no sampling) actually had some higher frequency, that granted we wouldn't hear, but if repeated throughout the music, in the long run we could actually "feel" in a non quantified way?
If you can't hear it you can't "feel" it.   (Well....  You can fell deep or subsonic bass in your body but that's a different topic.)

And if something makes you "feel different" that can be demonstrated (or debunked) with a  Blind ABX Test.
 
Quote
I know this is a somewhat controversial topic with lots of placebo and wishful thinking in it,
It's only "controversial" among people who "don't believe in" blind listening tests.   (And unfortunately, the "audiophile community" is dominated by irrational people.)

Quote
By the way I use terms like "feel" and warmth", because I'm kind of a noob at this level of audio knowledge
A lot of "experienced audiophiles" use also these meaningless words.    Even common words like "warmth" have more then one meaning.   You could fill a dictionary with audiophile nonsense.   

Audiophoolery defines the 4 characteristics that define/determine sound quality.    (There are more than 4 "words" but the terminology should fall-into one of the 4 categories.)

Quote
and not because I think vinyl is better "just because vinyl", since as long as the masters are good and the same between it and e.g a CD, in many instances the "warmth" of vinyl derives from it's amplifier, that "colours" the sound.
A good amplifier won't have any audible characteristics unless you over-drive it into clipping (distortion).   Sometimes there is audible noise (hum, hiss, or whine, in the background) but distortion and frequency response (the only other two things that affect sound quality of an amplifier) are almost always better than human hearing.     Phono preamps (and microphone preamps) are high gain and are more prone to noise because every active circuit generates some noise and amplification amplifies the noise along with the signal.

Vinyl has several "limitations" but again, the noise is the most obvious and the most unavoidable.    The low-level background noise may be part of the "vinyl warmth".   

It's OK if some people prefer the sound of vinyl and that's a matter of taste.   But technically. it's inferior to digital.   And you can digitize vinyl and the digital copy will sound identical to the vinyl (in a proper blind ABX test).

Quote
I see. More like a gimmick and "buy all this stuff again because theoretically is better".

I could build an amplifier that goes to 1 MHz.   (It's not that hard if you don't need high power.)   You could say it's "theoretically better" than an amp that only goes to 20kHz.    But for the purpose of audio it's NOT even "theoretically" better.   

Somebody once said, "The wider you open the window, the more dust comes-in".    And if you are converting SACD you should filter-out the ultrasonic noise because there is a possibility of it "causing problems" if you amplify it and send it to your speakers.  (I think most SACD/DSD DACs filter it out, but sometimes people convert to high-resolution PCM without filtering.)

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #19
If you can't hear it you can't "feel" it.   (Well....  You can fell deep or subsonic bass in your body but that's a different topic.)

And if something makes you "feel different" that can be demonstrated (or debunked) with a  Blind ABX Test.
Both are not mutually exclusive, though, granted, the "feel" isn't very scientific to say the least, and it may relate to something we only perceive at the subconscious level. All unproven of course. I can't "feel" the difference between them with my current equipment, but with better equipment in the future, who knows?
Then again, I tested yesterday my hearing: My best is 15KHz at my right ear, and a very low volume 14KHz in my left ear, though the results are a little better if I pump the volume a bit. It should be mentioned though, those sine waves came from youtube, which already applies lossy compression of AAC 128kbps, *but*, with a sound as simple as a sine wave, I think even a 64 kbps bitrate is more than capable of that.
 
It's only "controversial" among people who "don't believe in" blind listening tests.   (And unfortunately, the "audiophile community" is dominated by irrational people.)
(...)
A lot of "experienced audiophiles" use also these meaningless words.    Even common words like "warmth" have more then one meaning.   You could fill a dictionary with audiophile nonsense.  

Audiophoolery defines the 4 characteristics that define/determine sound quality.    (There are more than 4 "words" but the terminology should fall-into one of the 4 categories.)
Yes, seeing, or in this case, hearing, is -ou should be- believing.
Sometimes I believe more in other people ABX tests than my own, since they appear to be better trained. Other times I find praised remasters sounding more dull compared to the originals, even without high compression of dynamics on the former, but these must be the remasters themselves and not my earing. They may sound better because they are louder, but I compensate with replaygain, and they're the same, sometimes even add more noise than the original from more than 3 decades ago, which I think is unforgivable considering current technology. But remasters and "remasters" are a whole other topic of their own.

A good amplifier won't have any audible characteristics unless you over-drive it into clipping (distortion).   Sometimes there is audible noise (hum, hiss, or whine, in the background) but distortion and frequency response (the only other two things that affect sound quality of an amplifier) are almost always better than human hearing.     Phono preamps (and microphone preamps) are high gain and are more prone to noise because every active circuit generates some noise and amplification amplifies the noise along with the signal.

Vinyl has several "limitations" but again, the noise is the most obvious and the most unavoidable.    The low-level background noise may be part of the "vinyl warmth".   

It's OK if some people prefer the sound of vinyl and that's a matter of taste.   But technically. it's inferior to digital.   And you can digitize vinyl and the digital copy will sound identical to the vinyl (in a proper blind ABX test).
From the point that people spend unreal ammounts of money for simple wiring, is when they should be called out, for their own good. But it's easier to fool someone, than convince them they were fooled, specially with their big bucks wasted.
 For the wiring of both sound and power cables you only need, apart from good physical integrity of the cable, a good surface area (not too thin), bigger depending on the current flowing there, good shielding from interference, and in the case of power wiring, maybe even a capacitor in parallel to short-circuit noise from the main line, and this one maybe already a little overkill, but dirt cheap nonetheless. More that that is "electronic hocus pocus".

One case of intended amplifier distortion is present in Depeche Modes's Songs of Faith and Devotion, in which they used, among other instruments, a piano connected to a guitar amplifier (in over drive I guess), it sounded unique and distorted in a good way.

I could build an amplifier that goes to 1 MHz.   (It's not that hard if you don't need high power.)   You could say it's "theoretically better" than an amp that only goes to 20kHz.    But for the purpose of audio it's NOT even "theoretically" better. 

Somebody once said, "The wider you open the window, the more dust comes-in".    And if you are converting SACD you should filter-out the ultrasonic noise because there is a possibility of it "causing problems" if you amplify it and send it to your speakers.  (I think most SACD/DSD DACs filter it out, but sometimes people convert to high-resolution PCM without filtering.)

From what I got here is that the lossy codecs already filter that out, although the cuttoff ceiling could be pushed higher in cases of higher bitrates, whish would make the codec spend bitrate on ultrasonics, although on the other hand, since the bitrate was higher, maybe it could actually spare bitrate in those ultrasonics (maybe the codec engineers decided it could be useful?), but with benefits from little to none.


What about you, would you convert from a SACD, or a CD, assuming the codec will cut frequencies above some 16-17 KHz?   :)  And to which format? (AAC, Opus, lossy Wavepack since I heard it gives more dynamic,  another one?...)

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #20
What do you guys think about this video:  Sound beyond 20khz: The key to true high-fidelity?
He refers to music heard with high quality speakers that support that range (and the harmonics caused by it), recorded from microphones above 20KHz, and with good hi-fi equipment.
To his defense (he's a analog fan), he has another video where he says that when done properly, CDs are better than vinyl, and the best format in the world.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #21
Well, I believe that the target-question should be: "How to correctly feed a DSD into a lossy encoder in order to obtain the best possible quality result ?"

If so, I believe that discussion sould be moved to software DSD decoders' evaluation...
(for example the latest foosacd decoder implemented a new "ramp" parameter: how to exploit it ?)
Hybrid Multimedia Production Suite will be a platform-indipendent open source suite for advanced audio/video contents production.
Official git: https://forart.it/HyMPS/

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #22
Quote
What do you guys think about this video:  Sound beyond 20khz: The key to true high-fidelity?
I'm not going to watch the video, but the blind listening tests have been done.    If you low-pass filter at 20kHz you won't hear a difference.   

You can do your own ABX test.    ...If you use an audio editor (like Audacity) to make a low-pass filter, note that the cutoff frequency of a filter is defined as the -3dB point, so by definition a 20kHz low-pass filter is down 3dB at 20kHz, no matter how sharp the filter.   You probably still wouldn't hear it, but to keep it "flat" up to 20kHz, you'll need to use a higher cut-off frequency (and a steep filter, which is easy in software).

It's just the way the ear (and brain) works.   Our ears are something like a bandpass filter and our brain is something like a spectrum analyzer with some additional filtering. 

The guys who develop lossy encoders have determined that most of the time you can cut-off below 20kHz and not hear a difference because the highest frequencies in normal program material are weak and masked (drowned-out) by louder not-as-high frequencies, plus human hearing is weak at the highest audio frequencies.  Just because you can hear pure-loud 20kHz signals in a hearing test doesn't mean you'll hear them in music.   And not everybody can hear up to 20kHz. 

If there is no reason to limit frequency response (if you don't need to save file space or bandwidth) then there's no reason NOT to keep the full "traditional" 20-20kHz frequency range, or build equipment that can cover the full audio range.   And it "doesn't hurt" to go a bit beyond that.  

But if you have too much frequency range sometimes you can get unwanted side-effects with subsonic or ultrasonic signals or capability...   You don't want subsonic or ultrasonic "junk" going through your amplifier and to your speakers (unless you are trying to get subsonic "vibrations" from a subwoofer).  You especially don't want DC (zero Hz) in your audio.    Wave files are capable of 0Hz and I think MP3 is too.

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #23
For DSD64 to PCM, a target format of 24-bit 88.2kHz is enough. While there is some seemingly rise of noise above 20kHz, don't forget the noise floor of 16-bit is also 48dB higher than 24-bit, yet some CDDA releases still have noise shaping to pump up the noise floor beyond 16kHz or so.
X

The -1dB@25kHz.txt filter I made can be download here:
https://archimago.blogspot.com/2022/06/notes-on-dac-dsd-1-bit-pdm-measurements.html?showComment=1655906640637#c3453242716296098960
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Decimation should be set to 8 regardless of output sample rate as "Auto" seems buggy
Still true for the latest version of foo_input_scad.

Just to add, playing DSD64 directly on a DAC will result in MUCH higher ultrasonic noise than the illustrations above.
https://www.audiosciencereview.com/forum/index.php?threads/does-dsd-sound-better-than-pcm.5700/post-128312
https://www.audiosciencereview.com/forum/index.php?threads/marantz-sa-10-review-sacd-player-dac.31686/
https://www.audiosciencereview.com/forum/index.php?threads/digital-filter-game.23795/post-934885

Re: SACD to lossy vs. CD to lossy: do sample rate and bit depth matter?

Reply #24
Honest question:
One of those plots show that the 88.2 kHz signal generates ultra-ultrasonic noise at 30 dB lower.
If that sort of noise is ever going to be a problem, presuming you don't deliberately play bat sounds at tweeter-splitting levels?