24/96 and 24/192 is no good. MP3 sounds better! - Page 2 - diyAudio
Go Back   Home > Forums > Source & Line > Digital Source

Digital Source Digital Players and Recorders: CD , SACD , Tape, Memory Card, etc.

Please consider donating to help us continue to serve you.

Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving
Reply
 
Thread Tools Search this Thread
Old 28th June 2002, 09:27 PM   #11
diyAudio Member
 
Join Date: Apr 2002
peranders, you don't understand. ADC and DAC a very slow. They loss to much bits. When converting sound to digital at 16 bits, it ranges from 5 bits to 13 bits. This is not true 16 bit sound.

Read paragraph "Latency Determines Resolution" in the document http://www.jam-tech.com/Class-J.pdf

My AMD 386 can handle sound but it takes a long time to manipulate CD quality sound. Though the speed is 40 MHz with out a math co-processor.

Quote:
Maybe you aren't in the digital business but ADC and DAC chips are available to buy and they costs virtually nothing, we talk 24/192 and 24/96.
I already know that ADC or DAC are available to buy about several years ago. With school and about a hand full of hobbies that I do (programming, video editing, graphic editing, audio, computer building). It gets very busy.
  Reply With Quote
Old 28th June 2002, 09:41 PM   #12
tiroth is offline tiroth  United States
diyAudio Member
 
Join Date: Dec 2001
Location: Pittsburgh, PA, USA
Electro, I can't decide if you are a troll or really just clueless.
  Reply With Quote
Old 28th June 2002, 09:56 PM   #13
sam9 is offline sam9  United States
diyAudio Member
 
sam9's Avatar
 
Join Date: Jun 2002
Location: Left Coast
Default Limits

The sampling rate of the original Redbook standard was based on the proposition from information theory that in order to exactly reproduce the original signal you needed a sampling rate twice that. Since human hearing maxes out (for female teenagers, everyone else is lower!) at 20kHz, 40kHz should do. This assumes perfect algorithms, of course, going some what higher would seem prudent although after 15 years or so of listenen to CDs, I think the 44.1k is nearly sufficient. Remember, ther IS an upper limit to audible resolution -- space between where we arew and "perfect" is rather small and set the upper limit on realizable improvement.

So I'll by the case for 96kHz sampling on the grounds of conservative over-engineering. 192kHz is just silly unless there is some other purpose such as carrying multichannel signals.

The word length may be a different matter. I don't know what the human limitation is on perception of gradations in amplitude. There are psychoaccouistic factors that I don't understand. Keeping in mind that adding one bit doubles the amplitude resolution, it would seem that since 16 bits sounds quite good, going beyond 24 bits (improving resolution 128:1) seems sufficient. This should allow icreases to the dynamic range of recorded music up to the maximum safe level.

However, there is another possible consideration. Without knowing much about specific algoritms used in music encoding but relying on many years around engineers and programmers, just having the extra bits and sample frequency available to play with ought to make digital signal processing products easier to accomplish. Code can be developed quicker if you don't have such tight constraints. Or to put it another way, you can get away with being quick and sloppy. That sounds bad, but for the consumer it's good because it keeps the cost down because engineers and p0rogrammers cost more than silicon!
  Reply With Quote
Old 28th June 2002, 10:47 PM   #14
tiroth is offline tiroth  United States
diyAudio Member
 
Join Date: Dec 2001
Location: Pittsburgh, PA, USA
I read a white paper by Bell labs that stated that the S/N ratio of human hearing, as limited by thermally induced noise in the auditory system, was approximately 120dB.

Having a full 22-24 bits can help, though, because the digital chain gets longer when we add signal processors etc and these contribute noise--this is related to your "sloppiness" factor .

As far as sampling rate, higher sampling rates mean that shallower filters may be used, and also increase the theoretical S/N per bit. There is also the issue of step response; we're now diverging into an area that is more opinion than theory, but it is plausible that a 192kHz DAC could more accurately reproduce transient attacks.
  Reply With Quote
Old 28th June 2002, 11:51 PM   #15
sam9 is offline sam9  United States
diyAudio Member
 
sam9's Avatar
 
Join Date: Jun 2002
Location: Left Coast
Default Transient attacks

I doubt any transient attacks from any musical instrument (save synthesizers) rise faster than the equivalent of a 20kHz sine wave and if they did I don't think humans could percieve it. Nonetheless, I'll grant there might be value signal processing advantages to having more samples available.

I suspect, higher sampling rate DACs will hit the market because of non-audio applications. They will get used in audio because they are cheap or have some other advantage in circuit design. The higher sample rate will be irrelavant except to the marking departments who will now have 384kHz DACs to help distinguish their product. Something similar goes on with high performance opamps. Thet operate at frequencies irrelavant to audio but have other advantages unrelated to the frequency.
  Reply With Quote
Old 29th June 2002, 01:02 AM   #16
diyAudio Member
 
Join Date: Apr 2002
Quote:
I can't decide if you are a troll or really just clueless.
I'm not clueless. Many sites and books have experimented with ADC and DAC. ADC or DAC lose too much bits of information. The user can manipulate the audio in digital domain but it will always be different of what he or she intended.

Quote:
...put it another way, you can get away with being quick and sloppy. That sounds bad, but for the consumer it's good because it keeps the cost down because engineers and programmers cost more than silicon!
Quick and sloppy drives in more complaints about buggy software and hardware. I rather pay extra for hardware and software that works 100% of the time instead of going back to the manufacture for a refund.


Sound is like drawing a picture on the computer. If the user decreases the depth (bits) that it was first created in. The picture loses quality. The user can always add more information or depth to the original picture although it won't change much.
  Reply With Quote
Old 29th June 2002, 01:49 AM   #17
sam9 is offline sam9  United States
diyAudio Member
 
sam9's Avatar
 
Join Date: Jun 2002
Location: Left Coast
Default sam9

Quick and sloppy doesn't need to mean buggy. Just the opposite sometimes. Anyone who can recall the wierd things done to try to squeeze code and data into a ca. 1970 mainframe will know what I mean. By sloppy I mean you aren't constrained to write the most succinct code imaginable even though it may be nearly impossble to comprehend the logic later or even if machine dependent quirks are utilized. These add to the likelyhood of bugs.

16 bits provides you with 65,536 discrete amplitude levels. I for one have never been able to hear a level change on a CD (including in sweep tones) that hint there any discontinuty is audible. 20 bits gives you 1,048,576 discrete levels and 24 bits gives you 16,777,216. Actually I don't think all the bits get used but just the same the progression is valid. Considering what is achieved with 16 bits, it hardly seems credible that within the relm of simple 2-channel sound reproduction that 24 bits can add anything of benefit to the listener. 20 bits maybe ... but even that a huge jump.

Another benefit I suspect but can prove is that the higher word size and sample size insulates the consumer (me!) from accumulated noise and distotion that adds up from all the steps from session to mix to master to pressing.
  Reply With Quote
Old 29th June 2002, 03:33 AM   #18
diyAudio Member
 
phishead8's Avatar
 
Join Date: Sep 2001
Location: Pasadena, CA
Electro,

***Warning Extremely Off Topic***

About Quantum Computing:

A few months ago, I visited NIST in Boulder Colorado. There I talked with some scientists working on quantum computing, with an actual quantum computer. They gave some sobering information. First of all, quantum computers are not being used by any government. They are still in their infancy. The most complicated calculation done was the factoring of 15. Yes, 3 and 5 are the factors. They are still working on the algorithims of quantum computing, as the data processing is nothing like standard computing. They *will* work best with doing many identical things at the same time. After they solve the scalability problems, factoring a 16 digit number will take as much time as factoring the number 16. This is why gov't's are interested in them. With Quantum Computers, one could decode *any* encryption, while making codes that only other quantum computers could crack.
What is interesting, though, is that quantum computers won't be fast at anything else until the algorithims become very clever.
After seeing the computer I must say this: don't hold your breath for a desktop version unless your desk is a 2000lb. air springed optical bench.

Sorry about the diversion folks. I'll try and keep it to a minimum from now on.

-Dan
  Reply With Quote
Old 1st July 2002, 05:31 PM   #19
dorkus is offline dorkus  United States
diyAudio Member
 
dorkus's Avatar
 
Join Date: Jun 2001
Location: NYC
Send a message via AIM to dorkus
Default i like SACD.

it sounds good. it doesn't sound like PCM. even 24/96 and 24/192 still sound like CD to me, albeit a lot better. but SACD has a naturalness of tone and timbre that no PCM i've heard can match... only good analog is really competitive.

i personally couldn't care less for DVD-A, and would gladly see it disappear from the face of the planet while SACD takes off (which it may not even if DVD-A dies). i'm biased though, given that a lot of the music i listen to is on Sony Classical which obviously gets released on SACD. PCM is easier to work with but SACD sounds better for my music, and i believe the A/D conversion is less complex. the D/A conversion can be a little tricky, with the noise shaping and all, and i don't think we've seen anywhere near the format's potential yet. PCM still has a ways to go too, particularly at higher speeds, but i think it's benefitted from a little more refinement over the years than DSD, which is newer technology... although i suppose single-bit delta/sigma converters have been around for a long time now.
  Reply With Quote
Old 1st July 2002, 05:39 PM   #20
jam is offline jam  United States
diyAudio Member
 
jam's Avatar
 
Join Date: May 2001
Location: Auburn, CA, USA
Marc,

Are you happy with the mods to your Sony DVD NS-500?
You might want to post the list of mods you did and tell us how the sound improved.

Jam
  Reply With Quote

Reply


Hide this!Advertise here!
Thread Tools Search this Thread
Search this Thread:

Advanced Search

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are Off
Pingbacks are Off
Refbacks are Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
What types of sub woofer sounds good ataul Subwoofers 11 10th March 2009 07:56 PM
7868 amp sounds good but low volume PRNDL Tubes / Valves 9 9th December 2007 09:53 PM
JLH vs Mcintosh class B amp which sounds good wleediy Solid State 0 10th July 2007 09:17 PM
Sounds good, i like it, you may appreciate too. destroyer X Class D 95 22nd February 2005 10:09 PM
Who determines what sounds good? Theli Multi-Way 68 9th February 2004 06:01 AM


New To Site? Need Help?

All times are GMT. The time now is 10:12 AM.


vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2014 DragonByte Technologies Ltd.
Copyright 1999-2014 diyAudio

Content Relevant URLs by vBSEO 3.3.2