I have been using 16bit/44.1 Hz resolution for years. Now I would like to switch to higher bit/resolution. Do I use 24/48, 24/96 or else? Thank you.

24 bit will give you better specs on the way into your DAW. ie better noise floor vs headroom ie a greater dynamic range allowing you to record at lower levels without introducing noise or dreaded digital overs (distortion)

Sample rate really depends on your intended medium.

CD 44.1
DVD 48.

double the resolution for better high end resolution (theoretical) and if you have double the disc space.

I've deliberately tried to not get too technical and approached in very much layman's terms.

I used to use 48/24 but I'm now using 44.1/24.

Bit depth is arguably more important than sample rate (at the recording stage). I'd recommend using a minimum of 24 as a bit depth. You can always dither it down later.

Thank you for your replies. I agreed to try 24 bit, but what about sample rate? Media? I make music, no video, and was alwats thinking about it in CD terms- this is why 16/44.1. Where people store music now days on (other than iPad)? We still save to CD, not DVD, right?

How about MP3s? I know it used to be 128, then 256, BUT: from what? Do we convert .wav to 16/44 CD format before compressing/ What if I have 48/24, can I compress to MP3 and be compatable with downloading web sites such as iTunes? thanks.

48kHz /96kHz 32bit /64bit.

48kHz / 96kHz 24bit/32bit are standard in any professional studio. 


After years of stupid debate, nyquist freq, try convince the unbelievers it is nowdays a common knowledge and a fact that higher sample resolution and bits you use you better sound of the final mix and end result you will have. If you missed all years of "VS threads" you must been living under rock?!  
 
Why it sounds better? It's becasue all software filters, software synths and all other analog gear and recordings are captured in higher resolution. I'm not saying use 192kHz but you should at least use 48kHz or 96kHz 24bit or 32bit as standard.


http://www.sounddevices.com/notes/recorders/real-world-24-bits/
http://www.gearslutz.com/board/                          *make a search here and you can hear audiofiles too... VS 44


So your lowend MP3 files will still sounds better using from 48kHz /96kHz or 192kHz 24bit / 32bit VS down to CD 44.1 kHz 16bit/ MP3, DVD 24bit.
MP3 are by the way in 48kHz 16bit format.

vladasyn


Thank you for your replies. I agreed to try 24 bit, but what about sample rate? Media? I make music, no video, and was alwats thinking about it in CD terms- this is why 16/44.1. Where people store music now days on (other than iPad)? We still save to CD, not DVD, right?

How about MP3s? I know it used to be 128, then 256, BUT: from what? Do we convert .wav to 16/44 CD format before compressing/ What if I have 48/24, can I compress to MP3 and be compatable with downloading web sites such as iTunes? thanks.

I think you are confusing bit rate with bit depth, the 128 and 256 in mp3s are bit rate which is how many kilobits per second are used to reproduce the sound. Obviously the more bits used the more accurate the sound.

Bit depth is how much room there is to store a sound so a 24 bit file depth has more "room" than a 16 bit one. Get anything above 24 and the increase is academic as there are no convertors that can convert to a greater depth anyway.

It's also different to sample rate of say 44.1 or 48 which is how many times per seconds the sound was sampled.

Bit rate is basically equal to - Sample rate x bit depth x number of channels.

I personally can't tell the difference between a sound sampled at 44.1 or one at 48 or higher. You'll need to ask Freddie about that.

FastBikerBoy

vladasyn


Thank you for your replies. I agreed to try 24 bit, but what about sample rate? Media? I make music, no video, and was alwats thinking about it in CD terms- this is why 16/44.1. Where people store music now days on (other than iPad)? We still save to CD, not DVD, right?

How about MP3s? I know it used to be 128, then 256, BUT: from what? Do we convert .wav to 16/44 CD format before compressing/ What if I have 48/24, can I compress to MP3 and be compatable with downloading web sites such as iTunes? thanks.

I think you are confusing bit rate with bit depth, the 128 and 256 in mp3s are bit rate which is how many kilobits per second are used to reproduce the sound. Obviously the more bits used the more accurate the sound.

Bit depth is how much room there is to store a sound so a 24 bit file depth has more "room" than a 16 bit one. Get anything above 24 and the increase is academic as there are no convertors that can convert to a greater depth anyway.

It's also different to sample rate of say 44.1 or 48 which is how many times per seconds the sound was sampled.

Bit rate is basically equal to - Sample rate x bit depth x number of channels.

I personally can't tell the difference between a sound sampled at 44.1 or one at 48 or higher. You'll need to ask Freddie about that.

Have in mind all your software mixers / DAC to all you Audio interfaces, Lynx, RME totalmix, SSL mixtreme use often higher internal resolution over 32 bit, 40bit sometimes even higher bit then 64bit / 128bit. Why do you think they use that? Because its just so fun using many bits? No its a reason!
 

Internal

The essential criteria for the power of a digital mixer are the number of channels that must be processed simultaneously, and the internal resolution. Example Digiface: 26 inputs plus 26 playback tracks can be mixed to 28 physical outputs in any order. This means (26 + 26) x 28 = 1456 combinations simultaneously possible. The mixer must be able to calculate 1456 channels!
As long as monitoring would have been the only point for this concept, one would have got away with 16-Bit resolution. Including the playback tracks, this is no longer possible. Because they are always routed through the mixer, it does not only have to have higher resolution than 24-Bit in order to avoid rounding errors for gain changes, but it also has to supply additional headroom for gain rises and the summing of several channels.
Extremely high resolution does not make sense and causes a high consumption of resources within the FPGA. We have therefore decided to have a basic fixed-point arithmetic with the following values:

The multiplier has 40 bit resolution. This comes from a maximum audio resolution of 24-Bit, that can be altered in gain with a resolution of 16-Bit (65563 steps) with a fader (24 + 16 = 40.)

The adder has 36 bit resolution. From these, 7 bit are used for the necessary headroom (up to 52 channels with 6 dB gain can be mixed to one channel.) The audio signal has again 24-Bit resolution, the other 5 bit are used for the LSB to decrease rounding errors.

The output signal is truncated to 24-Bit without dither.

Faders: 16-Bit resolution, equals 65536 steps, which are provided following a special function for correct perception. Gain change input and playback channels: +6 dB down to maximum attenuation. Outputs: 0 dB down to maximum attenuation. Because of the screen display, the number of values that can be set with the mouse is dependant on the number of pixels. This is 137. In fine-mode (shift key), this value is multiplied by 8 for 1096 different gain values.

Pan: pan law 6 dB. This means, the signal is attenuated by 6 dB in center setting. From center to left and right, there are 50 steps on either side. Linearly processed and displayed, from L 1.00 via 0.50 (C) to R 1.00.

Did anyone see the article in Mix a few years back where they sat down several "golden ears" types, veteran engineers, producers and audiophiles, and asked them to evaluate recordings made at various sample rates and bit depths?  The results were a bit surprising.  In many cases, no difference could be perceived between recordings made at 16/44.1 and those at higher bit depth and sample rate.  They were using the finest A/D and D/A, audiophile speakers, etc. across a broad spectrum of musical genres.  I remember leaving the article with a sense that the popular bit depth and sample rate escalation had points of diminishing returns.  For me personally, I find 24/44.1 a completely satsfying balance between audio fidelity and computer resource usage.

Yes Freddie but there is a difference between making calculations at a bit depth and storing the result.

Yes bitSync I saw that article and it is very interesting. It is called The Emperor's New Sampling Rate and here tis:

http://mixonline.com/reco...w_sampling//index.html

I have done a similar experiment except I used a very high quality turntable, SME arm with a Shure V15 Type III pickup feeding a $1000 Fidelity Research RIAA preamp as the source.

For vinyl I used those Sheffield Lab albums that were recorded direct from the studio to the cutting lathe with nothing in between. (probably highest quality vinyl you will ever hear in your life! Makes Dark Side sound like crap and that has to be hard! I also used Dark Side as a reference)

I have also used analog two track masters running at 30 ips on a half inch mastering tape machine with super fidelity masters as an alternative analog source.

So I organised these analog sources onto one side of an A/B switch. I also fed the analog source into my Yamaha digital mixer and did A to D and D to A at 16 bit 44.1 KHz. This I fed to the other side of the AB switch. 

Once all the levels were perfectly matched no one could pick either source! I was switching blind in the middle of the music. No clicks or gaps etc. I have tried some serious engineers and also some expert hi fi enthusiasts.  In case you were wondering I used Mackie HR 824's as one set of speakers but also a class A valve amp (Williamson) feeding Quad Electrostatics as another speaker option and the listening conditions were pretty ideal too.

Tells you a lot about how good 16 Bit 44.1 K Hz is doesn't it. Sort of debunks the 96 Khz  24 Bit myth. If I could do it all again especially to Freddie he would fail this test a million times! Sorry Freddie but true. You and everyone else here my friend!