Hi Eddie,

I’ve always enjoyed your writing! I’m not really that much of a tech, but I have a lot of old stuff (I mean vintage), and your MIX column gives me faith that I will be able to find someone who can repair my prized gear.

Today I wanted to repair a Pro Jr. input jack, and I am certain that once upon a time you had an article in Mix detailing your repair of one of these. I can’t find it on the web. I hope this isn’t too annoying, but, do you have that article stored anywhere?

Sorry to bug you, and thanks!

David
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D!

I have a Pro Junior that I’ve done some mods to, but I don’t recall anything about the jack. That said, this link below should help! Antique Electronic Supply is very tube amp friendly.

http://www.tubesandmore.com/scripts/foxweb.dll/EXTPAGE@d:/dfs/elevclients/cemirror/ELEVATOR.FXP?PAGE=SUBCAT&SEARCH_TREE01=JACKS/PLUGS

I was off on vacation for a week so there’s some catching up to do. For starters, I am going to post some links and, over time, come back and explain in more depth how these links can help you…

OBSOLETE PARTS?
As mentioned, upgrading opamps in the audio path can a bit of a mine field. But not ALL opamps are in the signal path – the side-chain of vintage dbx compressor-limiters, for example, have opamps that do all sorts of fun stuff. So, when an opamp’s functionality is circuit-critical, check the manufacturer’s website first, because sometimes you can get lucky and find a compatible replacement.

As the link shows, the LM308 has been DISCONTINUED, but to the right is a link to the LM308 DATASHEET as well as it’s replacement – in the ALSO RECOMMENDED box – the LM8261. You’ll want to download the datasheet for both parts…
POWER SURPRISE
Power Supplies are often the weak link in many products, not necessarily by circuit design so much as the lack of consideration of how heat shortens life over long periods of time. In general, electronics components should not be so hot as to burn, but ‘too hot to fail’ happens all too often, especially when you consider how often rack gear is mounted with no space in between. Over time, parts just burn themselves out.
RECTIFIERS Convert AC to DC and there are many types, from Half Wave to Full Wave, Single Voltage to Bipolar as well as Voltage Doubler.

Can You NAME THAT RECTIFIER?

Once AC is converted to DC it needs to be regulated – it must tolerate variations at the power outlet – it can’t sag when the Air Conditioner or Heater comes on and must be there if the device being powered demands more juice.
SIMPLE REGULATION
Theses days, pretty much every solution comes as a single Integrated Circuit or IC, with a minimal amount of support parts. But if you work on vintage Gear, you’ll see many Variations on the Regulation Theme.

If you ever wanted to know more about regulated power supplies, National Semiconductor published this useful FUNDAMENTALS pdf.

CAPACITOR CONFUSION
Back in the vacuum tube daze, capacitors were big enough to tell you alot about themselves. Then, large and small value caps relied on color code. Now reading and translating much small conventional (axial or radial) capacitors can be confusing, because limited space forces manufacturers to abbreviate. It’s even crazier with surface mount (SMT) parts. I’ll be posting a chart soon, but in the meantime, here’s a cool link that explains alot about what some of the abbreviations mean. Note that some of these abbreviations might be on the schematic as well.

more to come…

Q: Why don’t PC mics don’t work on a standard console?

This remarkably common question most recently came from Andreas Polydoros from Athens, Greece.

A: Because there is a big difference between a conventional phantom-powered studio mic and the Electret microphones typically found in computers, cell phones, cameras and portable recording devices. The Electret condenser mic does require power (for its built-in amplifier) but in a totally different way from the Phantom-powered studio condenser mics that have been part of our audio-centric world since the sixties.

My first ‘phantom’ article originally appeared in the July 1999 issue of EQ.

SIGNAL DISTRIBUTION
In mic evolution, moving coil and ribbon mics deliver a balanced / differential signal by design and required no power. Initially, studio condenser microphones required power to polarize the capsule as well as the internal vacuum tube. When transistors were good enough to be low-noise amplifiers, condenser mics could then be powered by batteries either externally (like the sony C38) or internally (like the earliest version of the Neumann U-87).

VIVA LA DIFFERENCE!
In the pre-phantom audio world, mic- and line-level SIGNALS travel on a twisted pair of wires in a balanced, differential form. This can be seen on the left side of the image as a pair of AC (alternating current) sine waves of opposite polarity. In contrast, a Common Mode signal appears as a pair of identical polarity (in-phase) sine waves (in RED).

This type of SIGNAL distribution offers the best noise immunity because balanced / differential audio INPUTS recognize the desired differential waves as SIGNAL and the pair of in-phase waves as ‘common-mode’ NOISE. The ratio of signal-to-noise is called the Common Mode Rejection Ratio or CMRR. Surprisingly, Phantom power is injected as a common mode DC signal, the beauty of which is that no modification to the existing signal distribution system was required.

ELECTRET is the electrostatic version of a magnet – holding an electrical rather than magnetic charge. When the Electret material is combined with a diaphragm, the result is an ELECTRET MICROPHONE, which, unlike a conventional studio condenser microphone, does not require a polarizing voltage for the diaphragm, but does require power for its built-in FET amplifier.

From the schematic image, there are two connections to the Electret Capsule Assembly, which contains an internal Field Effect Transistor (FET) Amplifier. External to the capsule is the FET’s ‘load’ resistor – where the audio signal will appear – the opposite side of the resistor being connected to the power source. So, that’s two of three connections, the third being ground. The signal output is unbalanced.

To visualize what is expected at the typical 1/8-inch (3.5mm) TRS Electret microphone connector, we must first consider whether the source is a mono or stereo mic. For a mono computer mic input, Creative Labs publishes the follow spec for their sound blaster card:

Input Type: Unbalanced Low Impedance
Input Sensitivity: Approx. -20dBV (100mV or 0.1Volt)
Input Impedance: 600 to 1500. (Ohms)
Input Connector: 3.5mm Miniplug (Stereo Jack)
Input Wiring: Audio on Tip, Ground on Sleeve, 5Volts DC Bias on Ring

The spec above implies that the load resistor is most likely located at the microphone. But for cameras that have a stereo mic input, also on a TRS connector, a pair of resistors will be located near the TRS input connector where the left capsule’s output will be TRS-Tip, the right capsule’s output will be TRS-Ring and ground is on TRS sleeve.

RECENT UPDATE
Conversations with an Australian reader helped me zoom in on the problem: audio / music store salespeople are selling Electret mics – which we know require power – as being phantom-powered! Dang! You can just imagine what happens to microphone electronics expecting 1.5 to 5.6-volts when 48-volts shows up to the party!

Well, here’s a solution I’d love to have a box of and hopefully will review in the near future…

AMBIENT Electret Mic Adapters

On Day-2 of my new blog David Prentice made this comment about LEVELS and this is coincidentally followed by a reader question…

Hi Eddie!

Setting up a ‘direct line’ for questions sounds like a wonderful idea!

These days engineers are often application wise and audio signal path ignorant. Old dog concepts like gain-staging, level-matching and finding the control pot’s sweet-spot were essential to negotiating the limited dynamic range between noise-floor and distortion. Young engineers need “mentors,” more experienced engineers that can explain how things work and show how a little Knowledge can make things sound better.

Good luck on the new enterprise. (DP)

What a great setup for today’s question (ec).

Hi Eddie,

I have a decent home studio setup and would like to get the most from it. I have a Tascam DM-4800 mixer that interfaces to PC via an RME 9652. (The self-rolled PC is running Cubase 4.5.2 on XP64, with plans to upgrade to cubase 6 and win7.)

The RME has 24 channels (3 ports) of lightpipe but the DM-4800 only comes with 8 channels (1 port) of adat stock. You can get more but you have to buy adat expansion cards for it. To overcome this limitation I use another audio interface, the EMU 1820m, as a converter to get 8 more channels out of the RME/Cubase rig to the DM so that I am able to mix 16 channels from the DAW – basically 8 stems. Everything is slaved to the RME clock. I have a UA LA-610 mkII for vocals and guitars, a UAD-2 and various other plug-ins as well.

Q-1: how do I need to set gain structure based upon my signal chain? I’m starting to have some success with placements on the MTV networks for shows like “Real World/Road Rules,” “Married to Rock”, and several others and would like to make my music as professional sounding as possible.

Q-2: How hot would a seasoned engineer record the signal of a sound source using my setup? Should I be recording a synth part at -0.3 dbfs or more in the -12 to -16 dbfs range so that I have more room to use eq, compression and fx?

Q-3: How does recording at full scale affect the mixing process? What I notice, and what concerns me, is that if I have a synth going and then add rhythm guitar the mix bus clip indicators immediately go red on me. Is this the product of recording too hot? Thanks for the help and super-fast reply.

Ellis Lofton

Prophet Speaks Music
www.prophetspeaks.net

A-1: The most important detail is that 0dBFS = full scale = max, it’s the maximum recordable level AND the level at which the analog converters clip. To test both the AD and the DA converters (both in the Tascam DM-4800), plug a test tone into a line input and confirm that the Tascam’s metering is in agreement with Cubase’s metering. (The RME 9652 is ‘just’ the digital interface between the Tascam and Cubase and can not be calibrated.) Record this tone at 0dBFS, then -6dBFS, the -12dBFS and -18dBFS. Then play back and see what happens on the other side. You can copy the recorded track to simulate a session and confirm signal flow and headroom.

A-2: One ‘concern’ is that analog gear can have a max output that exceeds the Tascam’s input capabilities. To test / confirm, you’d want to inject a test tone into the analog gear, get the level to zero VU (0VU) on the analog meter and then see what level that shows up as on the DM’s meters. I’d expect that level to be anywhere from -16dbFS to -20dBFS. This is called NOMINAL level, which on a VU meter / Analog gear is +4dBu.

A-3: Analog gear has a minimum of 14dB to 18dB of headroom above Nominal and often more. The last thing you want to do is overdrive the converters, but IF you wanted to drive the analog gear a little harder, you’d need to insert a pad between the analog gear and the front end of the DM-4800.

I think there are rules being developed for mixing TV commercial audio, and as I understand it, TV programs and Films use -20dBFS as their dialogue reference level – this reserves headroom for the ‘suprise’ of sound effects.

A-4: It sounds like you are recording very hot, ‘across the board’ considering that you say it’s so easy to overload the mix buss. Recording level is program dependent, so you might try recording drums at -6dBFS max and other instruments an additional 6dB lower at -12dBFS.

A-5: If you need more headroom, try looking for a level trim option. EQ plugs on Protools and Adobe Audition have an additional GAIN control that allows overall LEVEL to be reduced to compensate for drastic EQ boosting. Surely Cubase EQ plugs have such an option. PT also has a Level TRIM plug, which, if you have one, can be placed BEFORE the EQ plug. The LEVEL TRIM PLUG can also be used as a “Line Attenuator, or PAD,’ reducing the overall level as a way to keep the faders up in a range where they have more ‘physical’ resolution.

A-6: When mixing, you should avoid overloading the mix buss at all costs. Try this: allow at least 6dB of headroom for the drums – peaks that are no higher than -6dBFS – and 12dB of headroom for instruments and vocals. I submix drums, bass and drum verb / ambience in both the digital and the analog domain. Doing that with other groups – instruments, vocals, etc – allows more overall level control. Please know that pulling the master fader down will not solve an overloaded mix buss.

Let me know if this helps and feel free to provide screen shots.

PS: I wrote this article on AUDIO LEVELS many years ago – It’s due for a rewrite (hopefully soon).

FINALE
Ellis and I wrote back and forth a few times to clarify details. That dialogue was integrated into this correspondence. Ellis’s first response was “Wow,” and ‘thanks for the quick and awesome response!’

For years, I have been helping people solve problems by remote control. When it’s stuff I don’t know intimately, my clients and I learn together. From there it seems a natural progression to bring this private interaction into the light of a public forum – a.k.a. this new blog, Ask Eddie – the essence of which will be distilled into my new monthly column of the same name.

Now it’s your turn to ask a question…

“What do you wanna know more about?” You provide links, schematics and images in advance and I’ll do my best to fill in the gaps by hunting, gathering, annotating and hopefully answering your question. When depth is required, a link to Geek-speak articles will be provided. Deal?

You can comment below the strawberry pic OR email edaudio@tangible-technology.com