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Topic: SACD DSD lossless to FLAC (Read 20816 times) previous topic - next topic
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Re: SACD DSD lossless to FLAC

Reply #25
Wrong EZ CD Audio converter sampling rate, sorry for that; those pictures and archive are correct.





Re: SACD DSD lossless to FLAC

Reply #26
Your Saracon .wav files contain a lot of zero samples at the end. You have to trim those samples so that RMAA can render the FFT plots correctly. I trimmed the 24-bit file for you and now the plot looks like this:
X

Also, foobar's "Direct" and "Installable" modes are not intended to be used with very high decimation ratios (DSD64 > 44.1k = 64x decimation). You can even identify in your 24-bit foobar screenshot, the 24-bit noise floor is somewhat wiggling and not as clean as multistage, EZ-CD-Audio-Converter and my corrected screenshot of Saracon. "Direct" is simply "Installable" without a user-selectable filter, and none of the bundled installable filters is suitable for more than 32x of decimation. You can see my tests here:
https://www.audiosciencereview.com/forum/index.php?threads/the-sound-quality-of-dsd.14773/page-24

You can see that these filters have around 44-50kHz of stopband. Ideal stopband for 44.1kHz sample rate should be at around 22kHz. An ideal installable filter for 64x decimation will be very long (several thousand samples) and conversion speed will be much slower than multistage.

Re: SACD DSD lossless to FLAC

Reply #27
So, going dsd64 to 44.1 pcm degrades signal?

Re: SACD DSD lossless to FLAC

Reply #28
So, going dsd64 to 44.1 pcm degrades signal?
SACD DSD is pulse density modulation, CD PCM is pulse code modulation.
That's unimportant and irrelevant. The main point is sample rate. FLAC supports up to 655350Hz, and DSD64 is 44100 * 64 = 2822400Hz. As long as the sample rate is not identical, the conversion will not be lossless. As mentioned earlier the principal of DSD is averaging a lot of 1-bit samples. You can have several numbers arranged in different orders that average to the same single number, for example:
1, 2, 3
2, 2, 2
3, 2, 1
...and so on, they all have an average of 2. But given a "2" and decompose to 3 discrete values, there is no way to know which three values in which order is the original one. It is not a reversible process, and therefore lossy, or "degrades", no matter how small the degradation is, audible or not.

Re: SACD DSD lossless to FLAC

Reply #29
I understand what your saying.
I know it can't be bit perfect, but it can be sonic "perfection" - precision; as close as possible to the source.
This is what I'm looking for; dsd to 16/44.1 pcm, best software that can make it possible.

Re: SACD DSD lossless to FLAC

Reply #30
And digital silence :)
Generated in Audacity, dsd64 in Saracon.
DSD64 to 16/44.1 in Saracon, Foobar2000 and EZ CD Audio Converter



Re: SACD DSD lossless to FLAC

Reply #31
The main point is sample rate. FLAC supports up to 655350Hz, and DSD64 is 44100 * 64 = 2822400Hz.
Came to think of: According to https://en.wikipedia.org/wiki/Comparison_of_audio_coding_formats#Technical_details  both  TTA and MPEG-4 ALS support up to 4 GHz sample rate and 1-bit resolution.

Re: SACD DSD lossless to FLAC

Reply #32
I attached the unfiltered 2822400Hz .wav file converted from the Saracon .dff file that @dev attached. It can be loaded into foobar, Audacity and so on.

Track 1: unfiltered
Track 2: 600kHz cutoff
Track 3: 200kHz cutoff
X

Re: SACD DSD lossless to FLAC

Reply #33
bennetng, you must be more precise at what your getting at.

Re: SACD DSD lossless to FLAC

Reply #34
A great article about DSD and comparisons between PDM and PCM is here: https://digital-audio-systems.com/pcm-im-vergleich-zu-dsd/
Unfortunatelly, the article is in German, but I'm guessing you could use Google translate.

The comparisons are taken directly from the suggestions made by Sony and Philips. Philips suggest doing production in PCM with 32B/352.8kHz (which is an even greater bit depth than DXD, which is 24B/352.8kHz). Sony suggests using "DSD-Wide" which is 4 or 8 bit PCM at DSD sampling rates.

PDM is in itself kinda interesting, I find the noise shaping characteristics and things like information density intriguing, but from a practicality standpoint, it seems to be more of a meme than anything. Also, a study (http://sdg-master.com/lesestoff/gesamtarbeitneu.pdf) found that there are no discernible differences to High-Res PCM at 24B/176.4kHz.
The "problematic" noisefloor above 15kHz removes the practicability of DSD/PDM further, such that PDM has a SNR of 120dB with modern encoders, and hence less than 24B PCM (which has 144dB SNR).

Pretty much no DSD production is creates only in DSD, this is pretty much impossible, if any audio editing is to be performed - which is almost always the case. And I don't remember ever having seen a direct-to-disk DSD recording.

Also, why even stop at DSD64, there's DSD512, too.

Re: SACD DSD lossless to FLAC

Reply #35
polemon, it's a good read, thanks.
Haha, I have to say it again i hate dsd :)

Re: SACD DSD lossless to FLAC

Reply #36
Did the PDM/PWM thing on disc emerge in the "1-bit Delta-Sigma DACs to everyone" bandwagon, or am I just confusing the timeline? Anyway, it is beyond limits of human hearing since long ...

Philips suggest doing production in PCM with 32B/352.8kHz (which is an even greater bit depth than DXD, which is 24B/352.8kHz).

Floating-point makes sense in processing. 32-bit float can losslessly contain whatever was stored in 24-bit integer, and storage is cheap.

Re: SACD DSD lossless to FLAC

Reply #37
Did the PDM/PWM thing on disc emerge in the "1-bit Delta-Sigma DACs to everyone" bandwagon, or am I just confusing the timeline? Anyway, it is beyond limits of human hearing since long ...
Well, ΔΣ isn't even used anymore in favor of other encoding systems.
Quote
Floating-point makes sense in processing. 32-bit float can losslessly contain whatever was stored in 24-bit integer, and storage is cheap.
I'm not sure they meant 32-bit float. Perhaps they did, but then they could've just specified 32-bit floating point. Instead the "32 bit" sounds a lot like 32 bit of a singed long int to me instead of double. Might be wrong though.

Re: SACD DSD lossless to FLAC

Reply #38
bennetng, you must be more precise at what your getting at.

Sure. A more precise one. Just lower the cutoff frequency. Anyway, a screenshot only has several hundred pixels of resolution, it is just an illustration of the averaging in action.
X

Re: SACD DSD lossless to FLAC

Reply #39
bennetng, you must be more precise at what your getting at.

Sure. A more precise one. Just lower the cutoff frequency. Anyway, a screenshot only has several hundred pixels of resolution, it is just an illustration of the averaging in action.
[attach type=image]21643[/attach]
Let me rephrase it, I don't understand the point that you are making, and I would like to :)

Re: SACD DSD lossless to FLAC

Reply #40
Let me rephrase it, I don't understand the point that you are making, and I would like to :)
So that people who are interested can try to inspect and process the file at the original sample rate themselves, including lurkers and other participants.


Re: SACD DSD lossless to FLAC

Reply #42
Did the PDM/PWM thing on disc emerge in the "1-bit Delta-Sigma DACs to everyone" bandwagon, or am I just confusing the timeline? Anyway, it is beyond limits of human hearing since long ...

Philips suggest doing production in PCM with 32B/352.8kHz (which is an even greater bit depth than DXD, which is 24B/352.8kHz).

Floating-point makes sense in processing. 32-bit float can losslessly contain whatever was stored in 24-bit integer, and storage is cheap.
1-bit SDMs were popular in the 90s. Multibit SDMs are more popular nowadays. Just find the datasheets from chipmakers like AKM, Cirrus, ESS, TI. For older products, this website has a pretty comprehensive list:
http://stephan.win31.de/dac-adc-hist.htm

With 1-bit, everything solely relies on averaging of merely two extreme values, with more bits, you can utilize more amplitude values plus averaging. In fact, even typical PCM dithering methods contain values at +1, 0 and -1. With DSD, there is no 0, silence is encoded with sequences of +1 and -1, and often result in audible clicks during track change.

It's just an evolution of IC technology like CPUs. Instead of simply increasing frequency, they also have more cores, more SIMD extensions, better floating point support and so on.