New presonus mixer

Re: New presonus mixer

Jeff Babcock said:
You're right. The one thing about 96K is the possibility of lower latency, however, that's about it. There are loads of examples that prove that most people (even those in the audio industry) can't reliably distinguish the difference between 320Mbps MP3 or 256 AAC versus uncompressed 192K, let alone between 48 and 96.

There are in theory benefits to these higher rates (if you have that capability already there is no reason not to use it), however, in the grand scheme of things, just about EVERYTHING else should matter more to you because it will make a much bigger difference than 96K vs 48K, despite what marketing departments will try to tell you.

Moving off axis by even just 1 degree on a microphone makes more audible difference if that says anything. The talent, the source, the acoustic environment, the speakers and their deployment, and many other things such as this are where significant improvements are possible.

I find myself posting about this a lot.... so many people are misled into spending money where it doesn't really make much difference - 96K, boutique preamps, converters, master clocks, etc, etc.

Spend wisely - those types of things are ones you should wait to spend on until you've addressed ALL of the other areas of greater importance I've mentioned first, regardless of whether we are talking live or studio (I do a lot of both)
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Thanks for the reply Jeff.

I can see how the latency could potentially be cut in half going from 48Khz to 96Khz, but I am wondering if that really makes any difference in a console that is sitting at ~1mSec total latency already. Additionally, all the internal latencies are supposedly time aligned and phase aligned "phase coherent" to prevent artifacts in processing.

I am pretty darned sure that no one can hear a frequency difference since most microphones and speakers can't reach 24Khz anyway (and certainly my ears can't any more ;) ) rather on double that.

IMO, the quality of the processing is what differentiates today's digital mixers. How well does the compressor actually work? How nice, lush and transparent are the verbs? How well does the channel PEQ perform. This along with the feature set and workflow are the biggest deal in my book..... not the processing frequency ..... or preamps for that matter.
 
Re: New presonus mixer

Scott Bolt said:
--snip--
I don't personally see any advantage to the 96KHz processing for live applications. Perhaps someone here with more knowledge than me can enlighten me.
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Scott, as I stated earlier, I'm near positive I couldn't hear any difference between 48 and 96KHz but, just to tap the extensive knowledge here, I seem to remember reading once (on the internet!) that modern compressors do a bit of "look ahead" to do their magic and given that wouldn't 96KHz give them a few more data points to look at than 48KHz? Or is that what you're accounting for in your statement that latency would be less with 96KHz ?

..dave
 
Re: New presonus mixer

dave briar said:
Scott, as I stated earlier, I'm near positive I couldn't hear any difference between 48 and 96KHz but, just to tap the extensive knowledge here, I seem to remember reading once (on the internet!) that modern compressors do a bit of "look ahead" to do their magic and given that wouldn't 96KHz give them a few more data points to look at than 48KHz? Or is that what you're accounting for in your statement that latency would be less with 96KHz ?

..dave
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Thanks Dave.

With 96KHz you could certainly get more data points for a look ahead; however, you couldn't get any more time without more latency. I am unfamiliar with the digital algorithms used for compression so I am not sure exactly how this would work.

There has to be a reason that DAW's are all 96KHz processing with recording applications.
 
Re: New presonus mixer

Scott Bolt said:
Thanks Dave.

With 96KHz you could certainly get more data points for a look ahead; however, you couldn't get any more time without more latency. I am unfamiliar with the digital algorithms used for compression so I am not sure exactly how this would work.

There has to be a reason that DAW's are all 96KHz processing with recording applications.
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DAW's can afford the latency. Look ahead is done in a buffer of the DSP. 96k is not needed for either of these to function well.
See Rupert Neve for an explanation of Dynamic range and both the speed and bit depth of digital coding and decoding.

i think of it as a staircase. One having 48k steps and one that has double for the same distance. The difference is resolution. In digital audio resolution increases dynamic range at high frequencies especially.
 
Re: New presonus mixer

Latency in a 96 kHz system would be half that of a 48 kHz system if and only if the clock speed of the processor doubled and all time constants in the processing were halved. Thus, double the high-pass frequency and half the attack times etc. Since, in most instances, this is not true, latency will only improve by the effect of doubling the speed of every one-sample buffer, and there are not that many, so typically you see only something like 0.1 mS improvement when the system is the same and sample clock speed is not altering system clock speed.
 
Re: New presonus mixer

The more this gets discussed, the less reasons I can come up with for 96Khz processing (aside from marketing).
 
Re: New presonus mixer

Scott Bolt said:
The more this gets discussed, the less reasons I can come up with for 96Khz processing (aside from marketing).
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It is better. MP3 at lower clock speeds doesnt sound as good as those coded higher. Not a great example but if it helps so be it.
 
Re: New presonus mixer

Scott Bolt said:
There has to be a reason that DAW's are all 96KHz processing with recording applications.
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Apart from the obvious, that people like to record at 96 and 192 KHz, one processing advantage of 96 KHz (or indeed 192 KHz) is that cumulative errors will be smaller, same thing with 24 bits.
By making absolutely sure that any errors (read noise and distortion), even after lots of processing will still be in the insignificant bits that are ultimately discarded when the master is converted to 44.1 kHz 16 bit, one can be confident that the end result is as good as it gets. In the old days (not really that long ago) one would sometimes utilize direct to master recording of symphony orchestras to ensure the quality of the recording. One would aim to eq only once, if at all, rather spend time to find the right microphone and placement, etc. etc., all to keep the path short and undisturbed.
Today there might be an exaggerated belief that as long as it is digital, it's error free and quality is not disturbed. It might be true in terms of harmonic distortion and noise, and some eq plug-ins are even very kind to phase, but it is still true that lots of cumulative eq processing smears phase, and lots of dynamic processing distorts dynamics beyond what is intended. Some of the processing methods that were available twenty years ago to avoid some issues have even gone out of fashion and practically disappeared because of the (erroneous?) belief that they are no longer needed.

Another reason for keeping all processing at 96 KHz is that you then only have to make (and optimize) the plug-ins for that sample-rate.
 
Re: New presonus mixer

Per Søvik said:
If I can hear the difference between 8 and 16 bit, that is proof that 32 bit is better than 16 bit :D~:-D~:grin:
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Ethan Winer (who has done AES presentations on audio myths etc) has an interesting video demo where he uses a plugin that allows you to alter the bitrate. He starts at 16 bit and gradually goes down one bit at a time. It is quite enlightening. I can't find the link or I'd post it here.
 
Re: New presonus mixer

Glenn Adams said:
Point taken. But If I can't hear the difference between 44.1 and 48 how will I ever benefit from 96?
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I'm not saying that you'll ever benefit from 96 other than in a theoretical way, but obviously going from 44.1 to 48 is such a small step that even if you couldn't hear that, you might hear a bigger step. Like you can't hear 0.1 dB but 0.5 dB is easily heard in an A/B even if some will argue that you can't hear less than 1 dB by definition.
 
Re: New presonus mixer

Jeff Babcock said:
Ethan Winer (who has done AES presentations on audio myths etc) has an interesting video demo where he uses a plugin that allows you to alter the bitrate. He starts at 16 bit and gradually goes down one bit at a time. It is quite enlightening. I can't find the link or I'd post it here.
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http://www.soundhack.com/freeware/the-boneyard/

MAC http://soundhack.henthorne.org/soft/+decimateX.sit.hqx

WIN http://soundhack.henthorne.org/soft/+decimateW.zip
 
Re: New presonus mixer

If I might add a quick signal processing query to the more educated here, does 96k help at all in the sense that a waveform might start between two samples? I know there's a reason this doesn't matter but I can't remember it.
 
Re: New presonus mixer

Max Warasila said:
If I might add a quick signal processing query to the more educated here, does 96k help at all in the sense that a waveform might start between two samples? I know there's a reason this doesn't matter but I can't remember it.
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http://en.wikipedia.org/wiki/Nyquist–Shannon_sampling_theorem

Going to 96khz sampling rate from 48khz sampling rate only helps if your signals have content above 24khz. It is left as an exercise to the reader to determine if audio for humans does indeed have content between 24khz and 48khz.
 
Re: New presonus mixer

Robert Lofgren said:
1 bit sampling with oversampling must sound awefull :D~:-D~:grin:
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Actually one of the earliest digital studio delay lines (delta-lab) used one bit digital encoding (delta-modulation). A simple above/below comparator drives a 1 pole integrator up/down until the comparator changes the direction of the fixed slope. If the clock rate, which is the same thing as the data rate, is high enough these could sound very good (say 1 meg bit per second). They were notorious for sounding very bad when pushed to too low clock rate in an attempt to get longer delays (digital memory was very expensive in the old days). A distortion called slope overload occurred in the margin but this is too much information about an obsolete technology.

JR
 
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