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  • Hey all, just changed over the backend after 15 years I figured time to give it a bit of an update, its probably gonna be a bit weird for most of you and i am sure there is a few bugs to work out but it should kinda work the same as before... hopefully :)

24bit/96khz & 192khz sorry for audio noob question

A

amrrahmy

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how much of a difference is there between one channel 24bit/96khz and 192khz

is studio quality 24bit/96khz considered a good quality for a film?
 
Well, it depends what is your final product.
If it will end up on Blu-ray in 5.1 surround then go for 96 or even 192.
It mean, you can bring audio on 48 but for manipulation in post use 96 minimum.

Like with the video, Mysterium is 12 bit but color manipulation, we do in 32 bit if possible.
 
96 minimum, meaning that there would be a clear difference in quality between 96 192.
is that difference also noticeable for on set recording(if on set was 96 and the rest of the sountrack and effects were higher quality)?
 
We mostly record 24 bits 48 kHz during shooting.
Good enough for dialog...

Music recording / sound effects: 24/96 is better.

I have never seen any project recorded in 192. I think the human ears cannot hear any difference between 192 and 96 recordings.
Between 16 & 24 bits there is a big difference.
 
Philippe Vandendriessche said:
We mostly record 24 bits 48 kHz during shooting.
Good enough for dialog...

Music recording / sound effects: 24/96 is better.

I have never seen any project recorded in 192. I think the human ears cannot hear any difference between 192 and 96 recordings.
Between 16 & 24 bits there is a big difference.
Click to expand...

Human ear maybe can't see the difference between 96 and 192 in terms of frequency response but in terms of phase of the signal coming to your ear we can detect 20 deg difference in the phase.

Just make simple experiment.
Get the software for surround sound editing and go to the sound placement editor.
Human can detect the direction of the sound based on the main 3 factors and number of secondary factors.
First is ..............no no loudness of the signal it is phase of the signal
Second, loudness
Third, frequency response of our ear lobe.

Now change the placement of the 2000Hz sound just by altering the phase by 20 deg.
Now switch it on and off, can you hear the difference?

We can hear 2000 Hz in the music very well. To reproduce the phase of this frequency you have to sample it say 3 times per each degree of the phase you want to detect.
Well 2000 x 3 x (360/20) = 108,000 (108kHz) Now what if you want to do it with 4000 Hz sound?

So even for the interview on the camera when I want to reproduce in just two channels the position of the speaker and then if I do zoom out on his face and zoom out to show the whole street where interview takes place I use phase shifting. Viewer expect sound to focus on the speaking person when you zoom in and to hear more of surround sounds when you zoom out.
 
48.000khz is broadcast standard !

I have a question to the sound guys here in forum:"
I think most tracks are mixed out at 44.000khz in a music studio,
is this for CD produktion ?
or do you mix them for music videos out on 48 ?"

thnx
bruno
 
The value that higher sample rates (above 48 kS/s) provide are much less about the ultrasonic content than they are about improving the capabilities of the anti-aliasing filtration mechanism. Given that to satisfy the Nyquist requirement the amplitude of the inbound signal has to be at or below the noise floor by .5 F, this requires a fairly aggressive LP filter when sampling at single speed data rates (44k1 or 48K). Consider that the same order of LP filter can be used with higher sample rates (2x or 4x) and achieve much improved image rejection while still providing adequate pass band performance.

The "phase" argument is not accurate as it misses the very nature of a sampled system. If one modulates the amplitude of a monotonic stimulus "...switch it on and off...", the resultant output has higher order components. The slope of the amplitude modulation is just as bound to the Nyquist requirements as is the fundamental energy component itself. A very simple illustration of this is RF amplitude modulation (AM). Obviously the carrier is monotonic but as soon as modulation is introduced sidebands appear. These sidebands exhibit increased bandwidthrequirements; the exact same effect occurs in the "...switch it on and off..." situation. On and off imply square wave amplitude modulation and this modulation profile requires a much greater bandwidth system to capture the modulation. In fact, to completely capture a perfect square wave modulation (or fundamental) the sample rate would have to approach infinite.

"Phase" is a term that is often misused and in fact, there is no actual property called "phase" rather it is the relationship of a signal or signals to another. An example of its misuse is "invert the phase" when what should have been said was to invert the absolute polarity.

Directional localization in human hearing relies upon several different systems; including arrival time differences (this is why localization of impulse sound events is more pronounced) relative amplitude differences (but only at midband and higher frequencies) and by moving the receptors (changing head position which is often done without a conscious decision on the part of the listener) and listening for the delta in the arrival events. Human hearing is also sensitive to absolute polarity (at least some humans are) but the original signal must be both recognizable and have significant asymmetry for this to be much of a contributor.

The argument for greater bit depth is certainly easy to make as this is simple resolution improvement, the argument for higher sampling rates is more complex for the reasons that I have sighted and many others.

In my work, we often use extremely high sample rates (8x or 384 kS/s) for the most important recordings, but we rarely deliver this all the way through to the end product (often DVD-A recordings). Rather we use the very high sample rates to ensure that we have nearly perfect passband performance while still satisfying the stop band needs for the bit depth that we are utilizing.

So, is 48 kS/s enough? It really depends upon the event being recorded, but given the types of microphones used in sound for film field recording, it would be more difficult to justify higher sample rates than for situations where transducers that have more extended bandwidth are employeed. Greater bit depth (24b) is easily justified no mater the sample rate.

Kevin Halverson
 
Andrew said:
Human ear maybe can't see the difference between 96 and 192 in terms of frequency response but in terms of phase of the signal coming to your ear we can detect 20 deg difference in the phase.

Just make simple experiment.
Get the software for surround sound editing and go to the sound placement editor.
Human can detect the direction of the sound based on the main 3 factors and number of secondary factors.
First is ..............no no loudness of the signal it is phase of the signal
Second, loudness
Third, frequency response of our ear lobe.

Now change the placement of the 2000Hz sound just by altering the phase by 20 deg.
Now switch it on and off, can you hear the difference?

We can hear 2000 Hz in the music very well. To reproduce the phase of this frequency you have to sample it say 3 times per each degree of the phase you want to detect.
Well 2000 x 3 x (360/20) = 108,000 (108kHz) Now what if you want to do it with 4000 Hz sound?

So even for the interview on the camera when I want to reproduce in just two channels the position of the speaker and then if I do zoom out on his face and zoom out to show the whole street where interview takes place I use phase shifting. Viewer expect sound to focus on the speaking person when you zoom in and to hear more of surround sounds when you zoom out.
Click to expand...

i think that goes to a point and stops.
u hear a diff sound because u changed the original frequency,(i think) the lower u go the more it gets pitched but when u reach the original freq the sound is the same, there is no change in quality.
what i was wondering was, if i recorded both 96 and 192(i wanted to know if i can record with a hardware that does 24bit/96khz). would i find a diff quality, or would it be the same?

not if i converted a 192 file to a 96khz, of course there would be loss from the original file and i would notice the diff, but that's due to the software and how the files were recorded and not the max ability of the 96khz(i think)

i wanted to record (24bit/96 if possible with 3 "duet" by apogee with a macbook pro)
 
amrrahmy said:
i think that goes to a point and stops.
u hear a diff sound because u changed the original frequency,(i think) the lower u go the more it gets pitched but when u reach the original freq the sound is the same, there is no change in quality.
what i was wondering was, if i recorded both 96 and 192(i wanted to know if i can record with a hardware that does 24bit/96khz). would i find a diff quality, or would it be the same?

not if i converted a 192 file to a 96khz, of course there would be loss from the original file and i would notice the diff, but that's due to the software and how the files were recorded and not the max ability of the 96khz(i think)

i wanted to record (24bit/96 if possible with 3 "duet" by apogee with a macbook pro)
Click to expand...

Your example illustrates my previous point. If you record something at 192 kS/s and then later properly down sample it to 96 kS/s (or lower) you will retain much of the perceivable benefit of the higher recording sample rate. This is mostly due to the improvement of the performance of the anti aliasing filtration mechanism and not due to the increased ultrasonic content of the original higher sample rate (since it will not survive the down sample process).

By the way, the correct way of expressing sample rate is not to use frequency units (Hz) but to use samples per unit time (S/s or Samples per second). Instead of 96 kHz, the correct term is 96 kS/s. This is an industry wide misuse of a term that has persisted for years (with a few notable exceptions).

Kevin Halverson
 
sorry i was writing when u posted, so i missed it. i only saw it after.
 
384 kS/s that hardware is not something portable i guess(right?), u cant record outside the studio with something like that?, can u even use that on a closed set in a shoot?
 
khmuse said:
The "phase" argument is not accurate as it misses the very nature of a sampled system. If one modulates the amplitude of a monotonic stimulus "...switch it on and off...", the resultant output has higher order components.
Click to expand...


Phase term I used as a time of arrival of sound to our ears.
It is the most directional information used by our brain, not the amplitude.
You do not need to switch it ON or OFF, just move the phase knob by 20 deg and you will see if 48kHz system will take it. Then try the same on 192.
 
Andrew said:
Phase term I used as a time of arrival of sound to our ears.
It is the most directional information used by our brain, not the amplitude.
You do not need to switch it ON or OFF, just move the phase knob by 20 deg and you will see if 48kHz system will take it. Then try the same on 192.
Click to expand...

It is the relative arrival time, not PHASE, that human hearing uses as a localization clue. Amplitude is but one of the inputs used to localize a sound source, delta T is another. The dominance of the importance is dependent upon the spectral characteristics of the signal (in essence its frequency components) and is dominated by the shape of the human head.

Your use of the term "phase" is yet another commonly misused one in audio. When one rotates a "phase knob" what is actually occurring is the changing of the delta T parameter of an all pass filter (in essence delaying a signal by a given number of samples in a digital system or by an R/C time constant in an analog one).

"Phase" or angular displacement of a signal is only applicable when it is compared to another signal! When you are noticing the localization change when you rotate a "phase knob" this is because some other signal (perhaps the other channel) is unchanged, hence there is an arrival time difference between the two channels. Don't become confused by the misused terms in most DAWs (and having designed a lot of the software and hardware that is used by audio professionals, I know how easily these terms get misused).

Regardless of it being mislabeled "phase" in your DAW, it is the arrival time delta that provides the audible effect. You can easily prove this to yourself by routing a signal through a "phase control" and listening ONLY to this signal (no other path). Other than the latency, there is no other change to the character of the signal.

Kevin Halverson
 
amrrahmy said:
384 kS/s that hardware is not something portable i guess(right?), u cant record outside the studio with something like that?, can u even use that on a closed set in a shoot?
Click to expand...

Currently, products based upon Si that is capable of 384 kS/s data rates are confined to non portable products (and there are but a very few of these currently available). There will be some portable ones introduced early next year (2009) assuming that the development continues on its current pace.

Again, the advantage of these types of sample rates is the improved image rejection made possible by the improvement to the stop band characteristics of the LP filters that proceed the ADC. Currently, there are no delivery vehicles for this type of data rate and in the standards committees that I work with, none are contemplated at this time.

Kevin Halverson
 
dude i'll have some problems just recording 5.1ch audio with a macbook pro, and i'm not sure if i'll also need a san or raid system to go with the macbook pro. 384 is not something for film in the near future, it's just too much for todays hardware to handle if it would even make a difference in the quality which is kindof doubtful
 
amrrahmy said:
dude i'll have some problems just recording 5.1ch audio with a macbook pro, and i'm not sure if i'll also need a san or raid system to go with the macbook pro. 384 is not something for film in the near future, it's just too much for todays hardware to handle if it would even make a difference in the quality which is kindof doubtful
Click to expand...

The great thing is that you don't have to record (store) high data rate material (96 kS/s, 192 kS/s, 384 kS/s) if you do a proper down sample first!

This is the wonderful thing about the concept, if the ADC runs at high data rates and the anti aliasing filter is designed to assure that the stop band performance meets the requirement of the bit depth you are recording, then all that is necessary is to pass the samples through to a highly accurate low pass filter (either hardware or software DSP), you can store (record) the data at much lower rates and still retail nearly all the benefit. I can assure you that many listeners can easily perceive the differences that this type of process makes (and I am not just talking about the "golden ear" types).

Don't worry about today's hardware not being capable of accomplishing this, it actually already is capable and this type of recording systems are coming from a number of manufacturers.

Kevin Halverson
 
not something that u can attach to a camera.
and i think what u say would have some delay, not something that u can notice easily, however there would be some 0.0000etc ms delay(right?).
 
amrrahmy said:
not something that u can attach to a camera.
and i think what u say would have some delay, not something that u can notice easily, however there would be some 0.0000etc ms delay(right?).
Click to expand...


Actually, some of these products are fairly small, the size of, or slightly smaller than many of today's field recorders. Though not intended to be mounted on a camera, they are certainly well within the size constraints of the intended application.

As for the latency of the process, its completely academic as it's in the recording signal chain. No matter the delay (up to a reasonable point) it's meaningless. Consider how long the delay is between the time something is recorded and when it's played back? This is not an issue at all.

Kevin Halverson
 
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