Personal Observations on Resolution, Sample Rate and Data Compression

Posted by: Huge on 28 July 2015

I've just been making some tests using a set of files of a clarinet concerto, using different data encodings, converted using dBPoweramp.

 

1  24/192 (the source file)

2  24/48

3  16/44.1

4  16/48

5  MP3 variable rate

6  MP3 320kbps

 

All files are played from a USB memory stick.

Files 1 - 4 are WAVE format.

Files 5 & 6 are MP3

 

 

Characteristics observed to change

 

A  There is a sudden orchestral FF tutti a few seconds into the piece, in good conditions this is sudden enough to make me 'jump' a little.

 

B  The solo clarinet is distinctly separate from the orchestra in timbre, it being closer to the microphone.

 

C  There is a difference in the perceived location of the orchestra and the soloist including general information from the 'sound environment'.

 

D  There is a subtle difference in the reverberation of the orchestra and the soloist due to their different location in the hall.

 

 

Results in comparison to file 1

 

2  24/48

A   The sudden jump at A is very slightly softened, but I still 'jump'.

B,C,D   Changes are below the threshold of certainty.

 

3 16/44.1

A   The sudden jump at A is considerably softened, to the point where I no longer 'jump'.

B,C   Changes are below the threshold of certainty.

D   The difference in the reverberation is no longer detectable

 

4 16/48

A   The sudden jump at A is somewhat softened (closer to 24/48 than to 16/44.1), I still 'jump' slightly.

B,C   Changes are below the threshold of certainty.

D   The difference in the reverberation is no longer detectable.

 

5  Variable Rate MP3

6  320kbps MP3

A   The sudden jump at A is considerably softened, to the point where I no longer 'jump'.

B   The individual timbre of the clarinet is less distinct from the body of the orchestra

C   The presentation is considerably 'flatter'.

D   The difference in the reverberation is no longer detectable

 

 

These are just my observations based on one movement of a concerto; but, it was chosen to show up potential issues, and I believe the original 24/192 to be a good recording:

Weber Wind Concertos, SCO, Janiczek.  Linn CDK 409

 

Posted on: 28 July 2015 by Bert Schurink

I think this sums it up very well. I also have heard real A-B comparisons and it's indeed different at higher qualities. I normally describe it as more air around the instruments, more tonal aspects, more natural.

Posted on: 28 July 2015 by SamS

Thanks for the analysis Huge, but why no 24/96? Based on the 24/48 conclusion it is likely that there will be little or no perceptible difference between 192 & 96 and thus paying extra for 192 material is probably rarely justified.

Posted on: 28 July 2015 by Huge

Hi SamS,

 

I wasn't trying to test all commercial distribution formats.

 

The reason for using both 16/48 and 16/44.1 was to test for the effect of the interpolation in the time domain - 16/48 is an integer ratio of 24/192, 16/44.1 isn't.

 

At some time I may try comparisons to 24/96, 24/44.1 and 16/96 to complete the picture.

 

 

I also don't know of anyone distributing variable rate MP3 as quite a few players can't handle that format.

Posted on: 28 July 2015 by james n

Interesting post Huge - I do find that i can tolerate lower rate files on certain types of music. In the end out of the different file types i have, mastering is the key. I've got some Red Book, that you'd swear was Hi-Res, it's that good. 

 

Thanks for posting your observations

 

James

 

Posted on: 28 July 2015 by Huge

James,

 

You're absolutely right, no technical encoding of the data can do anything to make up for poor sound engineering.  One of the reasons for picking that recording was that it's very well engineered as well as being a dynamic performance.  Using a good master (technically and musically) was essential to getting a valid set of lower order encodings to test.

 

If it's a choice between: a good performance and well recorded/engineered with red book encoding or a less good performance and/or not such a good recording/engineering technique but with high res encoding; then in all cases I'll take the better performance and recording/engineering over the technical capability of the container.

Posted on: 28 July 2015 by Bart

Where is the raw data from the linear potentiometer regarding the magnitude of your "jump?"  Absent that . . .

 

Posted on: 28 July 2015 by Huge
Originally Posted by Bart:

Where is the raw data from the linear potentiometer regarding the magnitude of your "jump?"  Absent that . . .

 

The dog left it in the oven and it got cooked!

Posted on: 28 July 2015 by Simon-in-Suffolk

Hi Huge, sounds like you were having fun .

you might want to try with different down samplers and down sampling algorithms.. You will almost certainly get differing subjective assessments.

As far as MP3 encoding, the LAME encoder was always the master encoder in my expierience and in the mid 2000s I encoded much of my then collection with LAME, and today those encodings sit head and shoulders above anything I have since downloaded at identical data rates and a country mile away from iTune's in built encoder when ripping CDs.. LAME did allow a huge amount of tuning to optimise the encoding for the music type and the stereo field type.. Again you can increase accuracy for a particular master file.. So it's worth playing with the settings.. the variable rate is also interesting as you can change thresholds and how aggressively the bandwidth changes... Again all this can make the end effect more accurate at the expense of data reduction.

Simon

Posted on: 28 July 2015 by bicela

Very good post Huge, thanks. I'm doing also some test with DSD down sampling because I can not use it on my Uniti but some materials I would listen will probably be available in this format.

 

With my surprise I get the best result without following the integer ration in the decimation.

 

I hope in future someone share his experince also in this complex transcoding process.

Posted on: 28 July 2015 by Aleg
Originally Posted by Simon-in-Suffolk:

Hi Huge, sounds like you were having fun .

you might want to try with different down samplers and down sampling algorithms.. You will almost certainly get differing subjective assessments.

...

Simon

DBpoweramp is apparently one if the best resamplers out there. On both frequency, phase, passband, transition and impulse behaviour.

Mr Spoon just pointed to a new test of his 15.1 resampler on src.infinitewave.ca 

It behaves even slightly better than Adobe Audition.

 

So not a bad choice at all by Huge and not one to better with another one.

Posted on: 28 July 2015 by Simon-in-Suffolk

Aleg, yes dbpoweramp is good, but there are many resamplers or SRCs out here and most vary in slightly different ways, as ultimately down sampling is a lossy process.

http://src.infinitewave.ca

It compares most of the SRCs out here going from 96 to 44.1kHz..  So it's another case of choosing your poison to a large extent.. It is about a subjective assessment and personal preference...

Simon

 

 

Posted on: 29 July 2015 by Huge
Originally Posted by AllenB:

Another interesting comparison might be starting with a Redbook source file at 16/44.1 and doing this careful considered assessment, particularly on the upsampled avenue.

Given that the Naim streamers (and CD players) upsample red book data in their hardware during playback and we (presumably) like the Naim sound signature, I think it unlikely that we'd find an SRC to upsample the data in a way we'd find substantially preferable to the Naim firmware.

 

However, having said that, there may be something that shifts the acoustic signature subtly, adjusting it to personal taste, so there may still be reason to look at this approach.  Are you offering to run the tests and publish your results?