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Mythbusters: Audio Production Special
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Kane
IsraTrance Junior Member
Started Topics :
23
Posts :
1772
Posted : Sep 2, 2008 14:53
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On 2008-09-02 05:50, OpenSourceCode wrote:
you're confusing khz
almost all humans can't hear above 22khz... that's the frequency (Pitch) of the actual sound. The number of times that the oscillator oscillates per second.
88.2 Khz is the number of times per second that the computer samples 16/24/32 bits (1's or 0's) of information when you bounce down a piece of audio.
totally different parts of the process.
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Exactly. I was afraid to point this out when no one had after like 10 posts lol. Human hearing at 22kHz is incredibly far fetched. A lot of people can't even hear above 17-18kHz.
Sample rate has nothing to do with frequency, it's just a measurement samples per second. Whether the difference in 88.2khz and 192khz is audible to you or not, 192khz is still higher quality..this is why Blu-Ray DVD audio is at 192khz.
  You believe in the users?
Yeah, sure. If I don't have a user, then who wrote me? |
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vegetal
Vegetal/Peacespect
Started Topics :
19
Posts :
1055
Posted : Sep 2, 2008 15:19
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On 2008-09-02 14:53, Kane wrote:
Sample rate has nothing to do with frequency, it's just a measurement samples per second. Whether the difference in 88.2khz and 192khz is audible to you or not, 192khz is still higher quality..this is why Blu-Ray DVD audio is at 192khz.
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Nothing to do with frequency lol ? the amount of samples taken per second, you said it yourself.
"Frequency is a measure of the number of occurrences of a repeating event per unit time"... here you go , wiki is your friend http://en.wikipedia.org/wiki/Frequency
If group of audiophiles can´t tell a 48 Khz from a 96KHz in a conducted blindtest, then why would people start noticing a quality improvement at 192KHz?
Samplerate is bandwidth
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-aeon-
Aeon
Started Topics :
10
Posts :
546
Posted : Sep 2, 2008 15:24
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does the confusion lie with our use of 'frequency'?
i.e. frequency as in pitch/note vs. sampling frequency...
frequency (pitch) is the number of cycles or periods per second
(sampling) frequency is the number of samples per second |
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Kane
IsraTrance Junior Member
Started Topics :
23
Posts :
1772
Posted : Sep 2, 2008 17:09
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On 2008-09-02 15:19, vegetal wrote:
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On 2008-09-02 14:53, Kane wrote:
Sample rate has nothing to do with frequency, it's just a measurement samples per second. Whether the difference in 88.2khz and 192khz is audible to you or not, 192khz is still higher quality..this is why Blu-Ray DVD audio is at 192khz.
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Nothing to do with frequency lol ? the amount of samples taken per second, you said it yourself.
"Frequency is a measure of the number of occurrences of a repeating event per unit time"... here you go , wiki is your friend http://en.wikipedia.org/wiki/Frequency
If group of audiophiles can´t tell a 48 Khz from a 96KHz in a conducted blindtest, then why would people start noticing a quality improvement at 192KHz?
Samplerate is bandwidth
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I was talking about frequency as a pitch..sorry about that. What I was pointing out is that if an audio file is rendered at 192khz (sample rate), that doesn't necessarily mean that it includes frequencies (pitch) up to 192khz. It just means that the audio has 192,000 samples per second.
And I didn't say that anyone would start noticing an audible difference in 192khz vs. 82.2khz or even 41.1...I was just saying that audio at 192khz is technically higher quality than audio at 82.2khz, 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.
You believe in the users?
Yeah, sure. If I don't have a user, then who wrote me? |
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forumuser
Started Topics :
0
Posts :
24
Posted : Sep 2, 2008 17:10
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On 2008-09-02 15:19, vegetal wrote:
If group of audiophiles can´t tell a 48 Khz from a 96KHz in a conducted blindtest, then why would people start noticing a quality improvement at 192KHz?
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Quote:
| subconsciousmind wrote:
But it is making sense to work in higher sampling frequencies in terms of the possibility of more accurate DSP Algorythms. Especially EQ-algorythms can be programmed more accurate for higher samplingfreqs, closer to analog.
As for the AD converters, some people praise higher samping freqs not due to the frequency range, but due to the fact that natural drawbacks from AD can be compensated by recording in higher sampling freqs.
Personally I have the impression that differences araise when the effects multiply. one track, one eq in 96khz.. who cares. ALL tracks, all FX in 96khz... little things still summ up..
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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.
If its rendered into 96 or 44 in the end doesn't really matter anymore. 96khz or 192 really comes into play during processing and when many things come together.
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Elad
Tsabeat/Sattel Battle
Started Topics :
158
Posts :
5306
Posted : Sep 2, 2008 17:21
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it has very strong relation i would think..
if i want to sample 22khz tone with stereo effect/modulation/something then i will use the total of the 44.1khz that most regular system work with.. its the same second and same frequency no? (just double cause of stereo?)
im no proffesor but thats what i understood so far , and also for sure change the DAW sampling rate makes the sound to change to lower frequency as well.. i thought its obvious connection but maybe not... will leave it to you pro's and come back in 2-3 more pages to find out lol
anyhow , there is no harm (exept HD space) in working in better resolution.. sure no need to go 196khz when nothing supports it , no synth no player nothing.. and if there is then sure not in evry house ya know..
check out the thread i have in the 'trance' section.. most people listening to mp3 on their computer probably with computer speakers , and also using mp3 players with the headphones you probably get with it when you buy it...
i dought the croud notice anything beyond 192kbps mp3 file anyhow even on fancy speakers so in the shitty ones.... oh , and most of them will check your tunes in mono 64kbps mp3 on myspace haha |
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Kaz
IsraTrance Full Member
Started Topics :
90
Posts :
2268
Posted : Sep 2, 2008 18:41
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On 2008-09-02 17:10, forumuser wrote:
If its rendered into 96 or 44 in the end doesn't really matter anymore. 96khz or 192 really comes into play during processing and when many things come together.
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Correct on the first sentance and so very wrong on the second bit. 192KHz, no matter how much you process, will not, in any way, whatsoever, give any advantages you can work with. 192KHz is a marketing ploy - 80KHz sampling rate will be enough to not have any artifacts due to digital process up to 20KHz. And even then - an 80KHz sampling rate won't be making any difference in listening unless you apply digital processing to the waveform (as in digital EQs, karaoke filters, things like that) - if the output has been properly dithered. Note that if you apply analog effects, 20bit 44.1KHz will give you all the detail you can work with. You just won't need more. You're just using more HDD space, more RAM, and more processing power (and making stability and performance issues much more problematic) by raising the sampling rate - without having ANY advantages that humans can hear.
A single repetition of a sinewave has two parts, one positive, and one negative, that's why a 20KHz signal requires at least two sample - one for the positive and one for the negative, otherwise, the system will not be able to differentiate it with a 10KHz sine wave. Hence, 44.1KHz sampling rates as a standard.
http://www.myspace.com/Hooloovoo222 |
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MadScientist
IsraTrance Full Member
Started Topics :
97
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1220
Posted : Sep 2, 2008 19:50
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kaz, do you want to say that you need higher bitdepth/sampling rate when applying digital effects than for recording analog signals?! 
https://soundcloud.com/hazak
"Have you ever had that feeling where you're not sure if you're awake or still dreaming?"
"Hmm, yeah... All the time, man - it's called mescaline. The only way to fly!" |
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Kaz
IsraTrance Full Member
Started Topics :
90
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2268
Posted : Sep 2, 2008 22:55
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On 2008-09-02 19:50, MadScientist wrote:
kaz, do you want to say that you need higher bitdepth/sampling rate when applying digital effects than for recording analog signals?!
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I'm saying bits don't exist in an analog signal - the dynamic range of an analog signal is limited only by amplification, and analog signals by definition already have an infinite sampling rate (continuous line instead of sampled dots). You lower the volume by X percent, you will lose no detail, the wave just has X percent less amplitude. A lot of the digital artifacts haven't existed on analog equipment... well, ever, as the name rightfully implies. There go the two main reasons for working above 20bit/44.1KHz.
You need a higher bitdepth/sampling rate when applying digital effects than the minimal necessary output signal, in order to make sure the output is as accurate as possible.
http://www.myspace.com/Hooloovoo222 |
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MadScientist
IsraTrance Full Member
Started Topics :
97
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1220
Posted : Sep 2, 2008 23:42
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you re right about the digital part, but I dont get where you get that 20bit/44.1kHz from.
if you mention yourself that an analog signal is infinite in view on bits/sr, and considering the a/d conversion and the aliasing filters applied during conversion, it seems quite mysterious that you say 20bit/44.1kHz is enough to cover it. but maybe I just get you wrong on that... 
for analog recording, I'd better suggest a sample rate of 88.2kHz and at least 24bit or 32fp.
https://soundcloud.com/hazak
"Have you ever had that feeling where you're not sure if you're awake or still dreaming?"
"Hmm, yeah... All the time, man - it's called mescaline. The only way to fly!" |
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UnderTow
Started Topics :
9
Posts :
1448
Posted : Sep 3, 2008 00:24
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On 2008-09-01 17:00:16, Kaz wrote:
Working in 32 bits has an audible difference to working in 24 bits.
People tend to talk a lot about such things as "dynamic range" when explaining why 32 bits is necessary.
The dynamic range for 24 bit work is 144.48db. The absolute, highest end DAC converters, in ideal studio design, have a 120db signal to noise ratio. That means that not only can DACs not convert 32bit audio, beyond 20bit, all you hear is NOISE. The ultimate hardware solution is not capable of the quality people say "you need".
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There is no point in recording above 24 bit but you mention "Working in 32 bit" which usually refers to the word length of the DAW engine and which is usually in floating point format.
It makes sense to use 32 bit float (or more) for the DAW engine as this gives 25 bit of resolution (sign bit defaults to an extra bit) and an exponent. The exponent gives the massive dynamic range of floating point systems. With 32 bit float you get 1536 dB of dynamic range.
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1) 32bit math is easier to work with than 24bit on a computer, therefor, 32bit may actually slightly improve stability issues in badly-coded plugins.
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It has absolutely nothing to do with stability. All (modern) CPUs have floating point engines that take off the load from the main logic fabric of the CPU. Using floating point allows much more calculations per second.
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2) Destructive effects (effects which change the character of the sound in irreversible ways) lose a bit of detail in the sound. The most conservative approach is to say that you lose 1 bit of detail for every time you process that - so while 24bit audio will only allow 4 (or more, but we're talking worst case here) different destructive processes on the sound before some detail degredation may become audible on extreme high end systems, while 32bit will offer 12.
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This makes no sense. It is all about rounding off of calculations. If a plugin or a summing engine is well designed, it will use the 80 bit accumulators available in modern CPUs. This then gets rounded off or truncated for the next step. If well designed, you should never loose an entire bit.
I prefer to run my DAW at 64 bit float (double precision) so as to minimise any rounding errors.
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Of course, 4 destructive effects should be plenty for mixdown work, and the added cost of using 33% more RAM and HDD space will put more pressure on your system - nullifying and more the gains of working in a PC-friendly audio bit-depth. Bottom line: The thermal noise of electrons bumping against each other in a resistor is louder than -144db. That means that as long as you have a resistor in your system (like in every amplifier, for instance), going beyond 24bit is not only useless, but hardware will LITERALLY never achieve this.
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Again, don't confuse recording bit depth and processing bit depth. And it is a bad idea to record signals that peak at or close to 0 dB FS as usually the last few dBs of the converter are not entirely linear so good gain staging means your digital noise floor will be above -144 dB FS.
The great advantage of 24 bit over 16 bit is that you can record at much lower levels without ever risking getting close to the audible digital noise floor. (Which becomes important when you sum many recorded tracks).
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Working in 192KHz has an audible difference to working in 88.2KHz.
This one is much easier. Humans with perfect hearing can't hear above 22KHz. A sampling rate four times as high ensures down to a mathematical certainty that you will hear that perfectly. Anything beyond that is inaudible by definition, we just can't hear those high frequencies or even the influences of neighboring frequencies and the such beyond 88.2KHz.
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The reason many converters sound better at higher sampling rates has nothing to do with us hearing frequencies above 22Khz. It has to do with the design of the anti-aliasing filters in the ADCs. Of course, the top converters have well designed decimation filters which don't need to run at higher sample rates.
Also, there is an advantage to running certain processes at high sampling rates. Non-linear processes benefit from higher sampling rates. Some plugins upsample the signal before processing. This can be beneficial but only if the interpolation and anti-aliasing filters are well designed.
Some people argue that it is safer to run the whole DAW at higher sampling rates to avoid cumulative degradation from poorly designed up/down-samplers in plugins. (I don't entirely agree but the arguments are valid in themselves).
UnderTow |
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UnderTow
Started Topics :
9
Posts :
1448
Posted : Sep 3, 2008 00:26
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On 2008-09-01 19:38, Upavas wrote:
+3 db doubles the intensity of sound. +6 db doubles the perceived loudness to the human ear!
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+6 dB is doubling in voltage. +10 dB is doubling in perceived loudness. +3 dB is the increase in level when summing two non-identical signals.
UnderTow |
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UnderTow
Started Topics :
9
Posts :
1448
Posted : Sep 3, 2008 00:33
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On 2008-09-01 22:21, Spindrift wrote:
32 bit fixed is AFAIK know only used in some DSP processors like SHARC and not in any native software.
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SHARC DSPs use 32 bit float. 
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Also if your DAW work in 32 bit float it makes sense to stick with that when recording as well.
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That is a waste of disk space. The way modern CPUs are designed, converting 24 bit to 32 bit float happens during the memory allocation cycle (which will happen anyway). In other words, it costs nothing.
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@Upavas:
Are you sure most plugins are 24 bit....seems odd to me since at least the VST format use 32 bit float for the input and output to the plugin.
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Or 64 bit float. (VST 2.4 and above).
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I know some plugs use fixed point processing, like for example waves that use 48 bits, but it seems a bit odd to implement 24 bit processing in a plugin for a 32 bit float system.
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I am not aware of any true VST plugin that doesn't use aither 32 or 64 bit float. Some of the Wave plugins do indeed use 48 bit fixed point but they are not true VST plugins. (They are wrapped with the WaveShell to appear as VST plugins though).
UnderTow |
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UnderTow
Started Topics :
9
Posts :
1448
Posted : Sep 3, 2008 00:36
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I'll comment on the other misinformation tomorrow.
UnderTow |
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vegetal
Vegetal/Peacespect
Started Topics :
19
Posts :
1055
Posted : Sep 3, 2008 00:52
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