whats ur view?
native or creamware mixing?
Sir -
King of Snake
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Personally I mix both in logic and in SFP, I tend to use sfp for the 'bussing' as well as adding summing additional outboard & pulsar synths...
However whatever your working style I would suggest that u remember to check for phase problems in those long SFP signal chains:
http://www.planetz.com/forums/viewtopic ... orum=16&11
However whatever your working style I would suggest that u remember to check for phase problems in those long SFP signal chains:
http://www.planetz.com/forums/viewtopic ... orum=16&11
SFP all the way. I used to use native sequencers as a tape deck when discovering SFP mixing. I recently bought the VDAT and the GraphEQ so now I do everything in SFP. It is so much smoother and better sounding to my ears. I also got my creative mood back. Now I could benefit from aquiring some more DSPs .
.I don't completely understand the phase issue - can someone explain in more detail?On 2003-09-09 12:01, valis wrote:
Personally I mix both in logic and in SFP, I tend to use sfp for the 'bussing' as well as adding summing additional outboard & pulsar synths...
However whatever your working style I would suggest that u remember to check for phase problems in those long SFP signal chains:
http://www.planetz.com/forums/viewtopic ... orum=16&11
Well this is my understanding...
With creamware's dsp code devices are allowed to load their code on more than 1 dsp chip (spread it out) so that the dsp resources are used as efficiently as possible. For example, say you load a large device and it uses 75% of dsp1. Now if you were to load a second device that uses say 50% of a dsp chip's resources it will load 25% onto dsp chip1 and 25% onto dsp chip2 (rather than leaving that leftover 25% of dsp1 unused).
However one drawback to this is that if a stereo device winds up splitting the channels between 2 dsps chips you can get a 3 to 7 sample delay (phase difference) between the 2 channels.
With Red Muze's device I've found the easiest way to do this is to send what I *know* is a mono signal down the stereo path in SFP and insert the device after each large plugin. Use the inv. L or R modes to see if they truly cancel (a mono signal with channels inverted should be silent-completely cancelled). It's a cinch to adjust this way too since all you have to do is touch one of the L or R dials and hit your arrow key on the keyboard until it goes silent. Easy to tell if you're adjusting the wrong direction too.
In theory you have the potential for as many phase errors in your mix as you have dsp chips, but in practice i've found that I tends to only have 3 or 4. Until RedMuze pointed all of this out to me I had no idea that any of this was going on (what ME check for mono compatibility?)
Anyway enough Alfred E. Newman impressions and good luck
With creamware's dsp code devices are allowed to load their code on more than 1 dsp chip (spread it out) so that the dsp resources are used as efficiently as possible. For example, say you load a large device and it uses 75% of dsp1. Now if you were to load a second device that uses say 50% of a dsp chip's resources it will load 25% onto dsp chip1 and 25% onto dsp chip2 (rather than leaving that leftover 25% of dsp1 unused).
However one drawback to this is that if a stereo device winds up splitting the channels between 2 dsps chips you can get a 3 to 7 sample delay (phase difference) between the 2 channels.
With Red Muze's device I've found the easiest way to do this is to send what I *know* is a mono signal down the stereo path in SFP and insert the device after each large plugin. Use the inv. L or R modes to see if they truly cancel (a mono signal with channels inverted should be silent-completely cancelled). It's a cinch to adjust this way too since all you have to do is touch one of the L or R dials and hit your arrow key on the keyboard until it goes silent. Easy to tell if you're adjusting the wrong direction too.
In theory you have the potential for as many phase errors in your mix as you have dsp chips, but in practice i've found that I tends to only have 3 or 4. Until RedMuze pointed all of this out to me I had no idea that any of this was going on (what ME check for mono compatibility?)
Anyway enough Alfred E. Newman impressions and good luck
phase issues are probably the most overlooked source for any signal degradation - be it mix or processing.
Phase fidelity in the analog world makes the difference between high end and just good gear - it's not a digital phenomenon at all.
The effect is frequency = time depended and may of course influence any signal processing.
Often the sound gets a dull character - it's just not right without an obvious reason - but it can also get quite punchy if the signals match in a certain way.
I've noticed it impressively on 2 occasions.
Once when accidently monitoring a signal on the mixer and Pulsar. The adat connection makes a 0.5 ms delay and when I moved the fader up (with was a pad sound) there was a nice comb filter effect
I really didn't get it. There was no filter in the chain but it clearly sounded as if.
Another time when listening to a classic concerto with cheap speakers, when the solo violin suddenly panned from left to right. They really don't do these tricks at Sony...
My personal guess is that phase fidelity is also an important aspect of the various audio engines in sequencer applications, not only bit depth and samplerate.
But no need to get paranoid: any audio gear sufferes more or less from this.
cheers, Tom
Phase fidelity in the analog world makes the difference between high end and just good gear - it's not a digital phenomenon at all.
The effect is frequency = time depended and may of course influence any signal processing.
Often the sound gets a dull character - it's just not right without an obvious reason - but it can also get quite punchy if the signals match in a certain way.
I've noticed it impressively on 2 occasions.
Once when accidently monitoring a signal on the mixer and Pulsar. The adat connection makes a 0.5 ms delay and when I moved the fader up (with was a pad sound) there was a nice comb filter effect
I really didn't get it. There was no filter in the chain but it clearly sounded as if.
Another time when listening to a classic concerto with cheap speakers, when the solo violin suddenly panned from left to right. They really don't do these tricks at Sony...
My personal guess is that phase fidelity is also an important aspect of the various audio engines in sequencer applications, not only bit depth and samplerate.
But no need to get paranoid: any audio gear sufferes more or less from this.
cheers, Tom
Now put the phase knowledge on a multi-band compressor which is not phase coherent. The different bands are split by filters, which often cause phase problems (see astroman's experience). Now not only L/R has phase differences, but also different frequencies on the same channel have it and these cannot be removed by using a simple tool as the PhaseFix...
For example Waves plugins have phase linear EQ's and multi-band compression -have a look at their shop prices
For example Waves plugins have phase linear EQ's and multi-band compression -have a look at their shop prices
You didn't specify what your native audio program was but in my case, I use tripleDAT.
Generally speaking, I always use a Pulsar mixer except in cases where my arrangements exceed 16 channels of audio (happens often). In that case, I'll do a submix in tripleDAT. I don't really notice a difference in sound quality between mixing in 3DAT and mixing in Pulsar.
It sure would be nice if CW could expand the I/O to 24 channels. It would be great to be able to mix 24 separate channels at once as opposed to 16.
Generally speaking, I always use a Pulsar mixer except in cases where my arrangements exceed 16 channels of audio (happens often). In that case, I'll do a submix in tripleDAT. I don't really notice a difference in sound quality between mixing in 3DAT and mixing in Pulsar.
It sure would be nice if CW could expand the I/O to 24 channels. It would be great to be able to mix 24 separate channels at once as opposed to 16.
Actually the channels themselves will remain phase-coherent in that the L&R channels will be allocated to the same dsp IF "PhaseComp" is enabled (and perhaps the whole mixer?).
Inserts on mixers, however, are not guaranteed phase-coherent even with this setting. I just verified it with an insert on a Dynamic mixer. As a test I loaded 6 voices on the ProOne and 2 on the Minimax to load my dsps up to 90% before loading an effect as an insert and had a phase error of 9 samples in the compressor as a result.
Inserts on mixers, however, are not guaranteed phase-coherent even with this setting. I just verified it with an insert on a Dynamic mixer. As a test I loaded 6 voices on the ProOne and 2 on the Minimax to load my dsps up to 90% before loading an effect as an insert and had a phase error of 9 samples in the compressor as a result.
I'm most impressed that Final Liza, Optimaster and the Sonic Timeworks Mastering Compressors and Vintage EQ's don't cause phase problems...
As for the test, I loaded 90% dsp and then added DynamicMixer and inserted the effects. A Wave Source's single output routed to both left & right inputs of a mixer channel and PhaseFix behind the mixer, routed to my monitors.
As for the test, I loaded 90% dsp and then added DynamicMixer and inserted the effects. A Wave Source's single output routed to both left & right inputs of a mixer channel and PhaseFix behind the mixer, routed to my monitors.