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Can two software players sound different from each other and why?

I was using WASAPI, probably the one that supports DSD. But foobar2000 only uses 32bit floating point format internally, so not enough for 32 bit integers (once you get rid of the exponent part). So this behavior is expected, it's a limitation, not a bug. The only time I ever saw the RME pass the 32 bit test was with JRiver.

Did you set volume at 100% in Foobar?
 
I believe I can hear a difference between Linux (Puppy Linux Xenial, Deadbeef ) on my ancient laptop and foobar 2000 with ASIO4All on my desktop computer.:eek:
I've put both on my music server at some point and done a matched level, but sighted comparison and I was guessing so if there was a difference it was very small. I was unable to do a fair ABX for a number of reasons which should be apparent to anyone running dual boot systems.
Now this isn't just a music player comparison, it's a hardware change and with Linux, no audio stack wrap.
One day maybe I'll find the interest to try a proper ABX on identical hardware and then different hardware with the same player.
Maybe the hardware makes a difference.:confused:
 
At one point in time, when MP3 ruled and WinAmp 2.x was the name of the game, there was a competition between different MP3 decoder plugins. I thought the discussion will be around those kind of decoders. But these days audiophiles switched to FLAC, and there is nothing to decode. If you have your doubts- record the output to WAV, compare them using Audacity.
 
He meant record digitally, with something to capture the whole OS sound (after mixer).

Many years ago, just when the first Spotify beta came out, I created an account. Soon I used soft to capture and save what I received. Since then nothing.

It would be appreciated if you recommended a free soft for windows. The other day I uninstalled Audacity, maybe it will allow it.
 
The ODAC indeed doesn't have a capture mode, but there must be a way in Windows to multiplex the output (it's possible even without Pulseaudio in GNU/Linux).
 
Did you set volume at 100% in Foobar?
Yes, I didn't try another volume level just to see it fail. Just to be clear, you need "exclusive mode" WASAPI, and I think the output that supports DSD takes care of that.
 
Yep. Instead of chasing good looking women. Shame.. :p

This kind of thing makes me so so doubt maty's claims of foobar not sounding 'good'.

In theory perhaps, in actuality, like a human telling the difference? I'd almost wager my home - not a chance.

I tend not to comment much on the ordeal since I like maty with the way he posts, he's charming in his grammatical style and use of highlighting for nice legibility. Also he posts cool pics of things.

But mostly I barely qualify as someone with the technical know-how to properly pick apart the ordeal. The default I fall to always is "You're making a claim? Sure, demonstrate it with evidence" "Oh you don't want to do that? Well sorry, but the burden of proof is on the person making the positive claim, I owe you nothing in telling you I don't believe you, proving a negative isn't a logical thing, so I won't bother".
 
The ODAC indeed doesn't have a capture mode, but there must be a way in Windows to multiplex the output (it's possible even without Pulseaudio in GNU/Linux).

Someone explain it to me how. I had to enable the internal card from the Bios. In windows I will have to enable more things that I have disabled and forbidden by that of security and privacy.
 
Installed Realtek ALC892. I have no sound -> I need to connect the analog output to the amplifier with a Y cable. I do not know if I still have it, I have to look for it.

https://www.realtek.com/en/products/computer-peripheral-ics/item/alc892
  • DACs with 95dB SNR (A-weighting), ADCs with 90dB SNR (A-weighting)
  • Ten DAC channels support 16/20/24-bit PCM format for 7.1 channel sound playback, plus 2 channels of concurrent independent stereo sound output (multiple streaming) through the front panel output
  • Two stereo ADCs support 16/20/24-bit PCM format, multiple stereo recording
  • All DACs supports 44.1k/48k/96k/192kHz sample rate
  • All ADCs supports 44.1k/48k/96k/192kHz sample rate
  • Primary 16/20/24-bit SPDIF-OUT supports 32k/44.1k/48k/88.2k/96k/192kHz sample rate
  • Secondary 16/20/24-bit SPDIF-OUT supports 32k/44.1k/48k/88.2k/96k/192kHz sample rate
  • 16/20/24-bit SPDIF-IN supports 44.1k/48k/96k/192kHz sample rate
Better idea is try with ODAC and Windows soft to multiplex. How and wchich? EarTrumpet: MS app store does not work for me, I would have to enable it. And Chevolume 0.6 (7 days) does not install. Other free soft?
 
I think you're supposed to use the Loopback option in Audacity. Didn't know Windows was so bad it didn't have a native loopback (but well, people using "ramdisk" applications should have ticked me). I won't be able to help more, as I haven't touched Windows in years.
 
@maty, any chance some nice people in your area or a dealer or studio can lend you a RME ADI-2 Pro for a day or two ...:)
 
Tarragona is a small town. For years there are no HiFi audio stores, only professional. The people I know in the city do not appreciate good music (or literature or Culture). At most, they want good movie sound, spectacular.

In Barcelona it is easier to find people with minority tastes, but years ago I returned to my hometown. If you do not live in a big city the cultural wasteland surrounds you. If I comment that I like jazz, classical or opera they look weird.
 
If one feels at home guess living in big city or smaal town shouldn't matter :) pity odds in outlook for a ADI-3 Pro nearby and pity i haven't one myself you could have borrowed for requested exercise.
 
Tarragona is a small town. For years there are no HiFi audio stores, only professional. The people I know in the city do not appreciate good music (or literature or Culture). At most, they want good movie sound, spectacular.

In Barcelona it is easier to find people with minority tastes, but years ago I returned to my hometown. If you do not live in a big city the cultural wasteland surrounds you. If I comment that I like jazz, classical or opera they look weird.
Lots of culture where I live @maty :) About an hour North of Barcelona.
 
https://www.vb-audio.com/Cable/

* Only works if JRMC v25: Cable Input VB (Direct Sound)

JRMC-v25-VB-Audio-VB-Audio-Direct-Sound.png


* And Audacity: Windows (Wasapi), Cable Input VB, Cable Input VB

Audacity-Wasapi-VB-Audio-VB-Audio-96kHz.png


I have done the test with 01. S0X 24-96 A1 Aqualung.flac and I have captured about 40 s. JRMC v25. Saved as WAV 24/96.

When I played the original (FLAC or converted in WAV too) and the captured, wiht JRMC v25 and KS or Wasapi the clips sound different!

JRMC and Audacity can not be simultaneously, during the process, using Wasapi.
 
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There is also culture in Tarragona like in Girona but the performances do not interest me. And the acoustics is usually very bad even in the cathedral. I can not stand the poor sound quality and less have to pay for it.
 
Using a simple Linux system the player (mpd in my case) sends that bytes to ALSA and they go out via the USB. Nothing else is involved and nothing can sound different.

I use Ubuntu with PulseAudio disabled, player is Audacious via ALSA in USB hardware out mode, bit perfect playback, it's awesome.
 
What would be nice is a simple test for each media player tool, like amirm did for the CCA review; playing a file to the CCA's TOSlink o/p, and sticking the digital output into the analyser, and comparing with the analyser doing a loopback test. That showed that Roon at least was able to stream bit perfect via Chromecast to the CCA.

https://www.audiosciencereview.com/...gital-output.4544/#lg=attachment15619&slide=0

https://www.audiosciencereview.com/...ents-of-chromecast-audio-digital-output.4544/
 
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