Studio Cue System Project

[Brian Roth]

The cabling for that Furman system I mentioned was a NIGHTMARE; shielded "telco" 25 pair cables with the required .....damn I forget the 50 pin connector's Ma Bell number.....RJ-21?


Yep:

https://en.wikipedia.org/wiki/Registered_jack#RJ21
Bri

 

[Brian Roth]

In this day and age, I dunno what OPEN SOURCE digi protocol would be appropriate for this application. Obviously very low latency would be paramount for the A/D and then back out D/A in the conversion processes. The throughput in "the wire" is not an issue.

Bri

 

[drtechno]

What’s missing is that corporate $$. You and me both!!

Its the connections accumulated. But to let you know, I get these things cheaper than they sell them online too. I don't have much talent in the area so it hasn't justify me turning a place into a studio permanently. I'm thinking one day moving to a music city and go in with someone who has made the other parts of the mountain like the rooms and facilities and business side.

 

[Brian Roth]

Dante seems the buzz name DuJour, But, I find no consistent "numbers" for a latency round trip for AD and back again DA. And, does that number stay as a constant with higher "payloads" all rumbling down the wire?

Bri

PS Dante is not open source...

 

[Brian Roth]

Aviom seems to have set the bar for latency in Big Time systems....well under 1 mS,

Bri

 

[JimJhn]

Not to throw a wrench in the system but I have two obersvations;

I really hate the personal cue-mixers. The only people who really get some good use out of them, in my personal experience, are seasoned daily studio musicians.

I've seen the Myteks, and the Hears a ton on my dates, so I have allot of experience with them just as a producer (I'm not an engineer).

If you do a jazz date, or a rock-band date, walk around during a break and hear how completely screwed-up players have adjusted their boxes out of ignorance.

They are confused with them (just something added to make their day more difficult and get in the way of playing), and you get the craziest mixes when you pick up their phones.

Weird things like no piano, the click slamming, one instrument slamming and another inaudible, etc., etc. ,etc.

Not their fault. they just want to play, and you just want to put them in the most comfortable and inspirational space possible to make music without thinking about technology.

I do get the reason for certain personal mixes...the vocalist, the drummer mainly.

For that the hippest thing I've seen in a long time, outside of a scoring stage, was actually a very small semi-pro studio, and they were doing their own version of what was referenced earlier about the scoring stages (and the Hans Zimmer reference!).

They had bought a simple used (Mackie? Behringer? Presonus?) 16 channel analog mixer for maybe $300.00 or less.

They had it at the side of the main console, and it was basically a headphone monitor mixer for the studio cue system.

They set up really nice individual mixes for each band member, and then if someone asked for "more whoever", or "less whatever", they made the adjustment themselves in a couple of seconds.

All of this for a grand total of I'm guessing $350.00.

The studio guys do like the Myteks and such, but my experience there is also those guys never whine about their headphone mixes. They for the most part play great no matter what and don't ever complain (at least the good "A" players).

Not a reason not to try, but I really like the side-console method above as a preference, not a fallback.

Just my two cents.

 

[JimJhn]

Not the Mix Magazine article, but a press-release about the same thing that was the inspiration for, according the studio owner, the cheap side-console idea I referenced above;

https://www.postmagazine.com/Press-...tes-Newman-Scoring-Stage-with-Yamaha-con.aspx

 

[kags]

FWIW..this was State of the Art 20-ish years ago:

https://furmanpower.com/wp-content/uploads/2020/10/hrm-16_manual.pdf
It was part of this studio:

http://brianroth.com/projects/curtis/curtis2.html
Bri

We had this Furman boxes - very unwieldy cabling and not great Sonics, but useable and reliable. Hear Technologies amps are pretty crappy but the Avioms seem nice and good enough.

I spend a lot of time convincing studios with Ethernet seadohone systems to upgrade the Ethernet cables to ballistic cable that pays flat when run across the studio floor. Makes a huge difference in user experience and only costs a few cents more per foot.

 

[kags]

I agree that some players can screw up mixes. I always provide a nice stereo mix and then give additional feeds for each instrument for “more me” options. Also click and talkback. The Avioms also provide talkback (intercom) for each players so I don’t need a separate talkback mic for the synth or bass player.

 

[kags]

The cabling for that Furman system I mentioned was a NIGHTMARE; shielded "telco" 25 pair cables with the required .....damn I forget the 50 pin connector's Ma Bell number.....RJ-

I thought it was some kind of ugly SCSI connector, but telco makes more sense. It was a PIA

 

[Brian Roth]

I think some ancient SCSI used that, as well as old printers (was called a Centronics interface).

Bri

 

[kags]

Dante seems the buzz name DuJour, But, I find no consistent "numbers" for a latency round trip for AD and back again DA. And, does that number stay as a constant with higher "payloads" all rumbling down the wire?

Bri

PS Dante is not open source...

Dante gear is expensive.AVB stuff can be a bit more affordable. Latency for Dante is described as 0.1ms per switch hop. However, if you use a core audio driver from a computer, the latency is now dependent on your machine. When I use my i7 MacBook with a Dante interface and Dante Virtual Soundcard, the latency is not acceptable for tracking, but fine for mixing.

A Mac Pro with Pro Tools and a Dante card works extremely well, as do digital consoles with Dante onboard.

If you require a scalable system, the cost of Dante becomes insignificant. It is an incredible technology, but it needs to become more price-friendly.

 

[Voyager10]

So ... we still have a crosstalk problem to solve? As in - when you turn a channel 'off' you can still hear it at low level?

If it's due to capacitive coupling in the cables, it will presumably get worse at higher frequencies. The mitigations here might be to reduce the pot values and other impedances in the circuit, and/or add pre-emphasis and de-emphasis circuits. (Basically the signal going to the cue box has an HF boost applied, and which is then removed on the return signal). Final option would be to put some active circuitry in the cue box to drive the L/R outputs from low impedance. You'd need some phantom power, but it's plausible.

The other problem might be return currents in the ground wire, which will get worse with lower pot values and is largely frequency-independent. The easy answer here is an on/off switch (or pot with a switch) to disconnect the input to the pot when it's not in use.

 

[Potato Cakes]

So ... we still have a crosstalk problem to solve? As in - when you turn a channel 'off' you can still hear it at low level?

If it's due to capacitive coupling in the cables, it will presumably get worse at higher frequencies. The mitigations here might be to reduce the pot values and other impedances in the circuit, and/or add pre-emphasis and de-emphasis circuits. (Basically the signal going to the cue box has an HF boost applied, and which is then removed on the return signal). Final option would be to put some active circuitry in the cue box to drive the L/R outputs from low impedance. You'd need some phantom power, but it's plausible.

The other problem might be return currents in the ground wire, which will get worse with lower pot values and is largely frequency-independent. The easy answer here is an on/off switch (or pot with a switch) to disconnect the input to the pot when it's not in use.

I did try rigging up an on/off switch to the mixer as this seemed to make the most amount of sense but if I recall the audio bleed did not go away. I will have to double check to see if I implemented it incorrectly. The problem I believe is signal is getting into the ground which affects all of the pots used for volume and panning so breaking the signal to one potentiometer wouldn't mute the signal.

I have tried increasing the pot values with no change in signal bleed. I have not tried decreasing it but I have thought about it. I could use 1k pots and 1k bus resistors. I'm not sure how adding the HF boost then cut like an RIAA filter would work in this scenario.

I did not think of sending phantom power down the audio lines. That might be the ticket. I already have coupling caps at the output of the INA134s so I could inject +48V there.

Thanks!

Paul

 

[Potato Cakes]

Not to throw a wrench in the system but I have two obersvations;

I really hate the personal cue-mixers. The only people who really get some good use out of them, in my personal experience, are seasoned daily studio musicians.

I've seen the Myteks, and the Hears a ton on my dates, so I have allot of experience with them just as a producer (I'm not an engineer).

If you do a jazz date, or a rock-band date, walk around during a break and hear how completely screwed-up players have adjusted their boxes out of ignorance.

They are confused with them (just something added to make their day more difficult and get in the way of playing), and you get the craziest mixes when you pick up their phones.

Weird things like no piano, the click slamming, one instrument slamming and another inaudible, etc., etc. ,etc.

Not their fault. they just want to play, and you just want to put them in the most comfortable and inspirational space possible to make music without thinking about technology.

I do get the reason for certain personal mixes...the vocalist, the drummer mainly.

For that the hippest thing I've seen in a long time, outside of a scoring stage, was actually a very small semi-pro studio, and they were doing their own version of what was referenced earlier about the scoring stages (and the Hans Zimmer reference!).

They had bought a simple used (Mackie? Behringer? Presonus?) 16 channel analog mixer for maybe $300.00 or less.

They had it at the side of the main console, and it was basically a headphone monitor mixer for the studio cue system.

They set up really nice individual mixes for each band member, and then if someone asked for "more whoever", or "less whatever", they made the adjustment themselves in a couple of seconds.

All of this for a grand total of I'm guessing $350.00.

The studio guys do like the Myteks and such, but my experience there is also those guys never whine about their headphone mixes. They for the most part play great no matter what and don't ever complain (at least the good "A" players).

Not a reason not to try, but I really like the side-console method above as a preference, not a fallback.

Just my two cents.

I am using an Allen and Heath MixWizard 16M for this very purpose and dial up everyone's cue mix as it will do 6 stereo monitor mixes. It lives on top of the rack with the preamps and I can patch the outputs of the preamps I want to use for cue mixes to the 16M and sub mix accordingly. The personal cue mixer design I have is 6 channels. 4 are mono with pan and one stereo, which is enough to give them some flexibility without a whole mixer setup being in their way.

Anyway.... back to the task at hand.

Thanks!

Paul

 

[abbey road d enfer]

Not to throw a wrench in the system but I have two obersvations;

I really hate the personal cue-mixers. The only people who really get some good use out of them, in my personal experience, are seasoned daily studio musicians.

I've seen the Myteks, and the Hears a ton on my dates, so I have allot of experience with them just as a producer (I'm not an engineer).

If you do a jazz date, or a rock-band date, walk around during a break and hear how completely screwed-up players have adjusted their boxes out of ignorance.

They are confused with them (just something added to make their day more difficult and get in the way of playing), and you get the craziest mixes when you pick up their phones.

Weird things like no piano, the click slamming, one instrument slamming and another inaudible, etc., etc. ,etc.

Not their fault. they just want to play, and you just want to put them in the most comfortable and inspirational space possible to make music without thinking about technology.

I do get the reason for certain personal mixes...the vocalist, the drummer mainly.

Poor planning is the cause.
I always have a basic usable stereo feed that is used as a starting point. Knob-fearing musicians can control the volume, if nothing else.
Some musicians are cleverer, those that have a personal studio, and manage quite well the system (behringer P16).
Good labelling of the sources is paramount.
For technically-challenged ones, I do it for them, not too different than the old way when cue sends were managed from the aux sends or an auxiliary mixer. More convenient actually, because I can check the mix on headphones standing by the musician.

For that the hippest thing I've seen in a long time, outside of a scoring stage, was actually a very small semi-pro studio, and they were doing their own version of what was referenced earlier about the scoring stages (and the Hans Zimmer reference!).

They had bought a simple used (Mackie? Behringer? Presonus?) 16 channel analog mixer for maybe $300.00 or less.

They had it at the side of the main console, and it was basically a headphone monitor mixer for the studio cue system.

They set up really nice individual mixes for each band member, and then if someone asked for "more whoever", or "less whatever", they made the adjustment themselves in a couple of seconds.

All of this for a grand total of I'm guessing $350.00.

The studio guys do like the Myteks and such, but my experience there is also those guys never whine about their headphone mixes. They for the most part play great no matter what and don't ever complain (at least the good "A" players).

Not a reason not to try, but I really like the side-console method above as a preference, not a fallback.

Just my two cents.

An auxiiary mixer has the big advantage of zero-latency. I know many studios where the auxiliary mixer is in the control room. Not the most ergonomic arrangement IMO.
Useful only when the console has not enough Aux.

 

[drtechno]

Dante seems the buzz name DuJour, But, I find no consistent "numbers" for a latency round trip for AD and back again DA. And, does that number stay as a constant with higher "payloads" all rumbling down the wire?

Bri

PS Dante is not open source...

Round trip from what? They are both point to point supermac networking. But if you talking about analog converting-> over the wire , then converting back to analog-> then converting back int dante-> then back to analog, its about 0.5ms Difference is Dante packets are shorter and they use two mac ip interfaces to the network One for media+clock the other is checksum and clock, and midi control formatted to AES67. You can change a dante device into AES67 natively on the dante device. So we are looking at nanoseconds of differences. They make linear conversion switches between the two formats(Acom<->Dante). Dante channel count limitations is based on Ethernet clock. So your 250Mhz is limited to 256x256 500 Mhz is 512x512 1.2Ghz (fiber) 1024x1024 @ 48Khz and the divisor of channel count to sample frequency as the normal affair that happens in sereial bitstreams.

Dante is just a structed version of the same thing that use seperate ip/mac for control, checksum,and midi over the same wire, and can run a redundant network scheme.

So basically Acom is just another structed version of AES50 with control data on the same packet. Dante is ip structured Acom with one ip address being AE50 data and another ip address for control, checksum, and midi I/o in an AES 67 packet.

So I don't know why anyone would want to debate any of these formats over another other than features because timing differences are hundereds of picoseconds to a few nano seconds. Dante is a littlte bit more flexible because control is seperate from the bitstream and the dante control can run in a regular computing network if needed.

 

[drtechno]

Poor planning is the cause.
I always have a basic usable stereo feed that is used as a starting point. Knob-fearing musicians can control the volume, if nothing else.
Some musicians are cleverer, those that have a personal studio, and manage quite well the system (behringer P16).
Good labelling of the sources is paramount.
For technically-challenged ones, I do it for them, not too different than the old way when cue sends were managed from the aux sends or an auxiliary mixer. More convenient actually, because I can check the mix on headphones standing by the musician.

An auxiiary mixer has the big advantage of zero-latency. I know many studios where the auxiliary mixer is in the control room. Not the most ergonomic arrangement IMO.
Useful only when the console has not enough Aux.

When I was all analog, (including tape machine) I ran a Yamaha PM4000-48 for headphones and a Modded Soundcraft Ghost for tape record and return and used both for mixing. When computer mixing came around I shrunk everything and ran everything from hardware protools, Then the firewire came out, protools dumped their DSP mixer, and I went back to a PA board for headphone mixes and that is when the monitor button in the computer became useless. Dante came around and I was exposed to that from live recording but I haven't fully integrate it with the daw yet, because I've been debating on the PCIe card vs using a DAD multi-face on thunderbolt (which is external pcie) I know I can get the same performance I did with the Protools ver2 architecture on a Dante card (hardware protools1.5 ms, Dante PCIe is 1ms) I haven't looked to see what users are getting with the DAD box I've been considering.

Protools was a really good system at one time, then its slowly degraded after changing hands and moved into the computer environment and lost its only real advantage over the other DAWs. I'm just glad there is solutions now I can use different Daw programs including protools equally.

 

[Voyager10]

I did try rigging up an on/off switch to the mixer as this seemed to make the most amount of sense but if I recall the audio bleed did not go away. I will have to double check to see if I implemented it incorrectly. The problem I believe is signal is getting into the ground which affects all of the pots used for volume and panning so breaking the signal to one potentiometer wouldn't mute the signal.

I have tried increasing the pot values with no change in signal bleed. I have not tried decreasing it but I have thought about it. I could use 1k pots and 1k bus resistors. I'm not sure how adding the HF boost then cut like an RIAA filter would work in this scenario.

I did not think of sending phantom power down the audio lines. That might be the ticket. I already have coupling caps at the output of the INA134s so I could inject +48V there.

Thanks!

Paul

I think a bit of detective work might be useful before setting off on too much more construction.

Do you get the same amount of bleed with, say, a 100Hz tone and a 5Khz tone? If so the 'capacitive coupling' theory is less likely (and the HF boost/cut idea won't help).

Do you still get bleed with just one cue box plugged in? What about one cue box with the relevant level pot disconnected?

If you plug in 2,3,4 cue boxes, does the bleed get worse? If so, it could be a grounding problem in the 'mainframe' box.

Is there a measurable resistance betwen ground on the 'mainframe' and ground in the cue box (when connected via the CAT5 cable)? If so, does connecting a fat wire between the two make the bleed go away?

 

[Potato Cakes]

The bleed is definitely a high passed version of the signal, but more like 1kHz and not 5kHz.

I have only tested with one cue box at a time.

I will have to check the resistance between cue box ground and mainframe ground. I believe I got 0 Ohms when I first tested this but I do not remember now. Once the parts show up to rebuild the mainframe board I will be more thorough when testing. At the time of the initial build I was distracted by numerous simultaneous projects and goings on that wound up setting it to the side till now.

Thanks!

Paul