What Sample Rate Do You Work In?

question: why manufacturers produce sound cards to 192kHz ? who is using these samplerates rattle-mouse ?
 
question: why manufacturers produce sound cards to 192kHz ? who is using these samplerates rattle-mouse ?
They just sell better when labeled "192 Khz ULTRAHIGH QUALITY !!! "
:lol:
But i think if you're doing recordings just for a DVD, especially for a HD-DVD, higher samplerates could be useful? Not shure about that, though..

(44.1 / 24 bit here, too)
 
It's interesting,

I pretty much feel the same way that working in 44.1 at 24bit is more than enough to give you a quality end product.Most times i prefer to work in this format,however for some reason,clients feel that a higher sample e.g 48k-88.2 will give them a better outcome.Today we took a client over to another studio to track a grand piano and the engineer loaded the session at 96k-24b.
On the way home my business partner was suggesting we do this on some upcomming recordings,i don't believe it will make a difference.
Hence i thought i'd find out how you guys felt about it,and if you did work in 88.2-96k did you feel that you had some sort of edge over recordings done at say 44.1 or 48k.
 
am i right in saying that the digital 44.1 khz approximation to a sine wave at 22.05 khz with amplitude 1 would be:

1, -1, 1, -1, 1, -1 etc.....

EDIT: those numbers are amplitudes BTW
 
No, you're not correct. There's 44100 samples taken per second (steps) and the height depends on the bit depth, eg. 24-bit -> 2^24 = 16777216 steps for peak-to-peak values. In other words, it's extremely close to an analog sine wave with that precision. The amplitudes would be more like (starting from origo):

0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0010
0000 0000 0000 0000 0000 0100
0000 0000 0000 0000 0000 0101
...
1111 1111 1111 1111 1111 1111
...
0000 0000 0000 0000 0000 0010
0000 0000 0000 0000 0000 0000
1000 0000 0000 0000 0000 0010
1000 0000 0000 0000 0000 0100
1000 0000 0000 0000 0000 0101
...
0111 1111 1111 1111 1111 1111

To get the rough idea. Those are binary values of course. In decimals it's something like 0, 0.000014, 0.000020, 0.000026, ..., 1, 0.999994, 0.999988, ..., 0, -0.000014, -0.000020, ..., -0.999988, -0.999994, -1

All the values are just some random rough numbers to display the idea. The actual values would be calculated with Fourier and difference equations with programs like MatLab.
 
I guess most people here work for CD releases, so there's no point in using non-native sample rates, because you'll have to downsample it to 44,1 kHz anyway for the CD.

Except when mixing out of the box.

44,1khz 24Bit here.
 
44,24 unless the drum tracks I received were recorded at 48/24 in which case I record the rest in 48 as well. When I'm done with the mix I use r8brain in very high quality mode to downsample and I feel it does a perfect job.

If you're wondering about something you can always try to find out how the pros are working. If I recall correctly, Andy uses 44khz,24bit, as well.
 
No, you're not correct. There's 44100 samples taken per second (steps) and the height depends on the bit depth, eg. 24-bit -> 2^24 = 16777216 steps for peak-to-peak values. In other words, it's extremely close to an analog sine wave with that precision. The amplitudes would be more like (starting from origo):

0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0010
0000 0000 0000 0000 0000 0100
0000 0000 0000 0000 0000 0101
...
1111 1111 1111 1111 1111 1111
...
0000 0000 0000 0000 0000 0010
0000 0000 0000 0000 0000 0000
1000 0000 0000 0000 0000 0010
1000 0000 0000 0000 0000 0100
1000 0000 0000 0000 0000 0101
...
0111 1111 1111 1111 1111 1111

To get the rough idea. Those are binary values of course. In decimals it's something like 0, 0.000014, 0.000020, 0.000026, ..., 1, 0.999994, 0.999988, ..., 0, -0.000014, -0.000020, ..., -0.999988, -0.999994, -1

All the values are just some random rough numbers to display the idea. The actual values would be calculated with Fourier and difference equations with programs like MatLab.


thanks, good explanation :)

however, i did say a 22.05khz sine wave

what you just described was an approximation to a sine wave with frequency 44100/(2*2^24)= 0.001314 Hz

(for anyone following along, that was the number of samples per second, divided by the total number of different displacements multiplied by 2 because a sine wave covers each displacement twice)



and "from origo"??? is that like "a priori"?
 
seeing as you seem to know a lot more about the technicalities of this than me.......

are these statements correct

1. A sine wave of frequency 22.05 khz has period 1/22050 = 4.54 x 10^-5 seconds

2. The spacing of the samples when recording at a sample rate of 44.1khz is 1/44100 = 2.27 x 10^-5 seconds

3. 4.54 x 10^-5 = 2 x 2.27 x 10^-5 Therefore 1 complete oscillation of a sine wave of frequency 22.05khz recorded at 44.1khz will consist of only 3 samples

i hope there's a mistake there somewhere, otherwise there are serious consequences for recording at 44.1khz IN THEORY
 
1. A sine wave of frequency 22.05 khz has period 1/22050 = 4.54 x 10^-5 seconds

True

2. The spacing of the samples when recording at a sample rate of 44.1khz is 1/44100 = 2.27 x 10^-5 seconds

True

3. 4.54 x 10^-5 = 2 x 2.27 x 10^-5 Therefore 1 complete oscillation of a sine wave of frequency 22.05khz recorded at 44.1khz will consist of only 3 samples

According to the Nyqvist-Shannon Theorem, that is enough.

i hope there's a mistake there somewhere, otherwise there are serious consequences for recording at 44.1khz IN THEORY

But no human hears 22 kHz. Theoretically the ear can detect (barely) 20 kHz, while most people 12 years old and older will commonly only hear up to 16-18 kHz. In music, the "money shot" range is probably about 100 Hz - 10 kHz
 
According to the Nyqvist-Shannon Theorem, that is enough.



But no human hears 22 kHz. Theoretically the ear can detect (barely) 20 kHz, while most people 12 years old and older will commonly only hear up to 16-18 kHz. In music, the "money shot" range is probably about 100 Hz - 10 kHz

Ok, i get what the Nyquist-Shannon sampling theorem is, although i couldn't follow the arguments for why it's true

In my simplistic way of looking at things, the 3 samples i mentioned before if you join the dots you get a triangle wave. Which is a pretty shocking approximation of a sine wave
 
96k recording

44.1 mixing


Why? Because I can. lolz

No srsly. I record at 96 because the amount of samples taken a second gives a clearer representation of the waveform as it was in the analog domain. Then I just edit in 96 and convert to 44.1 (so the SRC to 44.1 has more samples to chose from)