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Mythbusters: Audio Production Special
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Greententacle
IsraTrance Full Member
Started Topics :
53
Posts :
323
Posted : Sep 4, 2008 01:26
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Freaks!!!! |
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Upavas
Upavas
Started Topics :
150
Posts :
3315
Posted : Sep 4, 2008 01:32
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just as video at 100 frames per second is technically higher quality than video at 80 frames per second, even though the human eye can only perceive about 60 frames per second.
It is always a bad idea to compare audio and video this way. The analogies don't hold. One reason they don't hold is that at 25 frames per second, there is visible motion aliasing. (Like the wheels of a car that turn backwards and forwards as it accelerates away).
UnderTow
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As far as I know the human eye can perceive visual motion aliasing at 22 fps or lower. If 25fps would have serious aliasing you would be seeing it constantly in cinemas (24fps) or on Pal or Secam TV (25fps) and even more so on HDTV (usually and not always 23.976 fps)!
I am reminded to the fact that Bruce Lee moved so fast that they had to shoot the movie in higher fps rates when he was doing his action. The 24fps setting was too slow to record his movements. That is pretty wicked I think. To my knowledge he was also the only human to ever achieve this level of swiftness on camera!
As to higher quality on higher sampling rates, I think we agreed on the fact that while from a technical standpoint 192 khz IS better quality than say 48khz the difference cannot be heard by the human ear due to the fact that we cannot perceive sound higher than 20khz. The hp filter rule applies ...
Because of that I do agree with you that 192khz is nothing but marketing...
I did not know that rtas plugins run natively from the PC, interesting... I always thought they ran from PT... Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
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hugaw
Started Topics :
7
Posts :
319
Posted : Sep 4, 2008 01:53
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Also, Mac OS X is configured to work better in 24 bits, and windows in 32 bit float.This does not mean that u got a better sound with a PC by the way..
Psy stuff : myspace.com/neyaprod
Non-psy stuff : myspace.com/cheaperbits
french psy production forum : http://www.hadra.net/forum/viewforum.php?f=18 |
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Upavas
Upavas
Started Topics :
150
Posts :
3315
Posted : Sep 4, 2008 01:54
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On 2008-09-04 01:53, hugaw wrote:
Also, Mac OS X is configured to work better in 24 bits, and windows in 32 bit float.This does not mean that u got a better sound with a PC by the way..
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Right, that is where I got the 24 bit myth Spindrift...
Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
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Upavas
Upavas
Started Topics :
150
Posts :
3315
Posted : Sep 4, 2008 02:11
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Mind you on that note. If you have a wheel with spikes and you rotate it 24 times per second while filming cinema standard (24fps) the wheel will appear to be stationary, so from that standpoint you are right Alistair. However 24 times per second is damned fast...
While it is generally true that 24 fps can produce aliasing, for the most part it is not so. Otherwise we would have much higher fps rates when filming (see Bruce Lee)
Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
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UnderTow
Started Topics :
9
Posts :
1448
Posted : Sep 4, 2008 11:28
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On 2008-09-04 01:32, Upavas wrote:
As to higher quality on higher sampling rates, I think we agreed on the fact that while from a technical standpoint 192 khz IS better quality than say 48khz the difference cannot be heard by the human ear due to the fact that we cannot perceive sound higher than 20khz. The hp filter rule applies ...
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No we don't. That is my whole point. 
With good filter design (and everything else in the ADC), 192Khz sounds no better than 44.1Khz. Technically, there is no increase in quality. Only an increase in bandwidth.
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I did not know that rtas plugins run natively from the PC, interesting... I always thought they ran from PT...
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Yep. 32 bit float interface between the TDM hardware and the PC's CPU. HTDM plugins also run natively on the PC's CPU but they run in a sort of emulation of the TDM hardware.
UnderTow |
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Kaz
IsraTrance Full Member
Started Topics :
90
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2268
Posted : Sep 4, 2008 18:42
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On 2008-09-04 00:50, UnderTow wrote:
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On 2008-09-02 00:01, Kaz wrote:
I added the "neighboring frequencies" comment because we can, in fact, hear the difference between 44.1 and 88.2 KHz due to differences in character
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That depends on the converters.
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And we are talking about worst-case scenarios, hence, my comment is valid 
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You need at least double the maximum rate to get a frequency properly (one for the positive peak, one for the negative).
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Not the peaks. Anywhere in the waveform.
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Incorrect, unless all frequencies are in the same phase - using your circle analogy, two points are enough only if you know for certain one is the center, otherwise, two points can give an infinite number of frequency/amplitude combinations.
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By doubling the level of interpolation, you make sure that even the finer characters of the sound are 100% accurate within your hearing range -
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Level of interpolation? You interpolate to get a new value from exiting values. The term is not correct when you are talking about sampling. Also changing the sampling rate, increases the bandwidth, not accuracy.
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My bad, by doubling the sampling rate, hence making the interpolation twice as fine, hence, more accurate reproduction.
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Technically, 80KHz sampling rate is enough,
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60 Khz SR is enough. It needs a filter with a curve equivalent to the natural roll-off of air. Good enough by my book.
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True, but again, we're talking about 100% system compatability, and most filters are far from standard equipment on say.
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In films it was 24KHz, doubled, just because they wanted to be sure that they sound better than regular CDs.
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It had nothing to do with sounding better than CDs. It just seemed like a logical multiple of having 24 frames per second in film.
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That makes no sense, we are talking digital data - engineers don't care much for numbers like 1000, if anything - in such a case they'd work with 49152Hz (2048 samples per frame). It's nice to think that these standards make sense from a scientific point of view - unfortunately, they don't... not really.
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88.2/96 are just that doubled again, to insure that up to 22050/24000Hz, playback will be accurate down to the most minute detail
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Again, it has nothing to do with detail.
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Sure it is, set a 2xdowngrade in sampling rate and you will lose a lot of original character. Not on the amplitude/time scale, but rather on the amplitude/frequency one. Different kind of details, but details all the same.
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An anecdote for SHARC processors and most common DSPs: they work in 64bit (double-precision floating point).
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The fastest SHARC chips do 32 bit float.
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Of course - but DSP architecture allows for cross-influence, hence working in double precision has no performance hit, as opposed to classic 32bit processors like GPUs. Programmability is a wonderful thing, and that's the first feature that the high end plugins for these cards use (at least in any plugin that uses the word "convolution" or "emulation" somewhere).
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That is the #1 excuse of DSP card manufacturers not to endorse CUDA (nVidia's programing standard), since graphics cards use only 32bit math (80-90% performance hit in double precision). The actual reason is of course that TC, UAD and the likes just enjoy it when you have to buy their hardware in order to run their unpiratable plugins (considering that a high end graphics card, even with an 80-90% performance hit, is still stronger than another quad-core CPU in these types of processes).
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As you clearly argue yourself, competition is the only reason. The fastest GPUs are close to a 100 times more powerful than the fastest SHARC chips.
UnderTow
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Of course. BUT, in the next update of OSX, there will be OpenCL support, that means, any programmer will be able to work with GPU processing starting 1H 2009 (*prays for VST dongle*). And hopefully, that will be the last time anyone cares about CPU power in the audio production business. Just spend another $300 on a graphics card instead of $2000 on two high end CPUs. ATI has already dropped their proprietary programming and put their weight behind this standard, and just like OpenGL, this is going to be the heavy duty standard until it is replaced by the likes of DirectX or Core Video / Core Audio in upcoming versions (within a few years).
Apple quote about usage of OpenCL in OSX 10.6 and specifically on the audio production part of things is available on Sound On Sound Magazine (08/08 issue). My assumption is that optimization of internal Logic Studio plugins to use GPU processing power will soon follow (Logic 8.1?), and will later be joined by all software developers that aren't trying to push cheap hardware at pro-equipment prices (*cough*powercore*cough*).
http://www.myspace.com/Hooloovoo222 |
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Kaz
IsraTrance Full Member
Started Topics :
90
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2268
Posted : Sep 4, 2008 18:52
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On 2008-09-02 17:10, forumuser wrote:
Audiophiles could maybe not spot a SINGLE song either in 96khz or 44khz. But everybody can spot a complete production with eq, compression (digital) mixed in 96khz, or in 44khz.
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OTT - Blumenkraft
All digital audio files and outputs were at 16bit 44.1KHz, and if I remember correctly - there's a picture of OTT under the dictionary definition of audiophile.
http://www.myspace.com/Hooloovoo222 |
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Colin OOOD
Moderator
Started Topics :
95
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5380
Posted : Sep 4, 2008 19:06
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Kaz
IsraTrance Full Member
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Posted : Sep 4, 2008 19:14
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On 2008-09-04 01:05, UnderTow wrote:
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On 2008-09-02 22:55, Kaz wrote:
I'm saying bits don't exist in an analog signal - the dynamic range of an analog signal is limited only by amplification,
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Rubbish. You are still limited by the noise floor of the equipment (and the recording). If you boost the signal, you also increase the noise floor.
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What you're saying is not right. It's not even wrong.
I'm just saying there's a different set of problems in digital and in analog processing, and gave this theoretical bit of analog design philosophy - as this is an out of context reply as to why sampling rates and bits simply do not exist in the real world... the sound we hear is a vibration of air, not numbers.
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and analog signals by definition already have an infinite sampling rate (continuous line instead of sampled dots).
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By the time a digital signal is converted back to analogue, it is a continuous signal again.
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No, it's not. While DACs work in MHz, it's still just a lot of dots before it becomes an electrical signal - as long as we're working with samples, it's digital information. Once it is an electrical signal, it is analog (by definition, it is not digital information anymore). And an electrical signal is a complete wave, not just a collection of dots. Just like the electrical signal is not audio - until the membrane in the speaker moves . DACs use extremely high-speed calculations into making sure the electrical signal is a very accurate representation of the digital signal - and it's closer to linear work than in the PC, but it's not linear-based, it's still sample based on the digital form.
http://www.myspace.com/Hooloovoo222 |
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vegetal
Vegetal/Peacespect
Started Topics :
19
Posts :
1055
Posted : Sep 4, 2008 19:25
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On 2008-09-04 19:14, Kaz wrote:
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On 2008-09-04 01:05, UnderTow wrote:
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On 2008-09-02 22:55, Kaz wrote:
I'm saying bits don't exist in an analog signal - the dynamic range of an analog signal is limited only by amplification,
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Rubbish. You are still limited by the noise floor of the equipment (and the recording). If you boost the signal, you also increase the noise floor.
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What you're saying is not right. It's not even wrong.
I'm just saying there's a different set of problems in digital and in analog processing, and gave this theoretical bit of analog design philosophy - as this is an out of context reply as to why sampling rates and bits simply do not exist in the real world... the sound we hear is a vibration of air, not numbers.
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and analog signals by definition already have an infinite sampling rate (continuous line instead of sampled dots).
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By the time a digital signal is converted back to analogue, it is a continuous signal again.
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No, it's not. While DACs work in MHz, it's still just a lot of dots before it becomes an electrical signal - as long as we're working with samples, it's digital information. Once it is an electrical signal, it is analog (by definition, it is not digital information anymore). And an electrical signal is a complete wave, not just a collection of dots. Just like the electrical signal is not audio - until the membrane in the speaker moves . DACs use extremely high-speed calculations into making sure the electrical signal is a very accurate representation of the digital signal - and it's closer to linear work than in the PC, but it's not linear-based, it's still sample based on the digital form.
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Your not making any sense with your last post, if you send your digital signal through a DA-converter, ofc youŽll end up with a continuous signal,,,, what do you else get?
Demand recognition for the Armenian genocide 1915
http://www.devilsmindrecords.org/
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Kane
IsraTrance Junior Member
Started Topics :
23
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1772
Posted : Sep 4, 2008 19:28
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On 2008-09-04 01:32, Upavas wrote:
As to higher quality on higher sampling rates, I think we agreed on the fact that while from a technical standpoint 192 khz IS better quality than say 48khz the difference cannot be heard by the human ear due to the fact that we cannot perceive sound higher than 20khz. The hp filter rule applies ...
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Am I going insane here, or is everyone very misinformed about the relationship between sample rate and frequency (response)?
I'm not challenging anyone, cause I have no idea what I'm talking about compared to most of you in here.
What I've been saying throughout this entire thread is that when you render audio at a sample rate of 192khz, it DOES NOT necessarily mean that the audio contains frequencies (in terms of REESSPONNNSEEE) above 20-24khz, it simply means that the audio has 192,000 samples per second.
If I'm wrong in thinking this, someone please let me know and include the facts to back it up.
You believe in the users?
Yeah, sure. If I don't have a user, then who wrote me? |
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Kaz
IsraTrance Full Member
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Posted : Sep 4, 2008 19:30
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On 2008-09-04 19:25, vegetal wrote:
Your not making any sense with your last post, if you send your digital signal through a DA-converter, ofc youŽll end up with a continuous signal,,,, what do you else get?
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Digital = sample based representation of an audio signal.
Analog = electrical signal with the same waveform as the audio signal.
Explain to me how a digital signal is continuous when it's sample based.
http://www.myspace.com/Hooloovoo222 |
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vegetal
Vegetal/Peacespect
Started Topics :
19
Posts :
1055
Posted : Sep 4, 2008 19:47
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On 2008-09-04 19:30, Kaz wrote:
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On 2008-09-04 19:25, vegetal wrote:
Your not making any sense with your last post, if you send your digital signal through a DA-converter, ofc youŽll end up with a continuous signal,,,, what do you else get?
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Digital = sample based representation of an audio signal.
Analog = electrical signal with the same waveform as the audio signal.
Explain to me how a digital signal is continuous when it's sample based.
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*sigh* I think your missing the point and/or misunderstand, i never said that a digital signal is continuous.
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Meaning letting the digital signal going through a DA-converter you end up with a continous signal, just as you stated in the last post.
Demand recognition for the Armenian genocide 1915
http://www.devilsmindrecords.org/
http://www.myspace.com/vegetalmusic
http://www.checkpoint-music.com/ |
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Kaz
IsraTrance Full Member
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Posted : Sep 4, 2008 20:10
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On 2008-09-04 19:47, vegetal wrote:
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On 2008-09-04 19:30, Kaz wrote:
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On 2008-09-04 19:25, vegetal wrote:
Your not making any sense with your last post, if you send your digital signal through a DA-converter, ofc youŽll end up with a continuous signal,,,, what do you else get?
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Digital = sample based representation of an audio signal.
Analog = electrical signal with the same waveform as the audio signal.
Explain to me how a digital signal is continuous when it's sample based.
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*sigh* I think your missing the point and/or misunderstand, i never said that a digital signal is continuous.
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'k, fair enough. I was working under the assumption we were talking about internal DAC processing (and the problems of ADDA architectures) at this point, and we all know what they say about assumptions.
http://www.myspace.com/Hooloovoo222 |
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