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

[amrrahmy]

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?

[Andrew M.]

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.

[amrrahmy]

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)?

[Philippe Vandendriessche]

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.

[amrrahmy]

thanks dude.

[Andrew M.]

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.

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.

[Kreisky]

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

[Kevin Halverson]

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

[amrrahmy]

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.

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)

[Kevin Halverson]

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)

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