Work at 24bit 96KHz?

[VRW]

Hi there, just a short question.
How many of you do record/mix etc. at 24bit/96KHz these days just for audio (no film etc.).
Does it make sense generally?
Asking also because the "Mastered for iTunes" thing is kinda topic out there though as far as I know most of the
digital distributors still demand the files for uploading as 16bit/44.1KHz.

I´m used to record at 24bit/44.1KHz-at first because I´m only doing audio and secondly it feels kinda safer and less complicated
to me when converting it finally just from 24bit/44.1KHz to the standard 16bit/44.1KHz for the final product.

A further question is, that I read and heard it to be much safer and more reliable to use 88.2 etc. if the final product is at 44.1
instead of converting from 96KHz etc. to 44.1 (because of the linear math thing or something).
Is this still right for the average producer (the really high end guys of course
do have the equipment as well as the skills to do almost everything , I certainly know)?

So I´m just curious what´s the actual situation these days?
What do you think about it?

Thanks in advance, best regards,

VRW



MacMini i7Quad, 16GB Ram, MacOS 10.9.1, Apogee Quartet, Genelec Active, Yamaha NS10, Pro Tools 11.0.3, Logic 10.0.5, Waves, MCDSP, Duende Native, Softube, HOFA, IK Multimedia etc.

[JFreak]

I most often work with 24bit and 48kHz sessions. In case client pays me extra, I'm using 96kHz but no way I'm halving my system resources for free. There is one exception, and that is monitoring latency. If the artist is too picky about the latency, I might just record at 96k to make latency smaller, but then I convert to 48k for mixing.

I never do 44.1/88.2 sessions, instead I always convert to 48/96. Makes my life easier in the long run.

Sample Rate Conversion from 88.2 to 44.1 is not easier than 96 to 44.1 or 48 to 44.1 -- it is a myth that somehow 44.1*2=88.2 would make SRC easier. It doesn't work that way. In a properly done SRC, the actual waveform is reconstructed and then resampled to whatever target sampling rate you choose. Plugins also do SRC all the time. Many best-sounding plugs upconvert and process in higher sampling rate and then downconvert back.

[WombatStudio.Org]

Originally Posted by JFreak

If the artist is too picky about the latency, I might just record at 96k to make latency smaller

Everything I've read suggests this would double the latency.

[JFreak]

Originally Posted by WombatStudio.Org

Everything I've read suggests this would double the latency.

Nope. Latency of ADDA conversion is cut to half if you double the sampling rate. That is because converters work in samples and not milliseconds. Same amount of samples is dealt with faster if you up the SR.

[Raoul23]

I saw an interview with Mike Shipley (god bless him) where he said that he works in 96/24 because some plugins sounded better at 96.

Nice to no about the higher sample rate the lower the latency :)

[albee1952]

99% of my work is 24bit/48K.

[ArKay99]

I work at 24/96 all the time. I then mix 24/96k, then dither that to 16/44.1k for my CD mastering. It sounds better to me than if I work at 48k.

I did some basic tracks at 48k and at 96k. For 'similar' output meter ballistics the 96k recording sounds 'louder' and has more detail...reverb tails sound deeper and the stage around the sound seems more detailed. Maybe it's better transient response or the plugs I was using run at 96k so there was no src happening there, but it sounds better to me.

The downside for me is X-Form takes days to complete even small sections, and yes it uses much more cpu at comparable buffer sizes. And there is twice the audio data to deal with when working with it.

Maybe in the end, after it ends up 16/44.k and dynamically and sonically altered it makes no difference, I don't know. However, to me, if it sounds better before processing, it should sound better after the same processing...even if it's only a little bit. As stated above it's really a trade-off as to available resources vs. quality, vs. normal workflow.

[lexaudio]

Originally Posted by JFreak

I most often work with 24bit and 48kHz sessions. In case client pays me extra, I'm using 96kHz but no way I'm halving my system resources for free. There is one exception, and that is monitoring latency. If the artist is too picky about the latency, I might just record at 96k to make latency smaller, but then I convert to 48k for mixing.

I never do 44.1/88.2 sessions, instead I always convert to 48/96. Makes my life easier in the long run.

Sample Rate Conversion from 88.2 to 44.1 is not easier than 96 to 44.1 or 48 to 44.1 -- it is a myth that somehow 44.1*2=88.2 would make SRC easier. It doesn't work that way. In a properly done SRC, the actual waveform is reconstructed and then resampled to whatever target sampling rate you choose. Plugins also do SRC all the time. Many best-sounding plugs upconvert and process in higher sampling rate and then downconvert back.

No disrespect J, but there is a difference.

Here is the bottom line. In a proper room you will hear the subtle differences of recording at 88 or 96, granted that you aren't recording with crap gear.
The big reason for recording at 88.2 is that it is perfect math. 88.2 half is 44.1. There are no rounding errors.

48k to 44.1, is so small, your never gonna notice.

However, all the mastering engineers I have worked with have all said, whatever you record at, let them SRC.
Not all SRC's are the same.
If PT was perfect at SRC, why aren't the mastering engineers using it?

Weiss is far better in SRC. There are others that are better. Granted, if you aren't in a room where you can hear it, it doesn't matter.
It ends up listened to on ear buds.

But, listen to Daft Punk RAM. Listen to how in translates from ear buds, to laptop, to fair speakers to your studio.

96 to 48 is perfect math. But ultimately you still have to goto 44.1. That is 2 times conversion with a smaller rounding error.

Granted, this is all probably so trivial. This and that, 24, 32. 48. 96.

From my own experience and recording, there is subtle difference between 88.2 and 96k.
Pretty much to the point you really can't tell.

However, the difference between and 44.1/48k and 88.2/96k you can hear, given the room and speakers as well as well recorded material.

I stopped recording at 48 and went 88.2. I really can't say if mathematically it makes that much a difference but it was what I chose to do, but its only one conversion versus two, so I deliver 88.2 32 bit float mixes, and the mastering engineer prefers it at least according to him.

So whatever path you chose, these are some facts. Besides. Lower latency recording, less upsampling in the plugin as as well.
I notice a difference.
Random listeners notice the difference when we recorded at 48k vs 88.

Again, the room has alot to do with it, converters, ect.

Bottom line, if you can't hear the difference, and it is subtle, then it isn't worth doing. 48k will be fine.
There is no need to sweat over what you can't hear.

[propower]

Originally Posted by JFreak

Nope. Latency of ADDA conversion is cut to half if you double the sampling rate. That is because converters work in samples and not milliseconds. Same amount of samples is dealt with faster if you up the SR.

In the AVID I/O the A/D D/A conversion time is uniquely different at 44.1k and 96k giving more conversion time reduction at 96kHz than one might expect.. The conversion time is roughly = to the LLM delay.

44.1k conversion time = 1.9ms
96kHz = 0.47ms

Total latency in the system is the conversion time + (2X the chosen buffer)/(SR) + any delay for getting data on/off the relevant bus (PCI, FW USB etc). For PCI or TB systems the data bus delay can be assumed to be zero.

So for instance: Lowest possible latency with HD Native TB at 96kHz is 1.8ms (64 sample buffer) and at 44.1 it is 3.35ms (32 sample buffer). I work at 96k all the time solely for latency considerations. FWIW 44.1 on old HD systems was sonically fine for me before I went Native.

[Meads]

There's a fantastic article, that everyone should read, who wants to know the deal about SRC. Seriously, this explains it spot on. It's longish, but worth the read.

http://www.trustmeimascientist.com/2...-when-it-isnt/