Author Topic: audiophile mode  (Read 20260 times)

cooljazz

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Well I can only say, that ASIO4ALL sounds harsh compared to Onkyo SACD Player or Jriver.  Jriver used in Wasapi Event Style Mode makes the best picture concerning sound, better than ASIO4All and currently better than MusicBee from my View of Point, even if I can't really explain why.

Well theoretically ASIO4ALL and Wasapi should be equal sounding, given that both claim to be bitperfect?

audiojfr

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The difference is kind of hard to describe: Instruments just sound more natural, there is better placement on the stage.
MusicBee somehow doesn't sound as fluid as Jriver does... kind of less emotion, more static, less details. - No DSP Effect was used in either program. By default Jriver doesn't even have a working volume bar in Wasapi Event Style Mode.
However MusicBee performs better in any other aspect than this particular issue concerning audiophile grade quality. As posted above another user of the Benchmark Dac1 got a similar result.

So I tested it again today using a Arcam rdac and a pair of B&W P5: Same result, but not that heavy...

Maybe there are some others willing to put it to a test?
Hi
Just to let you i hearing the same as you  i'm using asio ouput not Wasapi  both software set up for
audiophile mode? no dsp no volume etc and i find JRiver as more detail in the high frequencies
and the depth is better more dynamics the stereo image is nice . Musicbee sounds good but the high are
better in Jriver to my taste and musicbee as a bit more of midlows frequencies ?
So even with the 2 software set up for audio bit perfect transfer they
still have a effect on the sound ?  JRiver is 64 bit engine   Musicbee i'm not sure but i think
it's 32 bit engine maybe that's making the difference i'm not sure but i;m sure they don't
sound exactly the same.


Steven

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On sites like hydrogenaudio they require people submit double blind test results before people can make claims about sound quality to have a basis to back up such claims, and eliminate imagined effects.
if you have a 44.1, stereo, 16-bit format lossless file (i wont include MP3 files as there is floating point arithmetic with them) that is sent to an output device with no resampling ie. the output device supports 44100 sample rate, volume at 100%, no EQ or DSP
where exactly do you think JRiver would get its extra detail from and how would a 64 bit engine (I assume for DSP and EQ arithmetic) come into play?? Its a straight transfer of bits from the source file to the output device.
The only plausible thing i can think of is MusicBee doesnt think your device supports 44.1 and is resampling to 48000 (perhaps jRiver is doing the same and maybe jRiver has a better re-sampler)
Last Edit: March 17, 2012, 01:55:03 PM by Steven

audiojfr

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Resampling is off my card support sample rate 32Khz to 96Khz and no if  i watch my soundcard internal software the samplerate
is whatever the song is   so no resampling. Don't get me wrong Musicbee sound reallly good but different then  JRiver.


audiojfr

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"Well theoretically ASIO4ALL and Wasapi should be equal sounding, given that both claim to be bitperfect?"

Asio4all is more like plug-in wrapper  thing for card that don't support Asioi? For ASIO it's better to Have a card tha thave Asio driver
 M-audio  , and most pro sound card .



cooljazz

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"Asio4all is more like plug-in wrapper  thing for card that don't support Asioi? For ASIO it's better to Have a card tha thave Asio driver
 M-audio  , and most pro sound card ."

Custom ASIO Drivers tend to have better performance and thus less glitches when used with really low latency, but ASIO4ALL should be bitperfect. For example Nuforce doesn't have a own ASIO Driver for their DACs. They officially recommend to use ASIO4ALL with their products over using DirectSound or WASAPI.

The 64bit mode in Jriver has no effect by default. It is however very usefull when DSP effects are applied. JRiver claims that things like Crossfeed for headphones are much more exact when applied to 64bit.

However, I want to make clear, that I think that MusicBee is the much better and most easy to use Media Manager.


Jan

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Benchmark didn't get back in touch with me :-(
But I just had some free time to play around and did some audio comparison between Jriver Media Center and MusicBee using a Benchmark Dac1, Sennheiser HD800 and Wasapi output.

Somehow I feel Jriver Media Center does sound more natural and detailed. I know that this can be subjective. But I'm pretty sure its not because I had some friends listening blindly and they could tell the difference without knowing which player was used.

The Dac1 only accepts audio presented as 24bit. So could it be that MusicBee does some weird calculation when served with 16bit audio instead of simply adding zeros?
When using Jriver playing 16bit audio utilizing Wasapi it tells me to set the output bitrate to 24bit, otherwise audio couldn't be played.
MusicBee doesn't prompt anything.

So I'm not sure what could be the reason for this but only that MusicBee is not really bitperfect.

MB is not bitperfect in WASAPI because it uses synchronous (push) mode (as Steven told me in this forum). MB is bitperfect in ASIO. Other players are bitperfect in WASAPI async (event) mode. Tested on a Naim DAC V1 (which comes with bitprtfect testing setting on the external DAC and relevant test files). Steven has indicated willingness to provide async WASAPI in MB lqter in the year.

If you wonder why this difference: WASAPI en ASIO use the clock of the external USB DAC to regulate the data stream. Sync (or push)  mode WASAPI uses the (very much less then stable) PC clock.

And after extensive comparison on High end HiFi this can be heard (NAIM DAV V1 compared to Naim CD5x on NAIM pre, QUAD II refurbished valve monobloc amps and ProAc speaker, so reasonably high end HiFi, not exceptional)

Regards,
Jan

electro

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Just saying: audiophiles are a lost cause. There are plenty of blind tests (ABX) that prove that no one can hear those very small differences. Also differences between a MP3 well coded and a CD. There are a lot of misunderstandings when it comes to digital audio.

Everyone interested in digital audio should watch this video: https://www.youtube.com/watch?v=cIQ9IXSUzuM
Also, here's a great article worth reading: http://xiph.org/~xiphmont/demo/neil-young.html
Last Edit: July 24, 2015, 02:38:16 PM by electro

hiccup

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Just saying: audiophiles are a lost cause.

What is the definition of an audiophile in your opinion?

Quote
There are plenty of blind tests (ABX) that prove that no one can hear those very small differences. Also differences between a MP3 well coded and a CD

That's simply incorrect.
With an audio recording of high quality containing ambient information (acoustic reverb, room ambiance etc), it is easy to distinguish in A-B comparison when you know what to listen to and have good equipment.
(done it many times myself)

Thanks for the links though.
Interesting readings. They don't give you a complete picture of the matter, but still very useful to get more insight in this complicated subject.

electro

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Just saying: audiophiles are a lost cause.

What is the definition of an audiophile in your opinion?
For me: audio quality maniacs. They're more concerted about nitpicking details than to enjoy music. Some of them claim that vinyl is superior to CD, despite the big flaws in every way (portability, size, sound quality - that "warm sound" is actually the original sound distorted, price, duration, side A/B, no tags, etc). Some of them are stupid enough to buy super-expensive cables that work exactly as the cheap ones. Some of them spend a lot of money for extremely little (and impossible to detect) gain in audio quality. Most of them are afraid of ABX tests because they don't want to risk disillusions. Also most of them are victims of charlatans that sell them overpriced stuff.

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That's simply incorrect.
With an audio recording of high quality containing ambient information (acoustic reverb, room ambiance etc), it is easy to distinguish in A-B comparison when you know what to listen to and have good equipment.
(done it many times myself)
Even with high quality audio equipment it's hard distinguish in A-B comparison, because there are too many variables in the "game". The weakest link is the human ears: they're simply not great. Most people don't even have the audio range of 20Hz - 20KHz.

PS: Here's a pretty popular audio test: http://www.npr.org/sections/therecord/2015/06/02/411473508/how-well-can-you-hear-audio-quality
Last Edit: July 24, 2015, 07:27:23 PM by electro

Jan

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Just saying: audiophiles are a lost cause. There are plenty of blind tests (ABX) that prove that no one can hear those very small differences. Also differences between a MP3 well coded and a CD. There are a lot of misunderstandings when it comes to digital audio.

Everyone interested in digital audio should watch this video: https://www.youtube.com/watch?v=cIQ9IXSUzuM
Also, here's a great article worth reading: http://xiph.org/~xiphmont/demo/neil-young.html


Yes, it seems that there are a lot of misunderstandings about audio. Claiming that the difference between CD and MP3 cannot be heard, is simply wrong. MP3 is lossy by (its own) definition. And the information that you point to even explains that. MP3 cuts off at lower frequencies (in the high spectrum), so the available numbers of harmonics to recreate large signal changes (black signal)  become smaller, so the recreated analogue wave is of lower quality.

Furthermore, the article has since its publication been proven wrong. The human ear can actually perceive timing differences down to about 5µsec. That is how your brain adds directionality to sound (you can hear someone coming up from the back and know whether that person is left or right, close or further away). On order to be able encode the samples so that 5 µsec differences can be incorporated, you need 192 KHz. It is correct that you do not need that high frequency to generate the audible spectrum, you need it for timing. So stating that 192 KHz is not needed, is old-school.

And by the way, timing is why Vinyl is better (in this respect) then digital: as its analogue, the timing is always right. If the master is analogue, it was never sampled, so the timing is perfect. If the master is digital, it should be well above the 192 Khz sampling rate, so the resulting analogue wave being copied to the vinyl will also be OK from timing perspective.

Recent research seems to suggest that timing in the analogue signal reaching our ears is more important that" bit perfection" in waveform for listening.

I have worked in IT all my life... we seem to forget that the real (biological) world is analogue...

Last Edit: July 24, 2015, 10:28:29 PM by Jan

hiccup

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Even with high quality audio equipment it's hard distinguish in A-B comparison, because there are too many variables in the "game". The weakest link is the human ears: they're simply not great. Most people don't even have the audio range of 20Hz - 20KHz.
PS: Here's a pretty popular audio test: http://www.npr.org/sections/therecord/2015/06/02/411473508/how-well-can-you-hear-audio-quality

That link only confirmed that I indeed could hear differences, even with the really rather bad selection of material, and discussable manner of relaying it.

In my opinion your statements are predominantly judging, oversimplifying and categorizing people and their different manners and capabilities of appreciating music and audio.

I'd rather enjoy Chet Baker on my mono clock radio, than Katy Perry or JayZ on DSD512.
So sometimes I am an audiophile, and sometimes I am a romantic drunk.
But I try to get the most out of both, and don't assume there is one truth which everybody should agree on.

And something else to think about maybe; you are contradicting yourself a little bit.
You say "some of those audiophiles" enjoy music on vinyl, while it is in your opinion "technical inferior". (which is by the way true in some aspects, and wrong in a few other)
So here you are stating that those 'audiophiles' shouldn't be enjoying music from vinyl so much since vinyl is 'technically inferior'?
So technique all of a sudden does matter then?

The one thing we can establish for sure though, is that this topic is probably not really for you, is it?

hiccup

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And by the way, timing is why Vinyl is better (in this respect) then digital: as its analogue, the timing is always right.

Interesting food for thought.
Timing is everything ;-)

Just curious, since you seem to have some knowledge on this matter:
Record grooves are cut  tangential in production.
Almost all record players have a pivoting arm, so there is only position somewhere in the middle of the record where the timing between left/right channel is correct.
Is this influential to the 'timing' aspect you explained?

electro

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Just saying: audiophiles are a lost cause. There are plenty of blind tests (ABX) that prove that no one can hear those very small differences. Also differences between a MP3 well coded and a CD. There are a lot of misunderstandings when it comes to digital audio.

Everyone interested in digital audio should watch this video: https://www.youtube.com/watch?v=cIQ9IXSUzuM
Also, here's a great article worth reading: http://xiph.org/~xiphmont/demo/neil-young.html


Yes, it seems that there are a lot of misunderstandings about audio. Claiming that the difference between CD and MP3 cannot be heard, is simply wrong. MP3 is lossy by (its own) definition. And the information that you point to even explains that.
Technically, it's a difference. But can you hear it? That's the question. There's a small quality loss, but impossible to detect with your ears. Many tests proved that people can't hear these differences. The MP3 encoders (esp. LAME) are very good these days compared to the early 2000. This is the same as JPEGs: a high quality JPEG is very hard to differentiate from the original source.

Regarding this topic, I suggest you to take a look at these 2 pages:
http://blog.codinghorror.com/the-great-mp3-bitrate-experiment/
http://blog.codinghorror.com/concluding-the-great-mp3-bitrate-experiment/

Quote
MP3 cuts off at lower frequencies (in the high spectrum), so the available numbers of harmonics to recreate large signal changes (black signal)  become smaller, so the recreated analogue wave is of lower quality.
MP3 cut off the higher frequencies, depending on the encoder' settings. For example, MP3 128kbps CBR cut any frequency above 16kHz and MP3 320kbps CBR at 20.5 kHz (above the standard human audio range). MP3 also discards "hidden" frequencies (frequencies that you can't hear anyway, because they're masked by other frequencies). Just use a decent setting when encoding MP3s (e.g. VBR V2 or VBR V0).

Quote
Furthermore, the article has since its publication been proven wrong. The human ear can actually perceive timing differences down to about 5µsec. That is how your brain adds directionality to sound (you can hear someone coming up from the back and know whether that person is left or right, close or further away). On order to be able encode the samples so that 5 µsec differences can be incorporated, you need 192 KHz. It is correct that you do not need that high frequency to generate the audible spectrum, you need it for timing. So stating that 192 KHz is not needed, is old-school.
By who? Audiophiles? I can't trust them at all. 192 kHz is a waste of time and space and can actually make your audio worse (read the article).

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And by the way, timing is why Vinyl is better (in this respect) then digital: as its analogue, the timing is always right. If the master is analogue, it was never sampled, so the timing is perfect. If the master is digital, it should be well above the 192 Khz sampling rate, so the resulting analogue wave being copied to the vinyl will also be OK from timing perspective.
If it's analogue the timing is always right? From where did you get this weird idea? You get way more errors on analogue medium than digital. On digital, every 0 and 1 is 0 and 1, and on analogue there are imperfections (especially if you transfer data). Try to copy something from analogue support to another analogue format (e.g. tape to vinyl, or vinyl to vinyl). You will lose quality every single time. Copying digital data is (or at least should be) always perfect.

And regarding timing, engineers worked many years to improve the hardware related to digital audio. I never noticed anything wrong with the timing of any digital devices that play audio (incl. my computer).

PS: By the way, why people don't demand lossless video as well? Why they don't want 600GB lossless videos instead of a 35GB Blu-Ray?
Last Edit: July 24, 2015, 11:30:40 PM by electro

hiccup

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You have made your point.
You know for certain that 'digital' is perfect by definition, and everybody that has different experiences and opinions is wrong.
Compression, both in audio and video can not be detected by the human ears and eyes. By nobody. Not by you, and by nobody else.
Anybody who says he can is a liar or a fool.

A bit funny though, you say: "And regarding timing, engineers worked many years to improve the hardware related to digital audio".
So that's all wasted effort? Digital was perfect enough already to begin with? You never heard imperfections anyway?

And now you also start about compressing video not having an impact?
That is really a completely ridiculous statement.
 
Let's all be wise and stop this off-topic nonsense.

Feel free to start your own topic somewhere if there is something on this matter that is bugging you so much.
But I can't take you serious here anymore.