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JRiver vs JPLAY Test Results


mitchco

Recommended reading first The reason is that I am not going to reiterate the baseline components and measurements of my test gear already covered in that post.

 

Here is a high level block diagram of my test setup:

 

JRivervsJPLAYtestsetup.jpg

 

On the left side is my HTPC with both JRiver MC 17 and JPLAY mini installed. The test FLAC file is the same Tom Petty and The Heartbreakers, Refugee at 24/96 that I have been using for my FLAC vs WAV tests.

 

JRiver is set up for bit perfect playback with no DSP, resampling, or anything else in the chain, as per my previous tests:

 

JRiversettings.jpg

 

JPLAY mini is set up in Hibernate mode and the following parameters:

 

JPlayhibernate.jpg

 

On the right hand side of the diagram, I am using Audio DiffMaker Audio DiffMaker for recording the analog waveforms off my Lynx L22 analog outputs of my playback HTPC. All sample rates for the tests are at 24/96.

 

Here is the differencing process used by Audio DiffMaker:

 

AudioDiffMakerProcess.jpg

 

Audio DiffMaker comes with an excellent help file that is worth the time reading in order to get repeatable results. One tip is to ensure both recordings are within a second of each other.

 

As an aside, this software can be used to objectively evaluate anything in your audio playback that you have changed. Whether that be a SSD, power supply, DAC, interconnects, and of course music players.

 

My assertion is that if you are audibly hearing a difference when you change something in your audio system (ABX testing), the audio waveform must have changed, and if it has changed, it can be objectively measured. I find there is a direct correlation between what I hear and what I measure and vice versa. I want a balanced view between subjective and objective test results.

 

First, I used JRiver as the reference and I recorded about 40 seconds of TP’s Refugee onto my laptop using DiffMaker. Then I used JPLAY mini, in hibernate mode, and recorded 40 seconds again onto the laptop. I did this without touching anything on either the playback machine or the recording laptop aside from launching each music player separately.

 

Just to be clear what is going on, the music players are loading the FLAC file from my hard drive and performing a Digital to Analog conversion and then though the analog line output stage. I am going from balanced outs from the Lynx L22 to unbalanced ins on my Dell, through the ADC, being recorded by Audio DiffMaker.

 

Clicking on Extract in Audio DiffMaker to get the Difference produces this result:

 

JRivervsJPlayminihib.jpg

 

As you can see, it is similar to when I compared FLAC vs WAV. What the result is saying is that the Difference signal between the two music players is at -90 dB. I repeated this process several times and obtained the same results.

 

You can listen to the Difference file yourself as it is attached to this post. PLEASE BE CAREFUL as you will need to turn up the volume (likely to max) to hear anything. I suggest first playing at a low level to ensure there are no loud artifacts while playing back and then increasing the volume.

 

As you can hear from yourself, a faint track of the music, that nulls itself out completely halfway through the track and slowly drifts back into being barely audible at the end.

 

According to the DiffMaker documentation, this is called sample rate drift and there is a checkbox in the settings to compensate for this drift.

 

“Any test in which the signal rate (such as clock speed for a digital source, or tape speed or turntable speed for an analog source) is not constant can result in a large and audible residual level in the Difference track. This is usually heard as a weak version of the Reference track that is present over only a portion of the Difference track, normally dropping into silence midway through the track, then becoming perceptible again toward the end. When severe, it can sound like a "flanging" effect in the high frequencies over the length of the track. For this reason, it is best to allow DiffMaker to compensate for sample rate drift. The default setting is to allow this compensation, with an accuracy level of "4".”

 

Of course this makes sense as I used a different computer to record on versus the playback computer and I did not have the two sample rate clocks locked together. The DiffMaker software recommends this approach, but I have no way of synching the sample rate clock on the Dell with my Lynx card.

 

Given that the Difference signal is -90 dB from the reference and that the noise level of my Dell sound card is -86 dB, we are at the limits of my test gear. A -90 dB signal is inaudible compared to the reference signal level.

 

I am not going to reiterate my subjective listening test approach as I covered it off in my FLAC vs WAV post.

 

In conclusion, using my ears and measurement software, on my system, I cannot hear or (significantly) measure any difference between JRiver and JPLAY mini (in hibernate mode).

 

April 2, 2013 Updated testing of JRiver vs JPLAY, including JPLAY ASIO drivers for JRiver and Foobar plus comparing Beach and River JPLAY engines. Results = bit-perfect.

 

June 13, 2013 Archimago's Musings: MEASUREMENTS: Part II: Bit-Perfect Audiophile Music Players - JPLAY (Windows). "Bottom line: With a reasonably standard set-up as described, using a current-generation (2013) asynchronous USB DAC, there appears to be no benefit with the use of JPLAY over any of the standard bit-perfect Windows players tested previously in terms of measured sonic output. Nor could I say that subjectively I heard a difference through the headphones." Good job Archimago!

 

Interested in what is audible relative to bit-perfect? Try Fun With Digital Audio - Bit Perfect Audibility Testing. For jitter, try Cranesong's jitter test.

 

Happy listening!<p><a href="/monthly_2012_05/58cd9bc11cee0_jrivervsjplayanalogdifference_zip.abc5ef36e963925ad0e4deb087100dfd" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28076" src="/monthly_2012_05/58cd9bc11cee0_jrivervsjplayanalogdifference_zip.abc5ef36e963925ad0e4deb087100dfd" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc122aa6_jrivervsjplaydigitaldifference_zip.20206be38ed0e9589a31ef13f8b678e6" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28077" src="/monthly_2012_05/58cd9bc122aa6_jrivervsjplaydigitaldifference_zip.20206be38ed0e9589a31ef13f8b678e6" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc94d683_jrivervsjplayanalogdifference_zip.a113b760512958701d5cb35ef7e6ddac" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28326" src="/monthly_2012_05/58cd9bc94d683_jrivervsjplayanalogdifference_zip.a113b760512958701d5cb35ef7e6ddac" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc9523e8_jrivervsjplaydigitaldifference_zip.2e148f06b06fbf3b249a96e630e6facb" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28327" src="/monthly_2012_05/58cd9bc9523e8_jrivervsjplaydigitaldifference_zip.2e148f06b06fbf3b249a96e630e6facb" class="ipsImage ipsImage_thumbnailed" alt=""></a></p>

64 Comments


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I think the word 'accurate' is the obsession of the digital classes and should not be foisted on the reproduction of audio as a whole.

 

 

 

The are are a great many analogue components in all of our systems. Each and every one of them can be designed and/or constructed in a multitude of equally valid ,but differing, ways. The design and construction of these components will have necessitated a great many, (hopefully!), carefully considered compromises. I do not believe that chucking money at these compromises will make then go away.

 

 

 

So, the most expensive system in the world is still going to be a carefully considered amalgam of compromises. Its sound will differ to that of its next lowly brother by virtue of the differing design choices made as much as the lower amount of money spent building it.

 

 

 

Accuracy in Hi-Fi is best left in the digital domain, where it belongs, it my opinion. Assembling an enjoyable system is not about accuracy at all, it is about the synergy between the listener, the room and the components. Which is about as inaccurate as it gets, I reckon. :)

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I think it is probably our computer audio stuff being basically so 'accurate' whatever we do is the reason many agonise about 'bit perfect', USB cables, minimising processing, merits of various players, etc. None of it makes much difference compared to changing speakers or power amp, but we would like it to. That's part of the fun of turntables, arms, and vinyl, and we don't have that.

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"I think it is probably our computer audio stuff being basically so 'accurate' whatever we do is the reason many agonise about 'bit perfect', USB cables, minimising processing, merits of various players, etc. None of it makes much difference compared to changing..."

 

 

 

There may be one area where computer audio is still a work in progress (other than the minor stuff). I'm only saying this because of what I've read (I don't have the funds or the equipment to test this out for myself) and because I believe that people like PeterSt. of XXHighend are either onto something or completely delusional. That is, NOS dacs that can accept very hi-res input, upsampled via computer (special means) from Redbook cds vs. normal dacs that oversample and may or may not upsample internally. In this case one would need special software such as XXHighend coupled with a special dac like the NOS1, or possibly other NOS dacs that accept 24/384 or higher rates.

 

 

 

My feeling is that the digital file > dac > music, link, isn't quite right yet, and perhaps the PeterSt thing is the closest to it. Too bad I'll never know, except by hearsay.

 

 

 

-Chris

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My “null” test above included the digital to analog conversion and analog line output stage, while recording the results in real-time on a different computer. The reason I did this was to capture any computer noise, jitter, and any other possible artifacts that could be a potential cause for any possible audible differences between the two music players.

 

 

 

As it turned out, even in this worst possible case, including sample rate drift between the playback and recording computers, the Audio DiffMaker result was -90 dB. In other words, an inaudible difference between the two music players relative to the program level.

 

 

 

The DiffMaker test below is using the digital loopback feature of my Lynx L22 sound card: http://www.lynxstudio.com/support_faq_result.asp?c=32

 

 

 

What this means is that I am able to record the digital output of my Lynx L22 directly to the Audio DiffMaker program on the same computer and eliminate any sample rate drift, the digital to analog conversion stage, and the 2nd computer. Here is the result, using the same Tom Petty 24/96 FLAC file that I used for my tests above.

 

 

 

JRivervsJPlaydigitalloopback.jpg

 

 

 

The result, worst case, -133dB. Inaudible. Attached is the Difference file. It is interesting and educational to hear what is left over. I encourage you to listen to the difference because this is exactly what should be left over when comparing two bit-perfect music players.

 

 

 

What does this mean? The two players are identical with respect to bit-perfect playback, on my computer, whether I null test at the digital or at the analog outputs. Or whether I use my ears to ABX, I hear (and measure) no difference whatsoever between the two players.

 

 

 

This is by design: http://en.wikipedia.org/wiki/Bit-perfect “In audio this means that the digital output from the computer sound card is the same as the digital output from the stored audio file. Unaltered passthrough.”

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Love your work here Mitchco.

 

 

 

Very entertaining to see response of those who wish not to believe the results are correct. One does wonder, is there any test that would convince them that reproduction of digital music is even possibly totally transparent. Perfectly done in the sense it has no flaws that a human can discern. Surely they must think at least such a result is theoretically possible. Alas I think for some of them, they do not believe in such a result being possible.

 

 

 

It becomes more and more apparent, that while you need great speakers for great sound, and an amp up to the task, almost everything before it can probably be done for peanuts. Doesn't mean just anything will do. But you likely don't need extraordinary computers or software or anything in the multi-thousand dollar range of upstream components to get audibly transparent reproduction. Nothing wrong with paying extra for convenience, appearance, features etc. Just that you don't need to waste money thinking it buys you better sound.

 

 

 

To me it is very encouraging. You likely won't see many speaker/amp combos that retail for $10k and more fronted by merely $1000 worth of equipment. But chosen carefully nothing may have any higher audible fidelity than such a system is capable of providing. I wish more would get over it and get on board. Then maybe more effort would be spent on things that really do matter like room interactions and such.

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@esldude "But chosen carefully nothing may have any higher audible fidelity than such a system is capable of providing. I wish more would get over it and get on board. Then maybe more effort would be spent on things that really do matter like room interactions and such."

 

 

 

Although you might be right as to your premise, your reason for posting seems not to match your words. IOW, you're being disingenuous.

 

 

 

I don't think less skepticism re digital front ends and/or less work on them would mean more work on room interactions and whatever else you so carefully don't mention. Much work is and has been done on room interactions, possibly more than on what you seem to be talking about; when you think of all the dsp stuff, acoustic traps and whatnot, audio gear mounting systems, and the many books on acoustics and the like not to mention exotics.

 

 

 

Come on, it sounds like you're just trying to apply another dig to those who don't see things your way.

 

 

 

-Chris

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I know this is an old thread, but .......

First I would just like to say that I am by no means an expert.

I just wonder........and I am not even sure if this is feasible, could there be a difference in the sound stage between the 2 players despite the wave forms being beyond human hearing ?

This is a poor example, but a whole bunch of people say one can not distinguish a 320 mp3 from a lossless version, but while I have heard mp3's sound good clarity etc wise, to me at least there is an almost immediate noticeable difference in the sound stage, imaging, depth etc.

As mentioned it is not a good example because with an mp3 you are losing data so is really an apples to oranges example, but I was just wondering if there could be a difference between players ?

I only ask because, to me, the sound stage is really important and I want it as life like as is possible with my system. I am to the point where I am very satisfied with all of my gear, so have now turned my attention to getting the computer set up as good as possible and obviously the player is a very important part and i have been wondering if i should "upgrade" from J river to Jplay.

So, while it sure seems like I should just stick to J river, I do wonder if there is any possible way that the sound stage presented by either player could be different ? I do not see how, but i have learned that I am very, very ignorant when it came to digital music and have been trying to learn a whole new skill set.

Anyway, was just curious.

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Hi donberry,

 

Thank you for your comment. I do understand what you are saying/asking.

 

In my ABX listening tests, I was unable to discern any differences between the two players, over speakers or headphones. I have a new DAC on the way, and this is one of the tests I was going to rerun, but my expectation is the result will be the same. However, I am open to being surprised.

 

There is a demo of JPlay, why not give it a try? Personally, I am happy with the SQ of JRiver.

 

Cheers,

 

Mitch

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JPlay 5.2 beta is out. Sounds great, more concentrated, less clutter than JRiver by itself. As an ex classical musician and current audiophile i trust my ears and i do not stick with products, tweaks whatever that do not work for me. JPlay does work, and i can even hold my interest in 16/44 recordings.

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Hi all and mitchco !

My first post here, sorry to up such an old thread but it's the post that made me come here ;-)

 

Very interesting articles.

 

One think still bothers me : why a difference (even small) between wav and flac or JRiver and JPlay while there should be absolutely no difference ?

 

I guess this should come from the measuring protocol : both DAC and ADC are not syncrhonous and the de-syncrhonization + compensation algoryhtm must generate noise (even if first goal is quite oposite).

 

I think it could be interesting to make a JRiver vs JRiver measure to see if it goes under 90db.

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Hi JAVA Alive, welcome to CA!

 

Yes, one of the difference factors indeed is the DAC and ADC not being synced, plus issues with DiffMaker's drift compensation function. However, listening to the analog diff file, there is a point where the music is no longer audible. Other noise factors include D/A and A/D conversion and associated analogue circuitry. But I would say the biggest contributor was my test laptops analog input/ADC is noisy. You can see the RMAA test in the previous blog post (FLAC vs WAV) with a noise level of - 85 dBA.

 

You also may find Archimago's DiffMaker protocol interesting and his results correlated virtually identical to mine. Archimago also tested a number of Windows and Mac software players with similar results, including JPlay.

 

Happy listening!

Mitch

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Hi,

 

 

Yes, in fact the FLAC vs WAV is already a test of "A" vs "A". ;-)

Who ever prentends a diffrence could exist is deeply missunderstanding what is behind playback in a computer.

 

The only potential difference is the impact of CPU usage on radioelectric noise and power supply.

And people who whant to take care about this should first invest in high quality mother board (like the ones dedicated to overclock) and high quiality power supply and, most important, undervolt and undeclock the CPU.

The underclock and undervolt is a much higher factor to reduce radioelectric noise and power consuption than using 0.5% instead of 1.5% of CPU (which is the arround difference between wav and flac).

 

One other point I don't understand is why such high consideration on latency (most of the time Asio buffer size).

This also has absolutely no impact on hifi listening. This factor is important only when you syncrhonize music form PC with other sources (like midi synth) or when you do multiple tracks listening and recording on the computer.

You need very low latency in this case. If not, you may create time gaps between tracks which can alter sond (like flanger sounds) and even tempo if latency is realy high.

 

For hifi, this has abslolutly no impact and buffer should be at the highest value to avoid drops.

 

Last point, I have a smal schema to show the loss of a DAC + ADC conversion signal distorsion :

Sinus - HostingPics.net - Hbergement d'images gratuit

 

On this schema (purely theorical and made under XLS) you can see in red a 10khz sinus sampled at 44.1khz.

In blue you see a re-construction of this signal based on a theorical re-sample at 44.1khz but clocks are not synchronized (clock of blue signal has a delay of 1/2 sample, which is arround 11.3 µs gap).

With this 1/2 sample gap, it is easy to calculate the blue line : bluesample(n) = (redsample(n)+redsample(n+1))/2

 

You see on the picture that the "0" of each chart are not alligned, this is notmal, it is the 11.3 µs gap.

But if you re-align the "0", you see that easily bye eye that charts won't match at all.

With a 96khz sample rate, difference will of course be samller but I prefered to use 44.1 to be more "eye friendly".

 

This means the measure protocol you use generates, even without consideration of noice and components limitation, a gap between original sample and final sample.

 

A a consequence, you have proven that difference is UNDER -85 ou -90db (and not equal).

I am prety sure that, in fact there is no difference except on potential impact of CPU usage on radioelectric noise and power supply.

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