iBasso DX100 Reference DAP - ES9018 inside

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loremipsum said:

Hell, why even bother recabling your headphones with silver or crystal if the voice coil of the driver is still copper?
 
I could make a killing off of replacing copper voice coils with silver wire... 

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That's a great question!

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Originally Posted by loremipsum /img/forum/go_quote.gif
 

Hell, why even bother recabling your headphones with silver or crystal if the voice coil of the driver is still copper?
 
I could make a killing off of replacing copper voice coils with silver wire... 

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The Hifiman RE252 has silver voice coil wires, the Beyer DT48 has an aluminium driver, there are Sony's with diamond and sapphire diaphragms, the technology to make a pure-metal IEM or headphone is certainly there.
 
If you only change your cables to silver, nothing will happen.  Ideally, you should dip your ears in liquid silver too, and coat your canals with diamond dust from Jupiter.
 
Make sure your music is 32bit/384kHz native in 2-channel, Kernel Streaming via crystal coax directly to the DAC installed in your neck, connected directly to your custom IEM with 7N OCC platinum cables under your skin.
 
The DAC chip itself must be soldered with 7N OCC platinum, as well as the connections on the silicon board (more crystal).
 
If you use a copper neutrik jack, you just threw a towel over the screen of your new 64" plasma and you won't hear any of this.
 
 

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knowhatimean said:

You would hope they record at 24/192; Labels that record popular genres of music generally aren't that particular except for rare "indie" labels. So someone has actually said 24/192 is overkill & just a huge waste of space, huh ? With that kind of attitude it really isn't much of a surprise that so many recordings have given "mediocrity" a whole new definition (as in garbage). It was'nt a so called "professional" that made this statement was it ? (Though it wouldn't surprise me as there doesn't appear to be any shortage of "pros" in any field who are so clueless they give the real pros a bad name! (this is only my non-professional opinion of course!!!) (& I generally don't care for lawyers either! Hehehe)

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It is in fact a giant waste of space.
 
http://www.benchmarkmedia.com/discuss/sites/default/files/Upsampling-to-110kHz.pdf
 
Also, that white paper doesn't cover it, but even the best 24-bit A/D and D/A converters cannot actually resolve the entire dynamic range afforded by 24 bits of quantization.  At best, you can get 20 bits or so.  That is, unless you cryogenically cool all your electronics.
 
So the absolute best that recording technology can possibly record is roughly 20 bit, 110 kHz, give or take a few kHz depending on the chip/implementation.  Anything higher than that is a total waste of space in terms of storage.  There are definite exceptions when higher (24 or possibly 32) bit depth can be useful in processing audio files, such as when mastering, mixing, or even using software volume control.  But as far as the capture and storage of audio goes, that's the best that electronics can possibly do.
 
And that's before we even look at audibility.  To sum it up shortly, the only difference between 24 bit and 16 bit audio is quantization distortion and quantization noise. Quantization distortion is nasty and can easily be audible because it distorts waveforms in distinct patterns.  However, quantization distortion is easily eliminated through the use of dither.  Proper noise-shaping dither masks audible quantization distortion patterns with very, very low level noise (that is entirely inaudible under normal listening volumes) added to the signal.
 
It's possible to hear this quantization noise if you listen at what would be extremely high, ear-damaging volumes with normal music - by turning up very quiet passages to very loud volumes.  But at normal listening volumes, level matching, and with proper dither (not always a given), the difference between 24 bit and 16 bit audio is for all intents and purposes inaudible.  Also, occasionally, intermodulation distortion in equipment can create audible artifacts at higher sample rates (i.e. 96 kHz vs. 44.1 kHz).  But that has nothing to do with higher sample rates themselves being audible.
 
Similarly, different masters can very easily sound different.  This is why high resolution audio sounds better most of the time - not because it is high resolution, but because it was better mastered to begin with.  This alone is a good enough reason to buy high resolution audio, but if the same master is available at 16/44.1 or even 24/96 versus 24/192, it's safe to go with the 16/44.1 files.
 
This is a good overview of the topic:
http://www.head-fi.org/t/415361/24bit-vs-16bit-the-myth-exploded
 
And hydrogenaudio extensively covers ABX'ing high resolution versus CD resolution audio:
http://www.hydrogenaudio.org/forums/index.php?showtopic=49843

On average, humans can only hear from 20Hz to 20kHz. 
 

 
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mischa23v said:

[size=medium]44.1KHZ vs  96KHZ vs 192KHz:[/size]
 
[size=medium]Human hearing is at 70khz something, which is optimal human hearing level. The closer it is to 70 khz the better it should sound to you.[/size]
 
 
 

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blackbeardben said:

It is in fact a giant waste of space.
 
http://www.benchmarkmedia.com/discuss/sites/default/files/Upsampling-to-110kHz.pdf

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Yeah, that's the guy I was thinking of, John Siau, he developed the DAC1 right?
 
Scroll down to "Recent Reviews" click on "Epiphany Acoustics", read comment "The DAC1 sounds serrated and awfully bright".
 
Anyway the article you linked seems to only cover upsampling, and not recording, or?  I believe the other article is the same, and neither of them have any pretty pictures, links or sources.
 
You can't say it's a waste of space, and then link to an article only covering upsampling, which has no effect on space, or?
 
It says something about the AD1853...
 
If they are only talking about measuring opamp chips, then I'm skeptical because the North-Western video guy - with the same line of dScope thought - says he thinks no one can hear the difference between NE5532 and OPA2134 in a blind test, which I find a bit weird, idk...
 

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kiteki said:

 
Yeah, that's the guy I was thinking of, John Siau, he developed the DAC1 right?
 
Scroll down to "Recent Reviews" click on "Epiphany Acoustics", read comment "The DAC1 sounds serrated and awfully bright".

 
 
Anyway the article you linked seems to only cover upsampling, and not recording, or?  I believe the other article is the same, and neither of them have any pretty pictures, links or sources... =/

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Yes, this is his (and by the copyright/hosting of the file, Benchmark Audio's) position on sample rates.  If you actually read the white paper, the bit on 96 kHz vs. 192 kHz sampling rate is in general terms, for D/A & A/D conversions period - not anything to do with upsampling.
 
I agree that no proof is given, and it would be nice to have (I can't remember if there was another paper that went into more detail), but I'm not about to argue with his conclusion when the difference between 44.1 kHz and 96 kHz is inaudible for all intents and purposes to begin with.

 
Okay, well... if no proof is given, I don't see much reason to believe in it, when there are other papers which state that 24/192kHz is ideal, and actually show at least one pretty picture to defend their claims - http://www.cirrus.com/en/pubs/whitePaper/DS668WP1.pdf
 
 

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kiteki said:

 
Okay, well... if no proof is given, I don't see much reason to believe in it, when there are other papers which state that 24/192kHz is ideal, and actually show at least one pretty picture to defend their claims - http://www.cirrus.com/en/pubs/whitePaper/DS668WP1.pdf
 
 

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I see no pretty picture depicting 192 kHz, nor do I see any more evidence concerning 192 kHz than in the Benchmark white paper.
 
You realize what the impulse response graph is depicting, right?  Bandwidth.  Take out the frequencies above what 48 kHz can record (i.e. a little under 24 kHz) from the 96 kHz graph, and you're left with the same graph.  Since frequencies that high are inaudible anyway (again, except for possible and unwanted intermodulation distortion in the audible band caused by said higher frequencies), the whole premise is a sham.
 
 
Considering that neither one (96 kHz or 192 kHz sample rates) is audible, why do we care in the first place?  I know I don't and I sure as heck am not wasting my valuable hard drive space (twin 750 GB hard drives in my laptop are full enough with 200000+ photos) on sample rates that are inaudible, much less sample rates that aren't necessarily even measurably better on the best electronics available.

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blackbeardben said:

...I sure as heck am not wasting my valuable hard drive space (twin 750 GB hard drives in my laptop are full enough with 200000+ photos)...

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Sorry for going even further off topic: Why the HELL do you carry 200k+ photos around?
 
 

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blackbeardben said:

I see no pretty picture depicting 192 kHz, nor do I see any more evidence concerning 192 kHz than in the Benchmark white paper.
 
You realize what the impulse response graph is depicting, right?  Bandwidth.  Take out the frequencies above what 48 kHz can record (i.e. a little under 24 kHz) from the 96 kHz graph, and you're left with the same graph.  Since frequencies that high are inaudible anyway (again, except for possible and unwanted intermodulation distortion in the audible band caused by said higher frequencies), the whole premise is a sham.
 
 
Considering that neither one (96 kHz or 192 kHz sample rates) is audible, why do we care in the first place?  I know I don't and I sure as heck am not wasting my valuable hard drive space (twin 750 GB hard drives in my laptop are full enough with 200000+ photos) on sample rates that are inaudible, much less sample rates that aren't necessarily even measurably better on the best electronics available.

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It looked like the impulse response graph was indicating there is less pre-echo in the 96kHz recording, now that I'm looking more closely... the Y axis on the graph is different, I'm confused.
 
It says "it is the removal of digital decimation and interpolation filters, not the extended frequency range, that produce the audible improvements offered by SACD over conventional PCM"
 
The thing is, people can install a simple upsampler in foobar, or sometimes it's in their driver, they upsample to 96 or 192kHz, and they hear a (slight) difference, so then they believe that there is an audible difference in native 96kHz and 192kHz material too, even if that's not the same as upsampling, what do you think?
 

is there any update about releasing date or price?

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kiteki said:

It looked like the impulse response graph was indicating there is less pre-echo in the 96kHz recording, now that I'm looking more closely... the Y axis on the graph is different, I'm confused.
 
It says "it is the removal of digital decimation and interpolation filters, not the extended frequency range, that produce the audible improvements offered by SACD over conventional PCM"
 
The thing is, people can install a simple upsampler in foobar, or sometimes it's in their driver, they upsample to 96 or 192kHz, and they hear a (slight) difference, so then they believe that there is an audible difference in native 96kHz and 192kHz material too, even if that's not the same as upsampling, what do you think?
 

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Yes, I understand that filtering is a legitimate reason to increase sample rate.  However, 44.1 kHz was chosen for a reason - it is considerably higher in frequency than what typical adults can perceive, so the reconstruction filtering has more or less zero impact on audibility.  Factor in psychoacoustics with actual music, and the audibility of sounds even in audible frequencies in the 16 kHz - 20 kHz range is almost nil (see lossy compression).
 
Regarding their impulse response graphs, I'm still unsure how this is supposed to apply at all in real-world conditions.  Again, our ear/brain system acts as a low-pass filter, so that we don't hear anything above our own ears' frequency response curve (i.e. X dB at 16 kHz, X-Y dB at 18 kHz, X-Y-Z dB at 20 kHz, etc.).  Additionally, such impulses as they are demonstrating just don't occur in real sound - the closest thing I can think of is if you cut off a waveform at some voltage at the start or end of a track.  That would occur if you don't have gapless playback and there's no sort of fade-in/out imposed on the signal.  Of course, this is an undesirable situation to be in in the first place and we could care less about reproducing it or not.
 
As for resampling and hearing differences; well, how the software and hardware handles different sample rates can affect what happens.  I can't detail individual cases, but more often than not hardware and the drivers that service it can treat different sample rates differently, resulting in audible artifacts.  Obviously not every DAC is going to have this problem, but it does exist and it isn't uncommon.  Another problem is that upsampling requires interpolated values between the actual values - and because the interpolated values aren't going to fall perfectly into the quantization levels afforded by the bit rate, you're going to get quantization error.  That means quantization distortion and noise, and thus dither to eliminate the distortion.  If it's not done right, it could be audible.  At the very least, you're raising the noise floor over the original file.
 
Now, if you were to talk about oversampling in the DAC, that'd be an entirely different thing...

 
Okay, well you said "I understand that filtering is a legitimate reason to increase sample rate.", and the discussion is valid since the PS3 community, and the upcoming Fostex HP-A8, cheaper SACD players, DSD over USB, and such... is increasing the popularity and interest a little, in 2012.
 
Not to mention there are cheap 32bit / 384kHz capable DAC/Amp's now (the Musiland MM03), it's going to make consumers think "Oh, my 16/44 is lousy!"
 
I'm not going to claim I've heard any differences with any of this high-rez stuff... it could just be some phantom distortion in my hardware for all I know, the same applies to the SoX plugin in Foobar, but it does sound different and I don't know what all those algorithms are.
 
 
128x oversampling (whatever that is) and NOS like PCM1704 is a different discussion yeah.
 
Anyway, this thread is interested in the former...
 

[size=medium]Going to clarify what I stated earlier:[/size]
 
[size=medium]Bitt Depth refers to the number of bits you have to capture audio. The easiest way to envision this is as a series of levels, that audio energy can be sliced at any given moment in time. With 16 bit audio, there are 65,536 possible levels. With every bit of greater resolution, the number of levels double. By the time we get to 24 bit, we actually have 16,777,216 levels. Remember we are talking about a slice of audio frozen in a single moment of time. [/size]
[size=medium]Now lets add our friend Time into the picture. That's where we get into the Sample Rate.[/size]
[size=medium]The sample rate is the number of times your audio is measured (sampled) per second. So at the red book standard for CDs, the sample rate is 44.1 kHz or 44,100 slices every second. So what is the 96khz sample rate? You guessed it. It's 96,000 slices of audio sampled each second. [/size]
 
[size=medium]Space required for of stereo digital audio [/size]
[size=medium]Bit Depth             Sample Rate       Bit Rate                File Size of one stereo minute    File size of a three minute song [/size]
[size=medium]16           44,100   1.35 Mbit/sec    10.1 megabytes                30.3 megabytes[/size]
[size=medium]16           48,000   1.46 Mbit/sec    11.0 megabytes                33 megabytes [/size]
[size=medium]24           96,000   4.39 Mbit/sec    33.0 megabytes                99 megabytes [/size]
[size=medium]mp3 file                128 k/bit rate     0.13 Mbit/Sec    0.94 megabytes                2.82 megabytes [/size]
[size=medium]So you see how recording at 24/96 more than triples your file size. Lets take a 3 minute multi-track song and add up the numbers. Just to put the above into greater relief, I included the standard MP3 file's spec. [/size]
 
[size=medium]Hard disk requirements for a multi-track 3 minute song [/size]
[size=medium]Bit depth/sample rate   number of mono tracks                size per mono track        size per song      songs per 20 gigabyte hard disk        songs per 200 gigabyte hard disk[/size]
[size=medium]16/44.1 8              15.1 megs            121 megs             164         1640[/size]
[size=medium]16/48     8              16.5megs             132 megs             150         1500[/size]
[size=medium]24/96     8              49.5 megs            396 megs             50           500[/size]
[size=medium]16/44.1 16           15.1 megs            242megs              82           820[/size]
[size=medium]16/48     16           16.5 megs            264 megs             74           740[/size]
[size=medium]24/96     16           49.5 megs            792 megs             24           240[/size]
 
[size=medium]you should be noting two things now:[/size]
[size=medium]1. Recording at 24/96 yields greatly increased audio resolution-over 250 times that at 16/44.1 [/size]
[size=medium]2. Recording at 24/96 takes up roughly 3 1/4 times the space than recording at 16/44.1[/size]
 
[size=medium]Now lets get to the subjective side of how music sounds at these different bit depths and sample rates. No one can really quantify how much better a song is going to sound recorded at 24/96. Just because a 24/96 file has 250 times the audio resolution does not mean it will sound 250 times better; it won't even sound twice the quality. In truth, your non-musically inclined friends may not even notice the difference. You probably will, but don't expect anything dramatic. Can you hear the difference between an MP3 and a wave file? If so, you will probably hear the difference between different sample rates. For example, the difference between 22.05 kHz and 44.1 kHz is very clear to most music lovers. A trained ear can tell the difference between 32khz and 44.1. But when 44.1 and 96kHz are compared it gets real subjective. But lets try to be a little objective here.[/size]
 
[size=medium]Lets talk about sample rate and the Nyquist Theory. This theory is that the actual upper threshold of a piece of digital audio will top out at half the sample rate. So if you are recording at 44.1, the highest frequencies generated will be around 22kHz. That is 2khz higher than the typical human with excellent hearing can hear. Now we get into the real voodoo. Audiophiles have claimed since the beginning of digital audio that vinyl records on an analog system sound better than digital audio. Indeed, you can find evidence that analog recording and playback equipment can be measured up to 50khz, over twice our threshold of hearing. Here's the great mystery. The theory is that audio energy, even though we don't hear it, exists as has an effect on the lower frequencies we do hear. Back to the Nyquist theory, a 96khz sample rate will translate into potential audio output at 48khz, not too far from the finest analog sound reproduction. This leads one to surmise that the same principle is at work. The audio is improved in a threshold we cannot perceive and it makes what we can hear "better". Like I said, it's voodoo.[/size]

Nice info there Mischa. Just as I stated before 24/96 is pretty much enough for our needs. I didn't said that 24/192 is bad, but it is just overkill for portable use, and needs a lot of space. Another problem with digital music is jitter. And AFAIR high sampling rate can benefit under condition of low jitter (FYI Jitter is the undesired deviation from true periodicity of an assumed periodic signal in electronics and telecommunications, often in relation to a reference clock source.). I hope iBasso will be able to design a good jitter correction system with master clock that has very low phase jitter.