Null test 88.2/24 and 44.1/16

[bachish]

Hi everyone!  First post here.

 

By way of introduction, I'm a professional cellist who has been dabbling in audio since I was a kid. I used to record my brother's band to cassette tape using radio shack microphones in the 1980s.  This eventually progressed to the point where I bought my first professional mics and mic pre and began doing my own recordings (a serious hobby since the mid 1990s).  Because I have a finite amount of funds and it gets split in three ways (cello, recording, monitoring and playback systems), I probably don't have as serious an audiophile system as many of you. My priorities are, in this order:  cello - recording equipment - playback system). But I do have some Paradigm Monitor 11s, v2 (tower speakers) that I bought back in the early 2000s, an older but transparent and detailed Harman Kardon Amp from which I bi-wire and bi-amp my Paradigms, a Grace M900 DAC, a Marantz CD player that set me back about $350 years ago, and an acoustically treated room to knock down the early reflections (makes a MASSIVE difference - highly recommended). 

 

Anyway, I posted a little null test I did over at gearslutz.com, the forum/hangout place for recording engineers.  It's not earth shattering but I thought as a first post I would share it here in a slightly different form.

 

For those who don't know, a null test is a way to see if two audio examples are identical and, if not, what the differences between them are.  All you have to do is flip the phase of one of the audio clips, make sure the levels are identical, mix them together, and see if they null.  If the audio is identical between them, the result will be a perfect phase cancellation.  No sound.  A perfect null. 

 

What I did was take an original recording I did of a jazz group at 88.2 Khz/24 bit and down sampled it to 44.1, flipped the phase, and dithered it to 16 bit. I then up-sampled back to 88.2Khz, which results in a 32 bit floating point audio (all processing results in more bits). I then truncated the 32 bit float back to 24 bits.

 

Here is the original file. Sorry the pics are so small. You may have to zoom in but you can see the extra high frequency content above 20KHz

 

 

Here is the down sampled then up-sampled version.  If you zoom in, you can see how the high frequencies are cut off in the down sampled and then re up-sampled version. The high frequencies are not regained despite the up-sampling. If you look carefully, you can see some of the added dither noise in the high frequencies as a band across the top. The other dither is hidden behind the audio.

 

 

 

I took the original 88.2/24 bit recording, loaded it onto a track in my DAW,  loaded the flipped phase version onto another track and mixed them together. The result was dead silence except at extremely high volumes during which I could hear some white noise. They nulled perfectly in the audible range (again, except for the extremely low level noise - mostly the dither noise) and so were identical in that respect. I heard zero music - not even slight residual sounds or artifacts.  Just hiss with the volume cranked full force!

 

You can see the two versions on tracks 1 and 2 and if you zoom in and look carefully, you will notice that the wave forms are mirror images of each other - they are inverted. You will also notice that the meters for the two tracks being compared are higher than the main meter to the right - so there isn't a perfect null.  But remember, the original has extra high frequency content. 

 

 

 

In a spectrogram, the nulled mix shows the dither noise (mainly shaped to the higher and lower frequencies and less in the mid-range) of the twice re-sampled version and the remaining high frequency content of the original 88.2 version - in other words, the only differences between the tracks.  All of the music from 20Hz to 20Khz nulled perfectly and so the original and re-sampled version are a perfect match in that frequency range.

 

 

 

Keep in mind that if any information was lost in the down sample to 44.1 and dithering to 16 bit, it cannot be regained by up sampling again to 88.2 and reverting to 24 bit. It's gone forever from that track.

 

This little test does not answer if the remaining high frequency content has any effect on the listener - that would necessitate another test. But it seems to me, assuming the re-sampling is done properly, that going from 88.2/24 bit to 44.1/16 bit does not change the audio between 20Hz and 20Khz - all stereo imaging, detail, transients, timbre, and anything else you can think of, remain perfectly in tact. If that were not the case, the two audio clips would not null as they did.

 

In a way it's comforting.

 

I hope this isn't seen as being too controversial as a first post!  Just a fun little experiment!

 

Here is the link at gearslutz for larger visuals and a more detailed description, if you are interested.

 

https://www.gearslutz.com/board/mastering-forum/1227563-my-null-test-88-2kh-24-bit-44-1kh-16-bit.html?posted=1#post13474865

 

Cheers

 

 

 

 

 

 

 

 

 

 

 

 

[bachish]

14 minutes ago, wgscott said:

Thanks. We need more of this kind of analysis   

 

Glad it is useful. Thank you!

[bachish]

11 minutes ago, firedog said:

Thanks. Not surprising. On my present system I stopped worrying about recording format. I mostly buy what the original was recorded in, if it is available. But most important is a good recording. Redbook can sound perfectly satisfying, and I agree there are certainly lots of examples where the difference between it and Redbook are negligible, if they actually exist.
 

But there could still be reasons hi-res would sound better/different to some people, ranging from equipment sweet spots to filtering differences changing the sound slightly between Redbook and hi-res.
There may also be other examples where transients and detail are easier to hear on the high res, even though they are also audible on the Redbook. I've also a couple of times thought I heard "new" detail in a hi-res version, only to then find out it was also audible on the Redbook version. But I didn't notice it on the Redbook till I heard it first on the hi-res version, where it was more apparent, or should I say easier to pick out?
 

Anyway, as a cellist, check out the just released Yo Yo Ma recordings of the six Bach Cello Suites, called "Six Evolutions".

Yes, there is the issue of how DACs handle Redbook audio regarding the filter. So this little test doesn't address that particular question. But I imagine, if your DAC is well made, and it probably is, that is does a pretty stellar job at 44.1/16 bit audio.  For my own recordings, I often listen to them in the high res original versions. It's partly psychological for me! haha

 

I haven't heard the new recording of the Bach suites. That's his third recording of them, I believe. The suites are something that never completely settle in a cellist's interpretation.  They are always changing. I suppose that is why the album is called 'Six Evolututions'. It's a great title. Thanks for letting me know! 

[bachish]

There definitely are frequencies above 20Khz emanating from a cello.  We also can play pretty high.  I also tend to use ribbon mics for recording the cello for their smooth, silky, analog-like sound and they do roll off around 15Khz.  But looking at a spectrogram they still record info in the ultrasonic range, albeit kind of softly.

 

Here is my position. I know many disagree on this topic.

 

I'm not convinced the ultrasonics make a difference in playback. My own hearing at age 52 is limited to 15Khz.  I think a downsampled file, assuming it is done correctly, has identical musical information as the original file up to around 22Khz as the null test above shows.

 

However, the above null test doesn't show whether a particular converter sounds the same playing the 44.1/16 bit version compared to the high res version. This is harder to test. All the null test shows is that 44.1/16 file is identical to the 88.2/24 bit file in the audible range with added noise. That is what a good sample rate conversion is today, fortunately - a low pass filter with added very low level noise. But how a particular converter handles the two file versions is now the question.

 

My own opinion is, assuming a high quality DAC, that the differences between the original high res and 44/16 versions have to be quite nuanced, if at all audible.  

 

BUT, nonetheless, I do receive some sort of psychological satisfaction from listening to the high res versions of recordings.  It's a geeky thing for me.  I know that the high res file, technically, has all the original info (ultrasonics) and a lower noise floor. And I don't have to wonder if the converter is doing a good job with the lower res version, even though it probably is doing a stellar job. 

 

In other words, why do I like to listen to high res files whenever possible? Because I can and I like to. 

 

 

 

 

 

 

 

 

[bachish]

31 minutes ago, pkane2001 said:

 

Can you please share the audio files you used for this analysis? I’d like to run them through my own software to see if there’s anything interesting it can find.

Sure, they are uploading now to the cloud.

 

Can I send the link privately in a PM? 

 

 

[bachish]

47 minutes ago, Jud said:

 

Insofar as the studio mics were able to capture it, and it wasn't removed by the ADC(s) during sigma-delta modulation or any downsampling.

Yes, I'm not saying the wav file has a perfect representation of the original source recorded.

 

I am comparing listening back to either the high res, original version vs the down sampled version.  

 

 

[bachish]

43 minutes ago, pkane2001 said:

 

Yes, of course.

I found where to do it...Wasn't sure the policy here.  I'll let you know when they are done

[bachish]

1 hour ago, testikoff said:

Shouldn't the spectrum of null-test delta look something like that (well, mine does)?.. 

 

I think it depends on what is captured in the recording and the software. The program I used above is a picture of the whole jazz tune in time (at the bottom it shows the elapsed  time).

[bachish]

52 minutes ago, testikoff said:

Post the spectrum of your null-test delta signal for comparison.

And the magic word is? ?

 

I did post the spectrum analyzer from Izotope RX. That is how it looks in that program. It is the last pic in my original post.

[bachish]

20 minutes ago, Jud said:

 

Right, just wanting us to remember that A/D is the flip side of what goes on in a DAC - first converted from analog to digital with sigma-delta modulation, then (with most ADCs) converted to PCM at 352.8 or internally/externally downsampled to the more common resolutions, 192 or 176.4kHz or below.  So even the "hi res original" from the studio is almost always downsampled.

Got it! Thanks!

[bachish]

2 hours ago, testikoff said:

What you posted is a spectrogram, not spectrum...

It's all I got. Sorry

 

Opening it up and looking at it closely again, everything except the high frequency content and perhaps the dither across the top looks to me to be -80db and lower, which is pretty quiet. And it all looks like random noise to me except, again, for the high frequencies at the top. 

 

 

[bachish]

BTW, if you are interested to see the results of sample rate conversions of differing software and hardware, this is a very handy tool,

 

http://src.infinitewave.ca/

 

I used Izotope RX 

[bachish]

1 hour ago, pkane2001 said:

Here's what my own DeltaWave software as able to determine from about a 2.5 minutes extract from the middle of the recording.

 

The comparison is between the original 88.2Khz/24 bit recording and a downsampled version to 44.1Khz/16bit, then upsampled back to 88.2Khz/24 bit, as described by @bachish (thanks for the files!)

 

By the way, the recording sounds excellent! I also listened to the difference between the two files, which is primarily white noise. I was able to hear occasional notes come through buried in the noise, but only with software digital volume adjusted to about 70dB gain and my DAC set to 0dB (max volume).

 

The files were delivered with the upsampled version with inverted phase. Otherwise, the files match perfectly in level and phase, no phase offset or drift was found.

 

First, a comparison of the spectra of both waveforms. Original 88.2/24 is in blue:

 

 

The drop off do to downsampling/upsampling starts around 21800Hz at -92dB down. Here's the zoomed in portion where the cut-off starts. You can also see very tiny differences in level due to dither and computational error during resampling:

 

 

 

Now the actual waveforms overlaid on top each other:

 

 

Stats below show excellent correlation between the two: 76dB correlated null and -75dB difference (rms).

 

Spectrum of the difference of the two files. Well below -115dB in the audible range:

 

Spectrogram of the original 88.2/24 file:

 

And spectrogram of the downsampled/upsampled file:

 

Spectrogram of the differences of the two files:

 

Interesting results in the cepstrum analysis of the two files:

 

 

This shows that the downsampled/resampled file has some ringing/aliasing going on at a number of frequencies that are not present in the original 88.2/24 bit file. Probably the side-effect of the resampling process and filtering applied. I tried to label the main ones. Note that the vertical value (Y axis) is a correlation coefficient. It's an indication of how strong the ringing is in the measured file, while the X coordinate is the frequency at which this ringing was detected.

 

While the artifacts of the resampling process are visible in the Cepstrum plot, they are not at all noticeable in any of the measurements or in listening to the differences between the two files.

 

 

 

 

Super impressive, Paul.  And that is your own software, correct?

 

I can see how software like this would be good to test differing settings.  There are a lot of options when re-sampling in the particular software I use, Izotope RX.

 

For those of you interested, here is the preset for sampling from 88.2 to 44.1 in Izotope.  It would result in some aliasing but my understanding is it results in less ringing.  Paul would know more than me on that subject. 

 

 

If desired, you can set an ideal filter with a very steep slope, which results in no aliasing but more ringing. 

 

So my understanding is it's basically a balancing act between aliasing and ringing. 

 

 

So I imagine with good software and a reasonable setting, any ringing and aliasing would be kept so soft to render it inaudible. 

 

But I am kicking myself right now because I should have saved as a preset the filter I used because that would have been really interesting to compare it to Paul's analysis. If my memory is correct, I used just a bit of a slope and may have caught just a touch of aliasing in high frequencies. So I put it somewhere between the preset and an ideal filter, I am pretty sure.

 

But what is interesting in Paul's analysis is that the amount of difference between the original and resampled version is well below -115 db.  This was a question that i was very curious about.  My understanding is that -80db below the signal is considered inaudible and covered by the masking effect.

 

Again, I can see how an analysis like this would be helpful to determine settings used, even if just for conscience sake!  Someone could spend hours trying different setting and running the results through a program like this. Very cool!

 

What also stands out to me is that the Izotope RX is not outrageously priced ($399 for the standard version I use) and it can perform sample rate conversion that seems to really stand up to analysis.  I remember back in the mid 1990s when you had to go to a mastering studio to get a smooth sample rate conversion. 

 

I missed the notes under the noise. But then again, I didn't listen to the every portion of the track. Perhaps I should have listened to the whole thing before posting. 

 

Again, very cool, Paul!  Thank you for taking the time to do this! It is very beneficial. Oh, and thanks for the compliment on the recording! 

 

Out of curiosity, do you sell your software? 

[bachish]

1 hour ago, pkane2001 said:

Here's what my own DeltaWave software as able to determine from about a 2.5 minutes extract from the middle of the recording.

 

The comparison is between the original 88.2Khz/24 bit recording and a downsampled version to 44.1Khz/16bit, then upsampled back to 88.2Khz/24 bit, as described by @bachish (thanks for the files!)

 

By the way, the recording sounds excellent! I also listened to the difference between the two files, which is primarily white noise. I was able to hear occasional notes come through buried in the noise, but only with software digital volume adjusted to about 70dB gain and my DAC set to 0dB (max volume).

 

The files were delivered with the upsampled version with inverted phase. Otherwise, the files match perfectly in level and phase, no phase offset or drift was found.

 

First, a comparison of the spectra of both waveforms. Original 88.2/24 is in blue:

 

 

The drop off do to downsampling/upsampling starts around 21800Hz at -92dB down. Here's the zoomed in portion where the cut-off starts. You can also see very tiny differences in level due to dither and computational error during resampling:

 

 

 

Now the actual waveforms overlaid on top each other:

 

 

Stats below show excellent correlation between the two: 76dB correlated null and -75dB difference (rms).

 

Spectrum of the difference of the two files. Well below -115dB in the audible range:

 

Spectrogram of the original 88.2/24 file:

 

And spectrogram of the downsampled/upsampled file:

 

Spectrogram of the differences of the two files:

 

Interesting results in the cepstrum analysis of the two files:

 

 

This shows that the downsampled/resampled file has some ringing/aliasing going on at a number of frequencies that are not present in the original 88.2/24 bit file. Probably the side-effect of the resampling process and filtering applied. I tried to label the main ones. Note that the vertical value (Y axis) is a correlation coefficient. It's an indication of how strong the ringing is in the measured file, while the X coordinate is the frequency at which this ringing was detected.

 

While the artifacts of the resampling process are visible in the Cepstrum plot, they are not at all noticeable in any of the measurements or in listening to the differences between the two files.

 

 

 

 

After I posted, I was looking again at your analysis and a couple questions occurred to me. 

 

In the Cepstrum plot, the y axis - I'm curious what measurement that is.  Just trying to get a handle on just how loud the ringing is. You mention the level is pretty low, I realize. 

 

In the 'Spectrum of the difference of the two files. Well below -115dB in the audible range' the left side of the graph looks like the noise shaping of the dither applied, which would make sense.  So essentially, that graph is showing that the difference between the two files is primarily in the dithering and the high frequency content from the original file. Am I reading that correctly?

 

Thanks!

 

 

[bachish]

3 hours ago, Jud said:

 

If you were trying to hear noise, that's correct. But we're trying to hear music.  How much very low level musical detail is the noise masking?  (Maybe none, considering the noise level of a typical listening room, but it's the way of conceptualizing noise and masking that I'm thinking of.)

I think when talking about noise in these files, we need to realize that -100db is the equivalent to 0.001 % distortion in audio equipment.  Highly unlikely we would hear this. 

[bachish]

8 hours ago, pkane2001 said:

 

Cepstrum attempts to find repeating patterns in the frequency domain. The Y axis in the plot is a correlation coefficient. All it's saying is that a relatively higher value has a relatively greater amount of ringing (i.e., the pattern is better defined). It doesn't say anything about the level of ringing.  Since it does point to specific frequencies, it might be interesting to see if the magnitude can somehow be dug up from the signal and noise in the frequency domain. Not sure that's possible, but I'll think about it some more

 

 

Right. Most of the differences are well below -115dB in the frequency domain. But, that represents an average over the measured period. For a time version of the difference plot, here is what it looks like in the time domain:

 

 

You can see some peaks rising to about 0.02. In dB terms, that represents about -34dB level. But that's just a few peaks. Most of the difference is well below -100dB level. By the way, this is the waveform I listened to to hear the difference. Most of it sounds like noise, with a few very occasional notes coming through. Probably corresponding to some of the peaks in the above plot. So, there is some difference between the files, they are not a perfect match, but they are very, very close.

 

Here's that same difference plot but with all the frequencies above 20KHz removed:

OK, thank you for the more detailed information. It would  be interesting to fine tune the conversion settings to see if those larger spikes couldn't be tamed. 

[bachish]

5 hours ago, Jud said:

I fully agree with what you both are saying.  Let me see if I can't also encourage you to think about this in a slightly different way, not necessarily for this particular example, but in general.

 

You're with a friend at a restaurant.  She wears a hearing aid.  She complains about the noise of conversation at the other tables.  You say "It's OK, it's not like we can overhear what they're saying."  She replies, "But I'm not worried about overhearing them.  They're making it more difficult to hear you."

 

Later, she mentions the A/C coming on and the noise that makes.  You say, "But the noise of conversation is louder."  She replies, "Yes, but both of them are making it more difficult for me to hear."

 

In other words: I'm encouraging thinking about noise masking music, rather than vice versa, since the music is what we're trying to listen to; and thinking about noise sources as additive, not as one excluding or drowning out another.

 

Now it may well be that in a typical situation, any additional noise from filtering, especially with good software like iZotope, amounts to a fart in a hurricane compared to the noise level in a typical listening room, so though additive, its contribution might be utterly negligible.  No problem with that at all.  I just want to encourage a mindset that looks at ability to hear musical detail, rather than masking and whether we can notice audible distortion.

 

Ok, I see what you are saying. You'd like the perspective to be reversed. In other words, let's not talk about masking the distortion or artifacts with music but how much music is masked by the noise. Its a fair point.

 

One thing that is rather cool in digital music, and you very well may know about it, is how one type of digital distortion, quantization distortion from truncating from higher bit depth to 16 bits, does actually block music at very low levels. There is a way to swap the quantization distortion with white noise (dither) which obscures the music far less. In fact, you can hear the music through the dither but not through the quantization distortion. The white noise is then shaped so it is not as noticable (moved to higher and lower frequencies where the ear is less sensitive).

 

So in a way, this is looking at distortion and noise from the kind of perspective you ar talking about, i.e. what masks the music less.  

 

So yes, I definitely think there is a place to think about distortion and noise masking music as you say.

[bachish]

4 hours ago, crenca said:

Interesting.  When the OP transmitted these files to Paul over the Internet, was the EM/RF "noise" of your computer, your ethernet cables and switches/routers, etc. taken into account?  Would "audiophile" networking equipment made any difference to any computational analysis?  What if you retransmitted these files several times and due to the differing paths these filed traveled over the Internet backbone (as any tracert will tell you), an analysis was done to see the effects these differing routes had?

 

Along these lines, what would happen if you took these files and transmitted them back and forth from your computer to an external HD connected via USB, say 1000, or 10,000, or 10,000,000 times?  How would this computational analysis be effected, and would it matter if your USB connections used "Audiophile" USB cables, USB decrapifiers, and differing USB design implementations?

I have to admit, you did make me chuckle. Point well made, though, in a somewhat smart @$$ way. Haha

 

 

[bachish]

4 hours ago, Jud said:

 

Following up on this -

 

Here are some recordings available from NativeDSD, and you can filter your search to come up with only those albums/tracks available in DSD256: https://www.nativedsd.com/new_browse

 

@bachish, @pkane2001, I don't know whether you're able to do the same analysis with one of these tracks, but it would be interesting to see whether a recording that presumably had experienced less processing would show any greater difference to a 44.1kHz resolution downsampled file than the 88.2kHz original you used before.

 

It would be interesting to try it with DSD. DSDis a different way to capture the analog signal (super high sample rates at 1 bit) so the math would be different. 

 

My understanding is that, unlike the past, editing can be done in DSD without converting it to PCM. 

 

Unfortunately, I dont have any DSD software. Otherwise, I would be curious to at least try a null test.

[bachish]

4 hours ago, Jud said:

 

I don't care.   

 

I'm just interested in the very nice analysis these two kind folks have done, and looking forward to anything further they're able to do along the same lines.  But to save these people the trouble, if you have got a link to the types of measurements they've done, comparing a DSD256 recording and a 44.1kHz downsampled version, that would be great.  If including DSD is a problem, there are the 2L comparison tracks that would enable comparative measurements of 352.8 and 44.1kHz files: http://www.2l.no/hires/index.html?

 

Thank you, Jud! Interesting. I'll have to check these out. Thanks. 

[bachish]

1 hour ago, crenca said:

 

 

What I find interesting about your analysis @bachish is not your results, but how expected and non-controversial it is.  The core "debate" about digital audio has largely sidestepped and ignored the robustness and repeatability of digital, really by itself, before you even append the qualifier "audio" on the more engineering oriented and computational side of things.  How could they not, for 1+1=2 is what it is, and is as certain as it is banal.  Like you point out and demonstrate, this modern software gives us near "perfect" results, trivial changes to the audio band, etc. etc.

 

However, there is this very large space where there is a debate about the last 10ft - what happens to digital once it is transmitted from what suddenly becomes a very noisy and influential desktop/laptop, through USB (or ethernet or whatever) to a DAC, which despite being a computer itself at least on the front digital side (i.e is a computer itself) is somehow in unexplained terms influenced by all these other factors that did not matter up until this point.  So you have a whole industry of "audiophile" computer gear that ostensibly solves these problems.

 

In any case this is your thread and I will honor your wishes, I just thought your efforts points to certain insights concerning this other "debate", the "sound" of digital which large numbers of Audiophiles claim they here which is apart from the formats (i.e. PCM of any sample rate, lossy vs bit perfect, etc. etc.)...

 

I didn't mean to come off as critical, crenca when I said your post was 'smart @$$'.  It was said more in humor!  I'm pretty 'live and let live'.

 

You do have good points.  I hadn't thought of how this little test would also pertain to the current debates amoung audiophiles.  And I admit I am not up on all the current debates or concerns in audiophile circles regarding digital audio so it is educational to read what you posted. My reading and online posting is more in the recording circles where, in some ways at least, things are less complicated (not that recording isn't complex). But you plug your AD/DA converter in and off you go. You use standard cables by Canare or Mogami, or similar such brands. Most of the big debates, in my experience, have to do with microphone placement.  The integrity of audio files as they pass through digital cables and get processed in the computer seems to be assumed in recording circles. 

 

 

 

 

 

 

[bachish]

59 minutes ago, mansr said:

Creating the DSD stream requires some additional maths, the sigma-delta modulator, but once that is done, the analysis is no different from regular PCM. 

 

Some software can do simple operations like splicing without converting the whole stream to PCM and back. Mixing, or even a simple volume adjustment, is impossible to do directly in DSD.

 

That's not possible. Each sample in a DSD stream can have one of two values, 1 or -1. The difference between two such samples can have four values. You can compute the difference, but it won't be DSD.

Ok good info. 

 

For a test betwen DSD and 44/16 I was thinking one could convert DSD to PCM 96/32 then downsample and dither to 44/16 and back up again to 96/32 and finally to DSD and see how it compared to the original file. 

 

My hypothesis is...it probably would compare quite well to the original in the audible range much like this test did.

[bachish]

7 hours ago, Jud said:

 

There are a few folks in recording circles who fuss about such things, but they are definitely outliers. 

 

Ok, interesting. I suppose it isn't too hard to test. I could take an original recording that is on the SD card in my field recorder that hasn't been transfered yet, upload it to the computer, load it into programs and export over and over, transfer between external hard drives, many times, and upload and download from OneDrive several times. 

 

I could then take it, flip the phase and load it into my DAW and load, just one time, the original from the SD card into the DAW and see if they null. Would be interesting.

 

 

 

 

[bachish]

4 hours ago, cookiemarenco said:

 

We have a "Test your systems" page for free downloads.  One song, 4 formats.  It's free.  The source file is DSD256 recorded to DSD256 and no conversion to PCM / DXD for the DSD256.  The other DSD, WAV, FLAC files were made from the DSD256.

 

Here's the link at Blue Coast Music

https://bluecoastmusic.com/free-downloads

 

I recorded it so I'm familiar with the processes used if you have any questions.

BTW..  we have found ways to keep the editing process in DSD on the Pyramix.  It involves mixing through an analog mixing console.  Sounds great.

 

Enjoy!

Cookie Marenco

founder and producer

Blue Coast Records (produces and owns music) and Music (store for all high resolution labels)

https://bluecoastmusic.com/

 

 

 

 

Thank you, Cookie! I'll check it out 

[bachish]

6 minutes ago, Jud said:

 

And/or you could ask Cookie.   

 

Whether I agree with what she thinks on various matters (bearing in mind she's an experienced producer, I'm just some guy), I really like the quality of the recordings she produces. 

Yes, I'm sure she does excellent work. I'm definitely going to take a listen.

 

You mean ask her to do the test or ask her opinion?