Panning and Matrixing deleted

[Guest]

[quote author="olafmatt"][quote author="Wavebourn"]I had an experimental console with "far-near" sliders that regulated balance between direct and reverberated signals, it was very convenient to play with dynamically. But it was not enough, since reverberator was a tape loop type mono reverberator, so I dreamed of the way to shift recording heads for each channel when controlling panoramic pot... Thecnicaly it would be possible, but too expensive!

Now I believe it is possible when DSP is so cheap and well developed, but unfortunately I have no experience with it.[/quote]

I once wrote a software that extended an existing amplitude based panning system (3 dimensional over arbitrarily positioned speakers) by adding a 'distance' control. In the end it was just adjusting the balance between direct and reverberated signal (for each speaker with an extra delay for each speaker) depending in the 'distance' you dialed in. Combined with the XYZ movements the basic system supports (see http://www.acoustics.hut.fi/research/cat/vbap/ for the original system) this was quite impressive.
The first version didn't do any DSP and instead was just using MIDI signals to adjust signal and reverb levels in a Yamaha digital mixer.
Maybe this could be coded completely in analog, but would need at least VCAs as far as I can see.

Olaf[/quote]

Olaf;

did you write some reverberating stuff?

 

[Jim Zuehsow]

[quote author="mediatechnology"]There used to be an FM broadcast stereo cart machine format that was matrixed L+R L-R onto the left and right channels. The azmith error, significant in broadcast carts, created separation errors not phase errors when matrixed to tape and decoded on playback. It had good mono compatability.[/quote]

Yep, that was the one from Pacific Engineering. I bought one off ebay for 5 bucks a year ago just for the electronics. They also made a mulitband limiter, and a unit similar to the UREI BL-40. They didn't skimp on the design. All high quality parts inside.
Speaking of pan pots, I remember the Steve Dove console articles in Studio Sound magazine back in the 80's and he was pretty specific that using a single panpot instead of a dual unit was the way to go due to tracking between the two pots. He also said it should be wire wound, to avoid crosstalk. His hurorous writing style convinced me he knew what he was talking about.

 

[olafmatt]

[quote author="Wavebourn"]did you write some reverberating stuff?[/quote]

Sort of, just adopted a public domain Schroeder/Moorer reverb implementation for my purpose.

Olaf

 

[NewYorkDave]

Jim,

You must be thinking of someone else. I have Dove's Studio Sound console design series and I'm pretty sure he doesn't say that.

(Of course, I have to go check now, just to make sure I haven't made a fool of myself. I'll come back later and edit, if I was wrong).

Do you guys have Richard Cabot's and Bob Orban's AES papers on panpots?

Here's one more thing to add to the mix:

http://www.dself.dsl.pipex.com/ampins/mixer/mixerdes.htm#5

Also possibly of interest:

http://www.audiosignal.co.uk/Gerzon%20archive.html

 

[mikep]

Ive always liked the idea of a stepped pan but Ive never seen it implemented in a console. an off-the-shelf, single deck, 11-position rotary switch would do it for me. 23-position if you want to get crazy. you can can have nearly perfect channel seperation at full pan, nearly perfect center position, repeatability, and reliability with a simple passive circuit. Im all for minimizing rotary switches (for cost reasons), but if I were building myself a console this is the one place I would use one in the channel module.

mike

 

[Guest]

[quote author="olafmatt"][quote author="Wavebourn"]did you write some reverberating stuff?[/quote]

Sort of, just adopted a public domain Schroeder/Moorer reverb implementation for my purpose.

Olaf[/quote]

Will your DSP be fast enough for live concerts when say 8 channels are used?

It is very interesting idea.

 

[NewYorkDave]

Resistor string for a stepped panpot--seen here as part of an Ambisonic decoder:

PDF

Soundcraft panpot with NIC, as outlined in D. Self article:

I also found this tantalizing nugget on Usenet:

[quote author="Graham Hinton"]
The figure of -4.5dB was established empirically by the BBC Research Dept
and it is a requirement of all consoles sold to them. They are more
particular about that than the fader laws.
...
The BBC pan curve is quite exacting, it is not just the centre gain , but the whole curve.
...
This is all discussed in a BBC Research paper that I unfortunately no
longer have a copy of. A lot of their reports are online now so I'll check
if that one is available.
[/quote]

I spent a little time with Google and couldn't turn up this paper. I'm not saying it's not out there--I didn't dig really deep due to limited free time.

 

[olafmatt]

[quote author="Wavebourn"]Will your DSP be fast enough for live concerts when say 8 channels are used?

It is very interesting idea.[/quote]

Running 8 ore more reverbs in parallel (on a 2GHz Pentium) was no problem performance wise. But I have to say that the reverbs were just made using 4 allpass and 8 comb filters and there was no simulation of early reflections.

I have a paper online that describes how the system works and how the levels get calculated and change with distance:

http://www.akustische-kunst.org/maxmsp/download/rvbap.pdf

Olaf

 

[NewYorkDave]

http://en.wikipedia.org/wiki/Negative_impedance_converter

 

[SSLtech]

[quote author="mediatechnology"]Does the Soundcraft have a nice law?[/quote]
Yes.

"Use a Mackie, Go to Jail".

:green:

:wink:

Keef

 

[Guest]

[quote author="olafmatt"][quote author="Wavebourn"]Will your DSP be fast enough for live concerts when say 8 channels are used?

It is very interesting idea.[/quote]

Running 8 ore more reverbs in parallel (on a 2GHz Pentium) was no problem performance wise. But I have to say that the reverbs were just made using 4 allpass and 8 comb filters and there was no simulation of early reflections.

I have a paper online that describes how the system works and how the levels get calculated and change with distance:

http://www.akustische-kunst.org/maxmsp/download/rvbap.pdf

Olaf[/quote]

I mean one reverb and 8 pan controls. Or do you mean 8 reverbs will be needed? What will be delay between input analog signals and digitally summed analog output?
I mean live concerts, not a studio recording!

 

[olafmatt]

[quote author="Wavebourn"]I mean one reverb and 8 pan controls. Or do you mean 8 reverbs will be needed? What will be delay between input analog signals and digitally summed analog output?
I mean live concerts, not a studio recording![/quote]

The system as it is right now uses one reverb per speaker. So with 8 speakers you end up using 8 reverbs. That way you can pan lets say 20 sound sources to 20 different positions without needing 20 reverbs. The algorithm could be changed to work the other way round, using one reverb per sound source, but in my case the first approach meant using less reverbs.
As for delay, the software has a lateny of 64 samples. There might be a few more being added by the audio interface.

Olaf