Originally Posted by music_man /img/forum/go_quote.gif doesnt every modern dvd player output redbook at 96khz on it's s/pdif output? every player i have hooked to a dac lights up the 96khz locked light when playing redbook cd's.
music_man
Or 48Khz - my Philips is switchable between the two.
Originally Posted by lowmagnet /img/forum/go_quote.gif Sure, if your speakers/phones can do it (most can't, my HD650 can, but who cares, I listen to 44.1/16)
it's about frequencies above 20khz and this paper shows that most instruments do have frequencies above 20khz. That's why sacd, goes to 100khz and turn tables sound better then 16/44.1khz cdplayers. They are simply not limited in frequency range.
I don't know if the it's only the digital amp that samples 11.2MHz or if the DAC also goes that high, or if there's any difference in concept between the dac and the amp in here.
I had a look at the specs and i think the dynamic range of 105 is a bit low for 11.2 mhz sampling! Most cdplayers have a dymanic range of 115 or 120 nowadays.
And my ATH-A900 go up to 40KHZ - so some of the harmonic distorsions make their way up to the ears - you feel it like a pleasant ringing, and instruments (violins, wind instruments) sound so much better.
Quote:
Originally Posted by lowmagnet /img/forum/go_quote.gif Sure, if your speakers/phones can do it (most can't, my HD650 can, but who cares, I listen to 44.1/16)
Originally Posted by hciman77 /img/forum/go_quote.gif Sampling rate directly dictates the maximum frequency available for PCM i.e most standard CD systems - 44.1khz sampling rate dictates that the highest frequency that can be returned is 22.05khz i.e 1/2 the sampling rate - on some(most) systems the cutoff is pragmatically set a little lower i.e 20khz.
Bit-depth i.e 16 or 24 dictates resolution i.e the maximum no of discrete signal levels that can be rendered (for PCM systems) .
So a 16/44.1 system will give you roughly 20 - 20k with each sample having a range of 0000000000000000 to 1111111111111111 i.e 0 - 63535.
It basically says that upsampling to 192 is bad because it gives more work for the DAC to do and hence, more room for error.
That's just a manufacturer explaining why they didn't use 192Khz.
I've tried all upsampling settings on my Stello 220: bypass, 48, 96 and 192. Most of the time 192 sounds best, but there are some numbers in my collection where bypass absolutely is better. Do note most of the time the differences are VERY slight (as in on the edge of audible). Nowadays I simply leave it on the 192 setting.
beyond 16/44, the biggest reason be be using a format like this is for people that take that file, and do stuff with it. Mostly multi-trackers. More bits are better for processing (plug-ins) the material so that high frequency "harshness" can be avoided throughout the "audio chain" that is the recording industry....
I am a bit sceptical about these mad high sampling rates.. but I have to say, the one album I have in 24/96 (DSOTM) does sound very very good. I should get the same album in 44.1 and compare them, though. Also, one thing that people don't usually consider: storage space. That one Pink Floyd album is around 3GB... and that is 96, not 192, 384, etc. So if you want your whole music collection in 96 kHz or higher, it's gonna eat your HDD for breakfast.
DSOTM was an analog recording, and it was remastered numerous times for both analog and digital playback mediums, the most recent one was for the SACD release for digital playback and it was remixed too. My point is making comparisons between earlier generation CD's to this current one would be difficult since the sources are different.
One thing that's not been mentioned. Most stereo equipment that handles 24/192 has better quality electronics than those that only do 16/48. So part of why 24bit is better is because the output stage is better quality and preserves the sound better. That said I've compared a number of CDs at 16/44.1 to some that are 24/96 or 24/192 be they from sources like HDTracks or from SACD. In most cases, My source DAC/AMP is a Fiio X3II and the headphones Beats Studio (soon to become PSB M4U 2 later this week). In most cases, the imaging is better. Not always night and day. The bass has more oompf and in some cases goes lower There is more energy to the midrange is better and is more pronounced. The top end can be smoother sometimes and there is more detail overall. This is the 24bit versions being better. I cannot say if this is because of a better source used for the 24bit versions or a remastering for the 24bit versions.
the other thing that wasn't mentioned (as far as I read it all) is that a higher sampling rate makes it easier for the filtering stage of a sigma-delta DAC. Filtering makes out a great deal of the quality of a DAC. At 16/44 a sigma-delta DAC has to quickly filter out a lot of nasties leading to temporal anomalies like (pre-)ringing, smearing and artifacts far into the audible range. The drastic filtering causes the remaining signal to sound dull, lifeless and artificial. A sigma delta is very noisy and needs steep filtering. Moving up to 96kHz makes it a lot easier, getting away with somewhat less steep slopes. This improves things but only 1 order of magnitude in filtering. Highs still sound artificial compared to a ladder DAC, analog, let alone live acoustic sound. Upping the ante to 192kHz is again 1 step further along from moving all the garbage out of the audible range. For a sigma-delta DAC 90% of the audio quality is in the I/V-stage and filtering. Increasing the samplerate of the signal just makes it easier to obtain a clear signal in the rang up to 20kHz. If you don't filter the range from 20-40kHz so you get a -40kHz audible range just means the old man with hearing -10kHz gets a cleaner signal as far as he can hear. As much as the cheap sigma-delta DAC will give him.
For an R2R- or ladder-DAC there is a lot less nasty HF artifacts. With 44kHz some filtering (1st order) is useful. At 96kHz it is not required... Strictly speaking and according to the original specsheets it is, but doesn't do any good for actual listening to music. It is the cleanest music in the temporal domain you can get. Highs are natural, seemingly soft but very exact. Placement and soundstage are very lifelike, i.e. like live.
Our human brain is very sensitive for correct phase and not just amplitude. Our brain gets very confused by pre-ringing because it's like time in reverse. It reminds me of the film Minority Report. You see the murder scene in the water with ripples moving along. You think you've seen it before so you discard it. But this is the whole plot of the movie. It isn't. It was something that wasn't, so it could happen again without consequence. Imagine throwing a stone in a pond. In real life you expect ripples expanding and dying out from where the stone drops in the water. But what if when you throw the stone the water would slowly start rippling, getting aplified and converging to the point where the stone is going to hit. And then the same thing in reverse, but now like 3 separate stones hit the water. You would not feel very 'Zen' about that would you? Very disturbing because you cannot tell actually when what stone hit the water where, causing all the ripples and interference patterns. This is not relaxing, causing hearing fatigue.
So I use a R2R without any filtering with 24/96 material (if I can get it) for best results. 16 Or 24 bit really doesn't do anything for playback. That is only useful for mixing. Or digital volume control.
I was curious about analog recordings made available in. 24/96 and 24 /192. HD Tracks does it and I'm not sure which I should get. My Oppo HA-2 handles both and I play back through a Pendulumic Stance S1+.
Analog records[ master tapes] don't quite reach even CD quality if one examines its SNR and dynamic capabilities, so dumping them into large digital buckets won't elevate their quality at all.
24/96 is already an overkill for them and 24/192 is just a waste of storage.
Analog records[ master tapes] don't quite reach even CD quality if one examines its SNR and dynamic capabilities, so dumping them into large digital buckets won't elevate their quality at all.
24/96 is already an overkill for them and 24/192 is just a waste of storage.
In that case, why not 24/88? 24-bits does make a difference. That gives more resolution to what's there. The thing is, DACs have gotten better and while a lot of older recordings seem to not need more than 16/44.1, they do in order to sound better. I'd go with 24/88 in those cases.
In that case, why not 24/88? 24-bits does make a difference. That gives more resolution to what's there. The thing is, DACs have gotten better and while a lot of older recordings seem to not need more than 16/44.1, they do in order to sound better. I'd go with 24/88 in those cases.
Actually - you have that wrong. The difference between 24 bit and 16 bit is simply the availably dynamic range (48 db difference to be precise) - although people talk about added resolution, it has no effect on audible playback at all. This was quite an excellent write-up which explains it pretty well:
http://www.head-fi.org/t/415361/24bit-vs-16bit-the-myth-exploded
The reality is that studios use 24bit for recording - especially multi-tracks if they want to lower the noise floor for complex mastering. But for playback, they may as well use 16bit as there is no music that will use full 144 dB of dynamic range anyway. For playback, 16 bit (96 dB) contains all the dynamic range we'll ever need (ie difference between the noise floor and the loudest sound.
Most music - even classical - would have max dynamic range of under 60 dB so in effect you have plenty of head-room. Modern recordings have far less (sadly).
And most DACs nowadays would be lucky to even achieve a full 20 bits of playback - and even if they improve, you won't be able to take advantage of it.
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