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[rdfornasero]

Hi everyone,

I recently upgraded to version 17.0.27 and noticed the ability to change the output sample rate from various input sample rates.

Does re-sampling (up-sampling) affect sound quality?

The reason I ask is, my music library consists of mostly 16 bit 44.1 kHz flac material but I also have some 24 bit 88.2 kHz and 96 kHz hi-res material as well. All my hardware is compatible with the various sample rates but when my Integra DHC-80.2 pre/pro receives a different sample rate is makes an audible click and playback is delayed a couple of seconds for hardware synchronization. If I set MC to output 96 kHz for the 44.1 and 88.2 kHz material there are no hardware clicks and sync issues however I'm concerned that sound quality is compromised. Will re-sampling the 44.1 and 88.2 kHz material to 96 kHz affect the sound quality?

Any help would be greatly appreciated.   

[jmone]

Short Answer:  No (MC is very good with it's resampler so it will sound the same but without the annoying click from the receiver)
Longer Answer:  Your mind may tell you they sound different just because you know you have changed a setting, so before others weigh in in the "Longest Answer", the only way you may be able to convince yourself is to do a blind test with the settings.
Longest Answer:  To be provided by others.....Some claim to hear the difference in just changing the Power Plug to a bigger one, so for them I'm sure they would also hear the impact of resampling....

[Tomk]

I think a lot of it has to do with providing the sample rate your device expects or works best with.  So if your device is technically capable of downsampling a 192khz sample to it's maximum rate of 96, but the capability to do so was an afterthought of it's designer, then letting something else (like MC) re-sample might give better overall results.  

So I guess the answer is, "it depends" and "you have to try it".

I used to have a CD player who's most celebrated feature was "defeatable oversampling".  I found that on some CDs the oversampling worked wonders and on others it made no difference or maybe even made the music sound a little "blurry" (warning audiophile jargon).

So having the ability to switch on and off was great.

To get the best overall answer would require you to know what alorythms are used in your device and MC so you could choose the one with the "best" approach.

Some devices are cable of just running at different rates, in which case I would bet any sort of bit fiddling would lead to "worse" sound even if you couldn't hear it.

[justsomeguy]

I'm don't know to much of the technical details about sample rate and bit depth but have always thought it was best to leave the audio in the format it is originally in. However recently I've been questioning that if up sampling the audio before applying any dsp effect would be better. If you aren't applying any kind of dsp then leaving it alone is probably still best.

[psam]

...

Does re-sampling (up-sampling) affect sound quality?
........

I am also doing some listening tests on that, and while they are not yet conclusive, here is my 2cents:

Upsampling / Oversampling is a very popular process in modern DACs, it involves stuffing the missing data on the added bits using some algorithm.

The individual algorithms result in different sound results. Not only from DAC to DAC, but even on the same DAC, selecting different "filters'.

Some people don't like the idea of upsampling, thus some manufacturers support the so called NOS (Non OverSampling) DACs.

It all comes down to whether the upsampling algorithm of the Media Center is better than the upsampling algorithm of the DAC!

I am not through my listening tests using

a) Media Center upsampling everything to 24/192 and sending it through USB to my Antelope Zodiac+ DAC, which accepts 24/192 through its mini-USB port (This solves the clicking on the DAC)

b) Media Center sending the standard 16/44 data stream to the Zodiac where it is upsampled to 24/192

The next step will be to use my Sound Card's (ASUS XONAR Essence STX) SPDIF output to feed the Zodiac.

Another concern of mine is jitter.

If the upsampling algorithm of the DAC is at least as effective as the one on Media Center, I suppose that sending 24/192 over the Toslink or SPDIF Coaxial will induce more jitter than sending a simple 16/44.1. Various tests have shown that (especially cheap) digital cables don't perform very well with 24/192 streams.

If this is true, it may be a bad idea to send upsampled streams to a mediocre DAC where jitter is not properly addressed.

I'll keep you posted....

[Bazza]

I am currently using the trial version of media center and cannot find the re-sampling feature.  Is it missing from the trial version?

Also when I use an external re-sampler and then put the files back into media center to burn on a DVD the tracks are out of order no matter if I queue them in the proper order.

[Blaine78]

personally don't like upsampling. to me, it sounds a little strained and less 'life' in most cases. this is upsampling in general, not just JRiver. this depends also on how revealing your system is, and how much you listen in on it. some wont notice.

[Trumpetguy]

I am currently using the trial version of media center and cannot find the re-sampling feature.  Is it missing from the trial version?

You will find it in the DSP studio - Output format. Trial version looked identical to registered version to me.

[Bazza]

Thanks very much.  Any advice on keeping track order when burning DVDs after reampling externally and then inporting the file to MC?

[EJR]

 I have been experimenting with it.
it does sound different at least it did with the song I was listening on my system.
I let my girlfriend listen to  the same song upsampled and normal without telling her anything. she also heard a difference and like me preferred the non upsampled version.
I have to experiment more with other music
the difference was Small and hard to describe i'd say upsampled sound a little smoother and normal sample rate a bit more direct
its hard to put it to words

will need more experimenting to come to a final conclusion



I'm using a metrum acoustics Octave NOS Dac.
http://www.nosminidac.nl/Octave.html
and various articles on the net said higher res feeds will sound better because it will move certain undesirable side effects that you get without filtering out of the hearing range.
according to that upsampling should actually work nice for this type of ac

[Pjotr]

There is no simple yes or no to this matter. Down sampling and up sampling is always a basic part of the digital audio game. Whether you do it in software (at the player level) or in hardware (at the DAC level). The results are the same.

Up sampling and down sampling also invokes filtering (either in software or in hardware). It is at this filter level and how this is implemented where the differences arises. I.e. basically studio material is recorded and edited at 24b/96kHz. To put it on CD it must be down sampled and limited to 16b/44.1 kHz. This is usually done in software on a PC/Mac. Part of the process is to low pass the signal to limit the content below 22 kHz first to avoid aliasing. There are many pro down samplers for this purpose on the market and they all differ in how this low pass filtering is done and can sound different. This is the same game as the low pass filtering in a hw ADC as they can also sound different.

There is a long long debate going on what is the best filter for a DAC: Must it be technically correct or must it be musically correct or should we leave it out completely (NOS DAC's without filter) and what is what? It made Wadia big in the past.

IMHO with a good quality DAC up sampling in software at the player is of no use, exempt you want special dedicated low pass filters. But these filters must also be implemented then. For me it is only wasting processor power.

[Vincent Kars]

Whether you do it in software (at the player level) or in hardware (at the DAC level). The results are the same.

Upsampling depends on the algoritm used and the precision.
In general upsampling in hardware uses a DSP chip.
The cheap ones are  a bit down in power. This limits the amount of tabs.
They have a limited word length. This increase the quantization error.

At a PC there is no problem performance wise to do the entire calculations in 64 bit and use more tabs.
Uli Brueggemann did a nice experiment.
He up- and down-sampled the same signal using different tools.
Some examples can be found here: http://www.aktives-hoeren.de/viewtopic.php?f=30&t=2242&start=44

Obvious not all sample rate converters are created equal.

[Pjotr]

Whatever, but you simply get no more information from a 16b/44.1kHz perfectly ripped CD with up-sampling. So my point: Use simply the native sample rate of the source material at hand. Modern DAC chips are vary capable to do the filtering themselves.

[pcstockton]

Hi everyone,

I recently upgraded to version 17.0.27 and noticed the ability to change the output sample rate from various input sample rates.

Does re-sampling (up-sampling) affect sound quality?... however I'm concerned that sound quality is compromised.

Any help would be greatly appreciated.   

As with any "better or worse" scenario, your best answer is your own ears.  Why not simply demo it yourself (and please let us know the results)?  Assuming you have been listening to the native bit-rate for a while, try upsampling for a few weeks and you will have your answer.  If things work as designed I am guessing you wont notice a difference.  But what other people think should be the last thing you should trouble yourself with.

Some people are very vocal and passionate about these activities on both sides of the question.  Dont confuse yourself with their polemics.  Try it for yourself.

Good luck!
Patrick

[Pjotr]

Some people are very vocal and passionate about these activities on both sides of the question.  Dont confuse yourself with their polemics.  Try it for yourself.

Good luck!
Patrick

Sure, luckily we are not in the "Don't try this at home" playground here  

[swm]

Is it possible to know the technical characteristics and the parameters of the up-sampler and DSD to PCM converter?

Thank you.

[JimH]

Is it possible to know the technical characteristics and the parameters of the up-sampler and DSD to PCM converter?

Thank you.

I'm sorry, but we don't provide that information.

When you want to post on a different subject, use "New Topic" in the upper right corner of the board's index page.

[Pjotr]

Why not Jim? Too much effort or something to hide?  :just joking:    All DAC chip manufactures are not scared to do so and specify in great detail the properties/behaviour of their components. But I have no doubts MC does not do that properly.

I don't think it is that important. I have experienced that hardware implementation of a DAC by itself has much more impact than things like (pre) up sampling at the player level. And then I mean at the board level of a DAC, not a power plug or cable such or so. I.e. noise and artefacts on the local power rails at board level has much much more impact on the sound quality such as "Airiness", "Placing", "Blankness", "Spaciousness", "Detail" or what people may call it.

Note that even if you pre up sample to 192kHz in software, you don't defeat the own filtering and up sampling to a much higher level of modern DAC chips in hardware. It is simply part of the nature of modern DAC chips and there will always be an amount of signal processing in the chip.