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Down-sample DXD 352 to 192 or 176, and 96 or 88???


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So here is the question.

Yesterday I downloaded the stunning DXD 352 recording of Carmen Gomes plays the Blues.

I also bought the flac 96 version for my portable media player.

Now a friend of mine claims that the label would have done better if they had down-sampled to flac 88 as that is a doubly even multiple.

Sound Liaison are selling 352, 192 and 96 pcm or flac as well as DSD 256, 128 and 64 .

 

I believe that they know what they are doing as I find their recordings among the very best available, however another favorite label of mine, Reference Recordings, seems to be doing 176 so ....? are Sound Liaison missing something? 

CSTB300shadowv2.png Carmen Gomes sings the Blues

 

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Integer resampling ratios can be less computationally intensive to perform, but there's no difference in quality if done properly. I assume whatever software they used is decent.

 

The real question here is why you didn't downsample the files yourself rather than buying the album twice.

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2 minutes ago, mansr said:

Integer resampling ratios can be less computationally intensive to perform, but there's no difference in quality if done properly. I assume whatever software they used is decent.

 

The real question here is why you didn't downsample the files yourself rather than buying the album twice.

convenience, and same friend told me that the converters at the various labels are mostly better than my jriver...

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3 minutes ago, PAP said:

Sox. Ok I will look into that, thanks!

But if you had the choice and were going to download one of those down samples would you choose 192 or 176,

or indeed do it yourself ... you would convert to 88 or 96?

If buying I'd pick the cheaper one, or the higher rate (just in case) if they were the same price. There's no reason the quality should differ.

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I've talked to several professionals who don't think the integer rate thing makes a difference in the sound of the final result.

Main listening (small home office):

Main setup: Surge protector +>Isol-8 Mini sub Axis Power Strip/Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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Several years ago there was a strong and long discussion in this forum regarding this, where several recording engineers and audio software developers participated. I can't find the thread !
The answers (as always) were controversial.

I personally (and a group of friends) prefer to respect the multiples. The same regarding our preferences toward Native DSD vs DoP.

At this time I try to download the version at 176 (the server is very busy :) ), before, I downloaded the 128 DSD. 

Unfortunately there is no information from Sound Liaison if the simultaneous recording on analog tape was used as a source for some of the formats.

 

Roch

 

PS/ Please note that is an opinion only, based on taste, and not to start an never ending discussion !!!

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4 minutes ago, elcorso said:

Several years ago there was a strong and long discussion in this forum regarding this, where several recording engineers and audio software developers participated. I can't find the thread !
The answers (as always) were controversial.

I personally (and a group of friends) prefer to respect the multiples. The same regarding our preferences toward Native DSD vs DoP.

Could you describe what you compared and the difference you perceived?

4 minutes ago, elcorso said:

At this time I try to download the version at 176 (the server is very busy :) ), before, I downloaded the 128 DSD. 

Unfortunately there is no information from Sound Liaison if the simultaneous recording on analog tape was used as a source for some of the formats.

You could ask them.

4 minutes ago, elcorso said:

PS/ Please note that is an opinion only, based on taste, and not to start an never ending discussion !!!

My opinion is based on maths.

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1 hour ago, elcorso said:

Several years ago there was a strong and long discussion in this forum regarding this, where several recording engineers and audio software developers participated. I can't find the thread !
The answers (as always) were controversial.

I personally (and a group of friends) prefer to respect the multiples. The same regarding our preferences toward Native DSD vs DoP.

At this time I try to download the version at 176 (the server is very busy :) ), before, I downloaded the 128 DSD. 

Unfortunately there is no information from Sound Liaison if the simultaneous recording on analog tape was used as a source for some of the formats.

 

Roch

 

PS/ Please note that is an opinion only, based on taste, and not to start an never ending discussion !!!

DoP contains exactly the same bits as the native DSD. All it does is store 16 bits in a 24 bit PCM wrapper so it can be sent to the DAC. The original data is extracted from the PCM wrapper by the DAC. The DAC receives the exact same bits as in the original file. In that regard  DoP quite different from PCM downsampling, since in that process the bits are definitely changed..

 

If you prefer native DSD over DoP it must be because of the effect the receiving and extraction process  has on the working of the DAC. I agree that this is not the place to discuss that issue, I just want to point out that the bits the DAC receives are exactly the same as in the original file. Again, that is a different situation than downsampling.

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I doubt it makes much sonic difference these days, but I see no reason not to go by the played file's native sampling rate or even multiples thereof. It cannot possibly hurt to do so, and I see nothing whatsoever to be gained by not doing so.  That is, provided your playback system is agile and can handle it.  

 

I do use Sox in JRiver, and I am limited to something at or below 192k PCM in order to use Dirac Live, which I prefer to do, although my DAC goes to 384k  and could handle up to DSD256 natively.  Most of my listening is to DSD64 rips from SACD, but also some DSD128 or DSD256 downloads, and I convert those on-the-fly to 176k PCM, as well as for PCM352k.  For PCM384k, I downrez to 192k, of course.  For all else, I just use the native rate of the file 44,48,88,96,176,192k.  I seldom play CDs, so I have not investigated whether uprezzing them to 88 or 176k might offer any sonic advantage.  

 

It is quite easy to set up all those rules in a table in JRiver so that it happens automatically exactly as you wish.  I also see or hear no credible reason to believe that JRiver sounds inferior to other playback tools - it's an unproven myth.

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24 minutes ago, Fitzcaraldo215 said:

I doubt it makes much sonic difference these days, but I see no reason not to go by the played file's native sampling rate or even multiples thereof. It cannot possibly hurt to do so, and I see nothing whatsoever to be gained by not doing so.  That is, provided your playback system is agile and can handle it.  

Exactly, apparently some systems require significantly more CPU effort to up- or down-sample between non-integer multiples than between integer multiples.  With hig-rez and multichannels, the difference can choke the processor leading to interruptions in the playback.   The difference can also be measured with any number of tools.

 

However, when all runs smoothly, I do not hear a difference between non-integer and integer conversions.

Kal Rubinson

Senior Contributing Editor, Stereophile

 

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1 hour ago, Kal Rubinson said:

Exactly, apparently some systems require significantly more CPU effort to up- or down-sample between non-integer multiples than between integer multiples.  With hig-rez and multichannels, the difference can choke the processor leading to interruptions in the playback.   The difference can also be measured with any number of tools.

 

However, when all runs smoothly, I do not hear a difference between non-integer and integer conversions.

The original question concerned offline resampling where CPU load is much less of an issue.

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26 minutes ago, mansr said:

The original question concerned offline resampling where CPU load is much less of an issue.

Even so, what is the advantage of converting, say, 352K DXD to 192k,  as opposed to 176k,  or to 96k vs. 88k?  I am not saying it will make a noticeable sonic difference.  It just makes more mathematical sense to stick with the integer multiple idea, and no possible harm can come from it.

 

I am saying if you have to convert, convert!  If you have to up or downrez, do it!  But, you might as well select a sampling rate that has an integer multiple relationship.  There is no reason not to, just in case.  

 

I must confess that I do all my conversions on-the-fly.  There have been no resource constraints of any kind that I am aware of.  I personally think offline resampling would be a waste of time and effort with no benefits.

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The idea that integer downsampling retains many of the original data points is why people think integer downsampling must be better than non integer down sampling.  In fact, modern methods use detailed interpolation algorithms that are far more accurate than just dropping intermediate data points.. They use the trend, not just the individual points.  You really have to look at the actual algorithms to be sure, but most people  who know the algorithms say integer downsampling is no more accurate than non integer downsampling.

 

That said, sometimes non integer downsampling can take more CPU time, so, integer downsampling may be preferred from a CPU perspective when doing downsampling. 

 

Unless their is something specific about your DAC that prefers a specific sampling rate, I would do integer downsampling,  just in case their is a difference and to minimize CPU time.

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3 hours ago, mansr said:

Simply dropping samples results in aliasing of the high frequencies into the remaining band. To avoid aliasing, a low-pass filter must be used. Integer downsampling can be regarded as a low-pass filter followed by dropping samples. A practical implementation is simply a low-pass filter with a short-cut to calculate only every N samples for a factor N downsampling.

 

Downsampling by a non-integer ratio is a little trickier since only some output samples coincide in time with input samples. A common method is a polyphase filter. Recall that the output of a filter is the convolution of the input with the impulse response of the filter. To calculate the output value between samples points of the input, we can simply perform the usual convolution with the impulse response evaluated at corresponding inter-sample positions.

 

As an example, suppose we wish to reduce the sample rate by a factor of 2/3. With an input sample period of T, we get an output period of 3T/2 and samples at times 0, 3T/2, 3T, and so on. Every second output sample coincides with every third input sample, while the rest fall in between. The first output sample is calculated by convolution of the input with the impulse response as usual. To obtain the second output sample, at 3T/2, we first evaluate the impulse response function at T/2, 3T/2, 5T/2, etc, then convolve the input with this sequence. The third output sample is aligned with an input sample, so we go back to the normal impulse response. We then carry on alternating between the two impulse response sequences for the remainder of the signal.

 

This approach works for any rational resampling ratio. In the example above, we needed only two impulse response sequences (or filter phases, hence the term polyphase) since the ratio is simple and every second output sample lines up with an input sample. More generally, an N/M ratio requires N filter phases. For example, resampling from 48 kHz to 44.1 kHz, a ratio of 147/160 needs 147 different convolution sequences. Integer downsampling is just a special case with N=1.

 

In practice, it is common to pre-calculate the impulse response for the required phases and store all the sequences in RAM. For very long filters, this can end up needing several megabytes of space. If this is larger than the CPU cache, it will impact the performance though it's difficult to predict how much.

 

As for accuracy, the actual calculations are essentially the same regardless of the resampling ratio. There is thus no reason for any ratio to produce more accurate results than another.

Thanks for the very thorough and knowledgable explanation.  

 

It seems the consensus and my own operational practice agree that, in case downsampling is necessary, it cannot hurt to use integer downsampling, and there is nothing to be gained by not using it in that case.  It probably does not make a big sonic difference either way, unless computer resources are extremely tight in on-the-fly conversion, where integer downsampling may be preferred over non-integer.  But, no downsampling may be best wherever possible.

 

Does that seem fair?

 

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2 minutes ago, Fitzcaraldo215 said:

Thanks for the very thorough and knowledgable explanation.  

 

It seems the consensus and my own operational practice agree that, in case downsampling is necessary, it cannot hurt to use integer downsampling, and there is nothing to be gained by not using it in that case.  It probably does not make a big sonic difference either way, unless computer resources are extremely tight in on-the-fly conversion, where integer downsampling may be preferred over non-integer.  But, no downsampling may be best wherever possible.

 

Does that seem fair?

All else being equal, choosing integer ratios will indeed not hurt anything, so by all means do that if it makes you feel better.

 

Btw, the explanation above applies with only minor changes to upsampling as well.

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On 01.05.2017 at 1:04 AM, PAP said:

Yesterday I downloaded the stunning DXD 352 recording of Carmen Gomes plays the Blues.

I also bought the flac 96 version for my portable media player.

Now a friend of mine claims that the label would have done better if they had down-sampled to flac 88 as that is a doubly even multiple.

Sound Liaison are selling 352, 192 and 96 pcm or flac as well as DSD 256, 128 and 64 .

 

I believe that they know what they are doing as I find their recordings among the very best available, however another favorite label of mine, Reference Recordings, seems to be doing 176 so ....? are Sound Liaison missing something? 

 

 

There are no difference between multiple/non-multiple resampling.

Because used "magic" coefficients 160 and 147 (noted above) with integer multiplication and division.

More about:

- oversampling http://samplerateconverter.com/content/how-convert-sample-rate-oversampling

- downsamplind http://samplerateconverter.com/content/how-convert-sample-rate-divide

 

For both cases main matter how implemented resampling filter.

 

Design of resampling filter is balance between:

- suppressing aliases, and

- ringing level, and

- calculation speed.

 

Thought ringing is not so big evil as seems at first sight.

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On 1/5/2017 at 0:04 AM, PAP said:

So here is the question.

Yesterday I downloaded the stunning DXD 352 recording of Carmen Gomes plays the Blues.

I also bought the flac 96 version for my portable media player.

Now a friend of mine claims that the label would have done better if they had down-sampled to flac 88 as that is a doubly even multiple.

Sound Liaison are selling 352, 192 and 96 pcm or flac as well as DSD 256, 128 and 64 .

 

I believe that they know what they are doing as I find their recordings among the very best available, however another favorite label of mine, Reference Recordings, seems to be doing 176 so ....? are Sound Liaison missing something? 

CSTB300shadowv2.png Carmen Gomes sings the Blues

 

On the site there is 176 download available did anyone compare that to the 192?

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