SPL MixDream analog DAW summing

Linear means the change is the same regardless of signal level. Non-linear means the change depends in some way upon the signal level.

A good example of linear would be a simple volume change: if you turn down all your faders by 3dB you get exactly the same result as if you turn down the masters by 3dB and (ignoring the slight difference in headroom and noisefloor in the analog version) this holds true for analog and digital mixing. Likewise, an EQ can be thought of as a frequency specific volume control, so if you notch out 3dB at 3KHz on every single channel, you get the same results as if you notched out that frequency on the master instead.

A compressor is linear so long as the ratio is 1:1 (hence the straight diagonal line you see in the transfer graph of some plug-ins) but as soon as you wind the ratio up you introduce a non-linearity, and the results start to depend on the incoming signal level. Inserting a comp on every channel is obviously NOT the same as inserting a comp on the master, even with identical settings. The same applies to distortion; loud signals are affected much more than quiet signals, and because the artifacts are level-dependent we also get inter-modulation effects when distorting the whole mix that would not have been present when distorting individual channels. Again: equally valid for analog and digital systems.

Of course, an analog EQ will have a much smaller linear range than a digital EQ. Part of the craft of analog mixing is to set your gain staging to stay within the linear range of the equipment. Perhaps the artistic side comes in when you begin to deliberately exploit the non-linearities at the extremes; that's not something I can do with gain staging ITB, but I could load a saturation plug instead...

They are all running in parallel, so they won't accumulate as you imply: that would require you to patch the output of each channel into the input of the next so one signal passed through all channels in series. To return to the volume example: if you turn 48 individual channels down by 3dB you get the same result as turning the masters down by 3dB. 48 * 3 dB = way too much!

 

Are you saying that your internally summed mixes null perfectly with your externally summed mixes, but you can still hear a difference...?

Not sure what point you are making...?

No need for that. I am entitled to ask you to back up your claims even if I've never used a Neve or an API.

As it happens I think I do have experience with great analog mixers; they generally had the name Midas at the top.

 

quote: some equivalent, summed, across the mean?

i.e. its not 16x3dB exactly (or anywhere close), but each channel has a variance of between 0 and +/- 3 and the total variance is not exactly 3dB but some contributed value-per-channel.

just to clarify that one. thanks for the explanation on what you meant by linear, i was wondering about linear frequency responses from system to system, not that.

 

The point is that I believe the same mix engineer mixing the same tracks ITB (digital) and by external (analog) summing would end up with two different-sounding results. One mix would not necessarily better than the other, but there would be audible differences. The tenor of this thread is that there is therefore an implicit and unexplained difference between summing digitally and summing externally in analog.

If you could capture the exact settings used for the analog mix and reproduce them in the digital mix, the differences should disappear. So maybe we should be looking at why the engineer would arrive at different mixes and see if we can re-work the user interface that governs DAW mixes to give it more of the feel of an external mix.

I would agree with your noise figures if we were talking about white noise, but D-A converter noise has other content such as code-dependent conversion errors and components from the power supply. I don't think it is possible to state accurate figures for this effect, simply to note that the mix noise floor will rise in the order of 3dB per input channel.

By the way, no-one in this topic so far has mentioned one common reason for performing external analog summing: sample-rate conversion. It was how I got into using external summing. I had 96KHz tracks from which I needed 44.1KHz CDs, and the digital SRCs that I had available sounded horrible. It was a breath of fresh air to mix in analog tracks with a 40KHz bandwidth and then re-sample two-track at 44.1KHz. A lot of the clarity I am sure could be put down to the elimination of contorted phase effects in the 15-20KHz range, but this is one way I know to achieve the sound of a "direct-to-stereo" CD from recorded tracks.

 

So: at least 3dB per channel, but possibly more...

Of course that is perfectly valid. But I suspect a really good SRC like iZotope 64 bit or r8brain pro would do better: SRC Comparisons

 

After reading that thread, I've immediately taken their suggestion and set to work on writing an analog summing plug-in. I think it should be a killer.

 

I like the joke a lot, but in fact I think the challenge posed by analog summing may force the digital world to take a harder look at the human/computer interface issues that usually take a back seat. Plugins may be a joke, but a better digital control surface isn't.

Edit: This comment is more in reaction to the Womb thread where there seemed to be a consistent implication that different summing methods implied different mixes. People here have been more careful about separating the two issues.

 

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