[manueljenkin]

Don't bother. Read first replies to this thread, make your own judgement. If you doubt, make your own test system, take data stream from S/PDIF output and feed it into a suitable sound card with S/PDIF input. Capture data stream and compare it bit-by-bit with the original. If doesn't compare, then your player is cheating on you, or there are dropped frames during transmission. Stereophile made such tests, it is easy to detect dropped frames as opposed to pre-processing on the player.

Do you have links to the test. I'm curious to see the results.

 

[redrol]

The results are and always have been bit-perfect.

@manueljenkin You are using off the shelf PCs that have no real low level control. Try a custom PC with a good motherboard for a change.

 

[Roseval]

it is easy to detect dropped frames as opposed to pre-processing on the player.

To the best of my knowledge, the procedure is the same.
Play a file (make sure to avoid the Win mixer otherwise there will be differences for sure) and record it.
Load original and recording in a audio editor
Time align
Substract.
The result should be zero.

If some DSP is going on, the difference wil start in the first sample
If a frame is dropped, from that frame on they will differ.

More elaborate: https://audiophilestyle.com/ca/bits...gital-audio-–-bit-perfect-audibility-testing/

 

[manueljenkin]

The tools are free, if the OP likes them he can try and decide for themselves, not sure what's your problem in that?

I have no obligation to share my pc build and components to you.

 

[ra990]

Before I switched to Roon, I was really happy with MusicBee, make sure to check that out - free, hi res/multichannel support, ASIO/WASAPI, really nice UI, tagging features, rock solid stability, has nice winamp style visualizers (if you're into that). https://www.getmusicbee.com/

 

[originalsnuffy]

I would agree that a Windows music player has to try hard NOT to be good. I typically use either Foobar or JRiver.

Best audio improvement every for me was to get a quiet power supply in the PC. I have a Corsair and its quite quiet. I had already put in a quiet system fan and my hard drives are barely audible.

 

[tentoes]

I've tried many free media players but always comeback to Foobar2000. It has veritable cornucopia of features and plugins. Some are gimmicks, but I have found a number of useful features such as the remote media server and the music database. I run SPDIF from a nine year old motherboard to a cheap AK4396 DAC and class A Beyerdyanamic clone amp which seems quiet enough wrt signal noise.

 

[Pondoro]

Thanks everyone. I appreciate all the responses. What I have learned is that there are:
1) No music players that are clearly higher quality (at least none have developed a crowd of followers here).
2) Many free players that have happy users.

I plan to try a few but with low expectation that I will hear a discernable difference. I don't plan blind testing and I do not plan any digital measurements. Unless I can easily hear a difference they are "all the same" to me.

I agree with Mike Moffat that "Hearing is integrative" so I plan to use each specific player for a while, not make snap judgements. But if one sounds really good or really bad initially I will note that. My agreement that hearing is integrative is based on many experiences with musical instruments. In one case I was in one room and my friend was in another room. He was playing a ukulele and there were several available, I noticed instantly when he started playing a ukulele of mine. I liked the sound of that uke but I had never thought it had a unique sound. Still I correctly identified it, with no expectation or warning. But months of playing it had taught me how it sounded without me knowing. Likewise I changed a sound card in a computer with little expectation that the sound would change (a specific piece of software was interacting badly with the built in sound card so I bought an add on to fix that.) The change in sound was noticeable, again it came as a surprise to me.

 

[sajunky]

It is a music player thread, but once a matter of music players is resolved, it needs a comment. Once a digital data stream (bit-perfect when it done right) leaves PC, it is subject to jitter. Jitter is inserted to a data stream on the source (PC) or caused by a noise affecting a receiver's internal operation. Once it is in, it is difficult to remove, never 100%. Audiophile sound cards deal with it somehow, but are expensive and not always give good results. Not to menion Audiophile PCs (despite what others claim), is a complete nonsence. PC is always a biggest source of ground loops in our system.

The other approach is to chose a DAC featuring USB interface synchronised with a high precision internal clock (not in reverse!). It works like that: all new packets arrive from the PC on a specific feedback request from a DAC. It require just inexpensive FIFO buffer. A small FIFO buffer, nothing else. Some implementations still do reclocking, while it is not needed in such situation, as reclocking is not 100% jitter free. I don't know why they do that, speak to them... However galvanic isolation is needed in most of cases, but it is not a common feature, it is why special devices were created. It is called DDC like DI-20 or DI-20HE. It alllows to separate a DAC from a biggest source of noise which is PC. To get the most of a DDC a DAC should support a new interface which becomes popular, called I2S. The alternative is S/PDIF coupled with external clock synchronisation, but it was reserved exclusively for high-end DAC's. I2S is a feature everybody should look for.

The alternative to DDC is a low powered network streamer with I2S output, but it require to use specific software like Roon.

 

[redrol]

It is a music player thread, but once a matter of music players is resolved, it needs a comment. Once a digital data stream (bit-perfect when it done right) leaves PC, it is subject to jitter. Jitter is inserted to a data stream on the source (PC) or caused by a noise affecting a receiver's internal operation. Once it is in, it is difficult to remove, never 100%. Audiophile sound cards deal with it somehow, but are expensive and not always give good results. Not to menion Audiophile PCs (despite what others claim), is a complete nonsence. PC is always a biggest source of ground loops in our system.

The other approach is to chose a DAC featuring USB interface synchronised with a high precision internal clock (not in reverse!). It works like that: all new packets arrive from the PC on a specific feedback request from a DAC. It require just inexpensive FIFO buffer. A small FIFO buffer, nothing else. Some implementations still do reclocking, while it is not needed in such situation, as reclocking is not 100% jitter free. I don't know why they do that, speak to them... However galvanic isolation is needed in most of cases, but it is not a common feature, it is why special devices were created. It is called DDC like DI-20 or DI-20HE. It alllows to separate a DAC from a biggest source of noise which is PC. To get the most of a DDC a DAC should support a new interface which becomes popular, called I2S. The alternative is S/PDIF coupled with external clock synchronisation, but it was reserved exclusively for high-end DAC's. I2S is a feature everybody should look for.

The alternative to DDC is a low powered network streamer with I2S output, but it require to use specific software like Roon.

This is why I use an old school low level PCI interface card. No USB It has kernel level drivers which also means it can crash the host system but you get no jitter.

 

[Blasterinn]

Greetings.
I am interested the player discussion. Now i am using musicbee and Peace eq for my listening with headphones.
The thing is, it seems that if i am using wasapi or asio exclusive, then i get full quality 88 or 192k or whatever is playing.
but then the Peace eq is not working.
but if i use directsound or shared wasapi the eq works great but i only get 41k output!
Any ideas???
A little edit here. I am using the ifi usb driver, not that it should matter.

 

[jiiteepee]

Greetings.
I am interested the player discussion. Now i am using musicbee and Peace eq for my listening with headphones.
The thing is, it seems that if i am using wasapi or asio exclusive, then i get full quality 88 or 192k or whatever is playing.
but then the Peace eq is not working.
but if i use directsound or shared wasapi the eq works great but i only get 41k output!
Any ideas???

Hmm... with Peace EQ you must mean EqualizerAPO with Peace frontend (with Peace you make config files for EqualizerAPO).

Have you tried Voicemeeter (I prefer Banana) integration with EqualizerAPO (Voicemeeter supports ASIO, WASAPI and KS output)?

You must mean 44.1kHz output only with WASAPI Shared Mode and Directsound (DS) (actually DS is just a layer on top of WASAPI)? You sure know that you be able to change sample rate through OS Sound options and playback software output options (if not changed automatically by played source (audio)).

 

[Blasterinn]

Yes you are absolutely right about the EqualizerAPO with Peace frontend.
And i do know about the manual bitrate in win OS but then it just stays there and upscales or downscales depending on how high you go right?
Then i think that Voicemeeter or Banana would be my saving grace. Or am i misunderstanding something?

 

[MajorTom]

Yes you are absolutely right about the EqualizerAPO with Peace frontend.
And i do know about the manual bitrate in win OS but then it just stays there and upscales or downscales depending on how high you go right?
Then i think that Voicemeeter or Banana would be my saving grace. Or am i misunderstanding something?

In exclusive mode the OS will defer the bit rate to the program that is using it "exclusively" but, as you noticed, you can't apply EqualizerAPO EQ that way as it only works in shared mode. Shared mode will do the opposite: the bit rate will be controlled by W10 at the default setting specified and will resample any audio outside of that range to match the Windows bit rate.

If you're not keen on Voicemeter you could probably set something up using virtual audio cable. I haven't used either in years but both seem so the above info might not be fully accurate but it points you in the right direction re: virtualised audio.

Re: OP I've always used foobar2k.

 

[sajunky]

In exclusive mode the OS will defer the bit rate to the program that is using it "exclusively" but, as you noticed, you can't apply EqualizerAPO EQ that way as it only works in shared mode. Shared mode will do the opposite: the bit rate will be controlled by W10 at the default setting specified and will resample any audio outside of that range to match the Windows bit rate.

If you're not keen on Voicemeter you could probably set something up using virtual audio cable. I haven't used either in years but both seem so the above info might not be fully accurate but it points you in the right direction re: virtualised audio.

Re: OP I've always used foobar2k.

^This. There are EQ extensions for Foobar. If you don't like any of them, there is an option to take Asio4all output from your favourite EQ app.