PCM 768kHz! - 24bit - FREE TRACK from Carmen Gomes Inc.'s Ray! Album, Is this overkill or does it make sense?

[APP]

Is this overkill or does it make sense?
and why not go all the way with a 32bit 768kHz file?

Ray! - PCM 768kHz! - 24bit - FREE TRACK (Let The Good Times Roll)

I do like the label a lot, so maybe it is time for a new DAC if their future releases are going to be in 768kHz.

The RME company is highly regarded by audio professionals.
Before we started working with Merging, RME was our converter of choice. So when we got offered to test the RME ADI-2 FS, a compact 2-channel AD/DA converter we gladly approved.

The first comparison with our Merging Anubis was immediately positive. Lots of definition, a beautiful soundstage with perfect placement. And after powering it with the Ferrum Hypsos external power supply, a sense of calm and control was added to the experience. This is clearly a serious converter.

The RME has a maximum sample rate of 768kHz. To really see what the RME ADI-2 is capable off, we created a 768kHz/24bit file straight from our Studer A80 tape recorder playing the ¼" reel to reel master tape from our latest release.
Listening and A/B comparing with the Studer the result is quite convincing. In our opinion the sound is very close to the analog master tape.

We would like to share the results with you.
Therefore we have made one 768kHz track from the album available for free for a limited time period.
The only favor we ask in return, is that you give us a bit of feedback; Is this a way forward?
Do you also hear an even greater sense of realism compared to the lower formats or are you perfectly happy with the formats you have been using so far?

For anyone who would like to purchase the entire album in 768kHz but has already purchased the album in a lower resolution in the past week, the price difference will be refunded. Send us an email and you will receive a refund within a few days.

 

[DasguteOhr]

Is this overkill or does it make sense?
and why not go all the way with a 32bit 768kHz file?

Ray! - PCM 768kHz! - 24bit - FREE TRACK (Let The Good Times Roll)

I do like the label a lot, so maybe it is time for a new DAC if their future releases are going to be in 768kHz.

Look at this, that is useably frequency response at sample rates, a good master 24bit/192khz is completely sufficient.
These are the response curves for different sample rates. 44, 96, 192 and 384 khz sample rates.
So for 192 khz you would expect good response to 80 khz and a droop beyond that. This particular device is down not quite 4 db at 80 khz. For the 96 khz rate which you would maybe expect good to 40 khz it is 5 db down at 40 khz. Now any of these sample rates would be quoted as - .1 db at 20 khz. And this isn't exemplary response. Some devices do better than this, but as you see the response while starting to drop does extend quite well above 20 khz at higher sample rates.

 

[APP]

Is a higher audio sample rate better?​

In theory, it’s not a bad idea to work in a higher audio sample rate, like 176.4 kHz or 192 kHz. The files will be larger, but it can be nice to maximize the sound quality until the final bounce.

interesting article; Is a higher audio sample rate better?

 

[Blue58]

Sounds better to me, deeper soundstage, bass that bit more focussed and more natural, to my ears, vocals. Can’t detect any negatives though perhaps the overall presentation is a little lacking in emotion.

cheers
blue58

 

[Kal Rubinson]

the overall presentation is a little lacking in emotion.

Yeah. I hear that in the music in any of the formats.

 

[DasguteOhr]

interesting article; Is a higher audio sample rate better?

dsd filters are theoretically set at 100khz bandwidth, in practice it is 70-80khz for most dacs. so does not bring any advantage. if dsd converts to pcm 192khz with eq program you won't hear a difference if that's done correctly. pcm has the advantage that the digital signal no longer has to be converted for all common connections exsample spdif. you relieve cpu and chip of unnecessary computing power.

 

[APP]

Different strokes for different folks. I think Carmen and her band are doing a superb job. Her way of phrasing reminds me of a good jazz instrumentalist, Dexter or Hank Mobley comes to mind.

 

[APP]

Different strokes for different folks. I think Carmen and her band are doing a superb job. Her way of phrasing reminds me of a good jazz instrumentalist, Dexter or Hank Mobley comes to mind.

Rushton Paul from Possitive Feedback Creative Arts Forum seems to agree:

With the wonderful lead vocals of Carmen Gomes, the Carmen Gomes Inc. band delivers an album I find totally engaging from start to finish. They make these songs completely their own while honoring the great Ray Charles in so doing.



Carmen Gomes Inc., Ray! Sound Liaison. 2021 (DXD) HERE

In making this album, Peter Bjørnild (bassist) writes, "We quickly realized that we did not want to make a tribute album. Rather we would pay our respects to Ray by following the advice Ray gave to Willie Nelson at a recording session: 'Don’t think about how anybody else does it, just do it the way that YOU feel it.' So we took the songs back to their bare essentials, tried to hear them as if they had just been written and in so doing making them our own."

And a fresh sound they indeed deliver. This is their music, with a nod to the master here and there. Never over-produced, always respecting each other, the musicians find a nice balance of supporting each other, challenging each other, complementing each other, never stepping on one another.

The song that I most enjoyed on my first listen was "CC Rider" by Ma Rainey and Lena Arant (circa 1924, also known as "See See Rider Blues"). As Bjørnild comments in the liner notes, "Here Tettero’s guitar is the 2nd protagonist in this story of a love affair taking a wrong turn. The guitar is of equal importance to Gomes’ voice, it’s the CC Rider Carmen is singing about. Listen how they keep on challenging each other yet never get in each other's way. That kind of communication and musical empathy is a rare find."

A rare find indeed. And a rare treat to hear.



Sound Liaison's recordings continue to impress me with their very clean, transparent, and precisely localized sound. Recording engineer Frans de Rond's signature is readily apparent in the purity of sound and precise placement of performers that he achieves in this multi-microphone studio recording. If you've heard his "One Mic" recordings on Sound Liaison, you will know how special his recordings can be. I was a bit hesitant at first knowing this would be a multiple microphone outing. But, he pulls it off superbly in a highly phase coherent, tightly knit, outcome that belies it's multi-mic roots. As with his other recordings, this displays great depth, and almost physically palpable placement of the musicians in the soundstage.

Highest recommendation.

 

[Ian B]

I haven't been able to listen because my own DAC does not go that high. However, 768khz is getting closer to DSD bandwidth. On the plus it should sound more natural and analog, but on the minus you also get lots of ultra high frequency noise building up that could be a hazard. I doubt this will take off as a format, but it continues to show that with digital the more bandwidth, the better.

A few years ago I tried some 768khz upsampling with a Chord Hugo 2, and it did sound a bit more natural and open, even compared to DXD, though the upsampling process was not totally benign. Based off that, I think you probably need 1.5mhz to perfectly get the same quality as DSD, and some way to deal with the ultrasonic noise.

 

[Ian B]

dsd filters are theoretically set at 100khz bandwidth, in practice it is 70-80khz for most dacs. so does not bring any advantage. if dsd converts to pcm 192khz with eq program you won't hear a difference if that's done correctly. pcm has the advantage that the digital signal no longer has to be converted for all common connections exsample spdif. you relieve cpu and chip of unnecessary computing power.

I hate to be contentious, but having done the DSD to 192khz and even 352khz conversion quite a few times, it is a big difference. And not good.

Also, DSD filters have typically a mild 6 or 12db/octave slope, and happen after the conversion, in analog. This effectively attenuation of HF noise, and not a brick wall cutoff frequency that you need with PCM. With 192khz PCM you need almost complete attenuation at 96khz and that usually means a 72db/octave digital filter.

 

[Al M.]

interesting article; Is a higher audio sample rate better?

This is nonsense. There are no stairsteps in digital. Rather, the sampling points are just that, points, and a perfect analog waveform connects them at the output of a D/A converter. At least, in any scenario where the sampling rate is at the minimum twice the highest frequency of an, emphasis, *bandwidth limited* signal. This, according to the Shannon-Nyquist theorem, which also holds for the Redbook CD format.

Thus, the reproduced waveforms will look the same in all three formats, left to right. No stairsteps with "errors".

 

[DasguteOhr]

I hate to be contentious, but having done the DSD to 192khz and even 352khz conversion quite a few times, it is a big difference. And not good.

Also, DSD filters have typically a mild 6 or 12db/octave slope, and happen after the conversion, in analog. This effectively attenuation of HF noise, and not a brick wall cutoff frequency that you need with PCM. With 192khz PCM you need almost complete attenuation at 96khz and that usually means a 72db/octave digital filter.

Use this filter with foobar works great for my ears,I have to admit that I have not a lot of sacd files.

S-Audio.Systems // DSD Filter

DSD FIR filter The DSD format is an unfiltered stream of a 1-bit SD modulator, for high-quality playback of which you need to completely filter out the enti

s-audio.systems

 

[Ian B]

Use this filter with foobar works great for my ears,I have to admit that I have not a lot of sacd files.

S-Audio.Systems // DSD Filter

DSD FIR filter The DSD format is an unfiltered stream of a 1-bit SD modulator, for high-quality playback of which you need to completely filter out the enti

s-audio.systems

I'm sure this design sounds reasonably good for DSD-PCM conversion, as do Pyramix's SRC filters, particularly the apodizing filter. However, the actual PCM D/A conversion will always have the same limitations, and will always sound like PCM. They inevitably have to use extremely steep digital filters and multibit modulators that exchange better noise performance for worse low-level linearity. The only way I've found to improve this is, ironically, to convert everything back to DSD. And even then, some detail is permanently lost, so better just to leave it native DSD.

 

[Jake Purches]

I do most of my commercial recordings at 352.8 kHz 24 bit. This is somewhat 'overkill' but it does yield a signal that is to all intents and purposes, 'analogue' as it would be understood, and exceeds the bandwidth of the best reel to reel recorders by 2 or 3 times. 192 kHz is in my view the minimum needed for a very realistic recording. 192 kHz is still a higher bandwidth than even 5.6 mHz DSD, but not far off. 352 or 384 comfortably exceeds 5.6 mHz and is closer to 11.2 mHz. DSD does have the advantage of a much more gradual frequency response fade out rather than a drop off. So for PCM - yes - 352 and double that ensures a completely analogue result from digital. And that is what we are trying to achieve. I haven't yet tried higher than 352 kHz, but this frequency satisfies what Tim de Paravacini said was 'exceeding what the ear and brain can resolve' in terms of timing and phase response. You do get a better stereo picture from high resolution. Its not just about bandwidth in the frequency domain but also in the pulse response and time domain. And why not? These frequencies are peanuts in this day.

 

[Jake Purches]

This is nonsense. There are no stairsteps in digital. Rather, the sampling points are just that, points, and a perfect analog waveform connects them at the output of a D/A converter. At least, in any scenario where the sampling rate is at the minimum twice the highest frequency of an, emphasis, *bandwidth limited* signal. This, according to the Shannon-Nyquist theorem, which also holds for the Redbook CD format.

Thus, the reproduced waveforms will look the same in all three formats, left to right. No stairsteps with "errors".

Its not entirely nonsense. This only holds true when the signal is bandwidth filtered. The music has to be filtered first for recording, to remove aliasing effects of higher frequencies, and after to mitigate the rubbish that comes out after the Nyquist frequency limit, what ever that may be. The sampling of the audio is a discrete sample and hold, so it is electrically a stair case, and whilst that is sufficient to satisfy the Nyquist Shannon theorem, this was the minimum standard not the absolute standard. Its no more strange than seeing pixels when you zoom in to a digital photo, and that has to be band width limited too, or Moiré effects are seen on detail smaller than the sampling rate. The filtering is why the music sounds 'fine' even at lower sampling rates. Most music that is orchestral, or even electronic, has very high frequency harmonics that often exceed 50 kHz. These are rudely beheaded by the CD audio 22 kHz limit and this is why CD audio can sound a bit 'hard', and the piano is often the victim of this. Even doubling to 88.2 kHz radically improves the situation and is quite audible, even for those who can't hear higher than 12 kHz.
Attached is a spectral analysis of Steely Dan SACD - at 96 kHz recording and then downsampled to 44.1 kHz CD audio. Its quite dramatic.

 

[Al M.]

Its not entirely nonsense. This only holds true when the signal is bandwidth filtered.

Of course, hence my phrase above:
"...of an, emphasis, *bandwidth limited* signal."

The music has to be filtered first for recording, to remove aliasing effects of higher frequencies, and after to mitigate the rubbish that comes out after the Nyquist frequency limit, what ever that may be. The sampling of the audio is a discrete sample and hold, so it is electrically a stair case,

The following video points out, not in theory but in measured practice on the instrument screen, that this is false:

and whilst that is sufficient to satisfy the Nyquist Shannon theorem, this was the minimum standard not the absolute standard. Its no more strange than seeing pixels when you zoom in to a digital photo, and that has to be band width limited too, or Moiré effects are seen on detail smaller than the sampling rate. The filtering is why the music sounds 'fine' even at lower sampling rates. Most music that is orchestral, or even electronic, has very high frequency harmonics that often exceed 50 kHz. These are rudely beheaded by the CD audio 22 kHz limit and this is why CD audio can sound a bit 'hard', and the piano is often the victim of this.

CD can sound a bit hard, but that has little to do with the medium itself. It depends on the recording/mastering, the DAC, and on the rest of the system.

Purchasing a great re-clocker between CD transport and DAC, cleaning up my power with better power cables and, recently, a Furutech 609 distributor box to further get rid of electrical noise,

e-TP609 NCF | FURUTECH

Features Axial Locking System [USA Patent No.7,648,391 / JP Patent No. 4616208] and EMI-Absorbing Formula GC-303 <br /> <br /> e-TP609 NCF - Rated 15A/125V AC (NEMA version)<br /> e-TP609E NCF - Rated 10A/250V AC (Schuko version)<br />

www.furutech.com

have all greatly helped in my system with getting rid of hardness, including on piano. Nothing to do with the CD sampling rate.

A higher sample rate may be better, but there is much more to digital reproduction in practice that is more important before sampling rate comes into play.

I am *not* saying that CD is perfect -- it isn't -- but I do have an issue with false myths about digital audio.

 

[Jake Purches]

First of all - this Monty video has done the rounds and is quite simplistic. Be that as it may, I don't subscribe to the notion of mains cleaners and DACs making much difference on CD Audio. Sorry I have to disagree with you nicely, at the engineering level as well as the audio level, that sampling is CRUCIAL to the quality of the audio. The DAC only has what it has to work with and can only improve a little bit. When information isn't there, it can't be magically returned. I regularly work with high sampling rates for my record label and its plain to hear the improvements when moving up to higher sampling. Its more important than going higher bit rates, and anyway trying to get more than 21 bits dynamic range in the real world is a challenge anyway. Of course that is where higher quality equipment does have a effect, like cleaner mains, I give you that.

 

[microstrip]

Is this overkill or does it make sense?
and why not go all the way with a 32bit 768kHz file?

Ray! - PCM 768kHz! - 24bit - FREE TRACK (Let The Good Times Roll)

I do like the label a lot, so maybe it is time for a new DAC if their future releases are going to be in 768kHz.

No system can have a true recoding resolution of 24 bits. At some point the lower bits are contaminated with noise and artifacts due to the process of sampling. We risk that high sampling rates introduces more nuisance than benefit, depending mainly on the implementation of the recording hardware and software. And unless we have measurements, all we have is an opinion on preference depending on listener and system. As there is a very small market for such recordings, we will only have a very limited number of opinions.

Again, sampling frequencies must be discussed separately for recording and for distribution to consumers - two very different objectives.

 

[microstrip]

Its not entirely nonsense. This only holds true when the signal is bandwidth filtered. The music has to be filtered first for recording, to remove aliasing effects of higher frequencies, and after to mitigate the rubbish that comes out after the Nyquist frequency limit, what ever that may be. The sampling of the audio is a discrete sample and hold, so it is electrically a stair case, and whilst that is sufficient to satisfy the Nyquist Shannon theorem, this was the minimum standard not the absolute standard. Its no more strange than seeing pixels when you zoom in to a digital photo, and that has to be band width limited too, or Moiré effects are seen on detail smaller than the sampling rate. The filtering is why the music sounds 'fine' even at lower sampling rates. Most music that is orchestral, or even electronic, has very high frequency harmonics that often exceed 50 kHz. These are rudely beheaded by the CD audio 22 kHz limit and this is why CD audio can sound a bit 'hard', and the piano is often the victim of this. Even doubling to 88.2 kHz radically improves the situation and is quite audible, even for those who can't hear higher than 12 kHz.
Attached is a spectral analysis of Steely Dan SACD - at 96 kHz recording and then downsampled to 44.1 kHz CD audio. Its quite dramatic.

As soon as I read words such as "rudely beheaded by the CD audio 22 kHz and presenting problems due to poor implementation of early digital as proofs of something I read the whole thing with care.

IMHO commenting childish movies or articles filled with half truths is not the proper way of promoting HiRez. Ulta high resolution must explain why it has benefit over the lower rates, such as 192/24 or DXD, not compare itself to CD. Digital recording is nowadays much more complex than it was twenty years ago.

 

[Jake Purches]

As soon as I read words such as "rudely beheaded by the CD audio 22 kHz and presenting problems due to poor implementation of early digital as proofs of something I read the whole thing with care.

IMHO commenting childish movies or articles filled with half truths is not the proper way of promoting HiRez. Ulta high resolution must explain why it has benefit over the lower rates, such as 192/24 or DXD, not compare itself to CD. Digital recording is nowadays much more complex than it was twenty years ago.

I wouldn't say its more complex to use - its certainly a lot easier. The technology and speed of the hardware has transformed what is now possible. I am setting up a orchestral recording with organ for Wayne Marshall next week, and we doing 7 channels at DXD 352.8 kHz 24 bit. An ordinary Macintosh Laptop that is 5 years old can easily accommodate such a bandwidth that would have been unimaginable 20 years ago.