Mic Level Directly to AD converter

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yodermr

Well-known member
Joined
Sep 7, 2005
Messages
53
Location
Illinois
Hi, I'm fairly green to DIY except for forever rewiring my guitar and making my own cables to keep things cheap.

I was pondering as I have been reading about building mic preamps and such, if recording equipment will evolve so that there is no need for mic preamps, where the AD converters accept mic level signals and microphones more commonly have mic level eqs, variable pads or just digital outputs.

Could this happen or is there something special about line level? I always assumed that you need the higher level as a standard interface between equipment and an extra umph to handle long cable runs.

Or maybe I'm alot greener than I imagine?
 
Mic signal levels are extremely unpredictable, ranging from below 1mV to a full volt or more, depending on mic type, sound source, and setup.

The mic preamp accepts these levels and adapts them into the "predictable" line level, that is needed to do sensible A/D conversion.

Jakob E.
 
I guess I thought that it was a linear relation between the range of voltages at the mic line level to the line level.

So the components of a mic preamp actually put limits on the swing in voltages? Like a compressor?

Again, excuse my ignorance

:roll:
 
> is there something special about line level?

"Line level" implies high enough to stay away from the universal noise, low enough not to overload on reasonable size amps.

The other extremes are:

- Microphone level, which ranges from 0.1 microvolt to over 1 volt

- Loudspeakers, which often demand unreasonable size amplifiers.

Line level originally comes from telephone. It IS the level on a telephone line. Early telephone systems had no amplifiers, so line level is also earphone level. Telephones were not practical until they devised microphones that made a "line level" (or earphone level) output. Actually, the Carbon Mike "is" an amplifier: it takes battery power and modulates it with weak speech power to give enough electrical audio power to hear in an earphone.

So line level, in telephony, is a compromise between the maximum practical output of a carbon mike, and the minimum usable signal into an earphone.

If, instead of high-output microphones, we had developed high-gain earpieces, we would have had trouble with line noise. Miles of overhead wire will pick up lots of atmospheric static.

People accept that a telephone is pretty much unity-gain and imperfect: won't carry from a whisper to a shout cleanly.

The carbon mike is hot, but has high noise and distortion with limited bandwidth. All the "better" mikes have lower output.

While communicating speech has a narrow range of level, fancy speech and music has a wide range.

So for "good" radio and recording work, we need gain, and adjustable gain, right after the microphone.

We need to put the 100+dB range of performance through a 40dB-60dB recorder, wire-line, or radio link. We need to adjust gain between the mike and the "line" (most recorders and transmitters expect "line" input, fairly hot, levels already set near optimum).

Line level from a carbon mike is ~about~ 1 milliWatt average, with very wide range depending on the talker.

Maximum "line" level, while originally the level of a loud carbon mike, soon became the maximum level of one (or a few) "small" tubes. Tubes don't get any cheaper below about 1 Watt dissipation, which is good for about 100mW audio. For best fidelity on noisy lines, more than 1 mW average levels were used. US Telco standards limited level to avoid interfering with other customers in the same cable: +8dBm or 6mW on a VU-like meter. For most early work, rare clipping was not the biggest problem, so peaks were assumed to be -12dB above meter reading or +20dBm (100mW). As productions got more advanced, rare clipping became an issue, and most recording gear was calibrated to +4dBm nominal to give 16dB headroom within a +20dBm max system.

> I always assumed that you need the higher level as a standard interface between equipment

It sure can be convenient.

> and an extra umph to handle long cable runs.

Levels far above the millivolts of modern microphones are helpful in overwhelming interference. One Volt (or so) is fine. We could design for much higher line levels to reduce interference more, but above around 1mW nominal 100mW peak the cost of bigger output stages becomes a problem, and few lines need that much level.

Most lines are run with very small net loss. NYDave's mile-long line only loses a few dB. If you get into many-mile lines, you don't use a standard gear line output stage, you use a special Line Amp with +30dBm output (most can peak over +36dBm).


> where the AD converters accept mic level signals and microphones more commonly have mic level eqs, variable pads or just digital outputs.

First we designed mikes "hot" because we had no amplifiers. Then with amplifiers, we soon designed mikes as "weak" as possible, sacrificing output to get more bandwidth. So the soft-sound output of many mikes is right at the lowest level an analog system can handle cleanly. In general, digital systems can't work at such low levels (though those clever digital boys are making a liar out of me).

Also, with no analog gain control, and no restriction on mikes or performers, the range of level is 0.2 microVolts to 2 Volts. While ultimately we will deliver this in a 16-bit range of 2V to 32mV (output of a CD player), our job as recordists is to adjust the level for a nice playback. If we had to take 0.2 microVolts to 2 Volts direct at the A/D converter, we'd need 24 bits of A/D converter, with the bottom bits around a tenth of a microvolt. That may still be beyond the limit of audio A/Ds. Of course we could put a fixed gain, say 100X, between the mike and A/D, to keep the lowest levels up where a A/D can read them properly; but then the loudest mikes and performances come out at hundreds of volts which is far more than a practical A/D really wants to see.

So we still want analog gain with an analog gain control. Of course for $0.50 we can hide this inside the mike along with a $5 A/D converter. Then we need an upstream path so I can sit in my easy chair and adjust the gain of the mike in the studio or high in the balcony.

FWIW, one of the mega-mike makers has a "USB mike" intended for project studios that eats air-waves and puts-out USB digital. So the whole analog path is an inch long. USB obviously has an upstream path so you can set analog gain, other frills too if they want to do it (though once in digital, you could shape it in the CPU). While this is very cute for a new studio, it doesn't make sense for people with legacy analog gear all intended for line level analog in/out.
 
Sir, I mean professor, you are awesome. The history behind it makes it all make sense. These lessons are lost in the race to build something for today. Facinating

Where do I send my tuition check?
:thumb:
 
Yes, we are not going to see digital microphones in studios as long as we have legacy analog gear we love to interact with to create the sound we want.

As told the A/D converters are not yet there. What we need first is a good quality floating point A/D converter that has enough gain for the weakest signals and is able to adjust the range for louder signals. So it would always have the precision like the 24 bits (25 bits?) in case of 32-bit IEEE floats (6.02*24=144 dB SNR), and still have a huge dynamic range (8 bits in exponent, that's 6.02*2^8=1541 dB DNR). For such system we also have to redefine the audio I/O systems like connectors, pinouts, phatom powering scheme, and solve the master clock issues etc. kind of new problems introduced. Maybe we don't need wires at all. Microphone picks up a radio signal which includes the synchronization and time code, and send's the converted data using Bluetooth, WLAN, 3G etc. techniques.
~Mikko
 
[quote author="mhelin"]Yes, we are not going to see digital microphones in studios as long as we have legacy analog gear we love to interact with to create the sound we want.
[/quote]

That's disappointing - was hoping to be able to buy a direct-to-mp3-mic this xmas... :cry: :wink:
 
FWIW, an interesting coincidence (?) is that the voltage used in telephone systems is 48V, and that's also the recognised potential for phantom power.
In my youth I remember asking a Telephone Engineer how much current was available from the telephone exchange.... his answer was "melt a spanner lad!" I instantly acquired a great respect for telephone systems; this was before I learnt all about constant current sources!

Back on topic.... I was going to add a 2 cents worth; but I think PRR has said it all..... and seriously, I would like to add my thanks to PRR for a masterly piece, beautifully crafted.
 
[quote author="TedF"]FWIW, an interesting coincidence (?) is that the voltage used in telephone systems is 48V, and that's also the recognised potential for phantom power.[/quote]
Thanks for the information, next time I build a mic pre it will be telephone line powered...
 
[quote author="TedF"]FWIW, an interesting coincidence (?) is that the voltage used in telephone systems is 48V, and that's also the recognised potential for phantom power.[/quote]

My first ever job was doing engineering on early computerized telephone switches at Ericsson. Yes, these use 48V DC power, actually -48V (positive ground) for some reason I have long since forgotten. This was established way back when the automated switches used neither tubes nor transistors.

The switches used a massive battery system (the size of a decent living room) filled with led-acid batteries. These acted as backup in case of power loss.

We used a lot of -48V to +5V converters to drive the digital hardware we designed to replace those pesky old relays...
 
[quote author="OddHarmonic"]I'm not gonna be able to add much to the discussion of theory and potential advancements, but Neumann has exactly what you speak of on the market for a few years now. I vaguely remember a paper presentation at AES a few years back that I poked my head in on, perhaps worthy of an archive search...

http://www.neumann.com/infopool/mics/produkte.php?ProdID=solution-d[/quote]

Preview command is sending everything into limbo :mad: Recommend saving everything to word or something before you get too far.

Now

The mic is indeed the Solution D. Sounds pretty good on vocals judging by its first use in a Tierney Sutton session iirc. They do some clever interleaving of A/D's to get the requisite dynamic range.

I went to a presentation at LA AES by Juergen Wahl. I was a mite pissed (actually in both the British and American senses come to think of it). Juergen made some fairly nonsensical remarks about fundamental noise limits on conventional microphones and I challenged him, which provoked a petty reply. Might have had something to do with his formidable deutsche bosses being in the audience :razz:
 
It's always good to challenge! :grin:

I have read up on the 'Solution D' ; even when I knew nothing about it I thought it was a solution looking for a problem; after finding out about it I think the same..... The question has to be why?
Is going from analog to digital in the mic head better than doing it after a gain stabilising amplifier? Is there an advantage in stabilising gain in the digital domain? I think not.
Listening to a voice recording through a lovely tube mic, a few transformers and digits used purely as a storage medium... I'm quite happy. :wink:
 
They state the core problem well:

"the best delta-sigma A/D converters currently available as integrated circuits provide a dynamic range of 115-120 dB (A-weighted).... In contrast, a high-quality analog condenser microphone has a dynamic range of up to 130 dB."

Not endorsing the plan, just being argumentative:

"Information transmitted by the microphone includes the name of the manufacturer, the model, the serial number... During production, the sound engineer can continuously monitor all the microphones...."

"Continuously monitor... make, model, serial number...": I've been at gigs where mikes walked off the stage. And of course it is routine in some studios to log all mikes used: if it sounds good you might want to do it again, and if one of several same-model mikes is lightly screwing-up, such detailed logging helps repair. Logging the mikes is no longer a tedious manual chore.

I actually have wished for large adjustable gain in the mike, to overwhelm line noise. While I can often estimate my maximum usable gain pretty-close before the show, I can be wrong, and need to trim in sound check. Of course that opens the danger of overload, which may make an in-mike limiter a wise plan. All of that is perfectly possible in a custom analog interface, but too darn much trouble.

On-Air light, possible with analog, is dead-stupid with digital. Cueing the talent, of course. But also good when you have 75 mikes all over an orchestra, paying $1000/hour overhead, and three idiot helpers running around re-setting mikes. Instead of tracing a lost mike from board to snake to stage, the mixer blinks the On-Air button of the mike in question, and the idiot helper goes right to it.

Going digital actually eliminates one motivation: line noise should not be an issue. We can lay EtherNet right over a bass-amp's power Xformer without buzz. Of course I used to get away with laying low-Z UNbalanced dynamic mike lines over the small Traynor bass-amp in concert, but there are times when a noise-immune audio cable could be handy. And a back-link to the mike is fairly easy in digital systems, so we can set gain, pattern, limiter settings from our comfy-chair.

"developed by Neumann... significant cost savings."

Hard to read "Neumann" and "cost savings" in one sentence. But the buffer and digi-chip may now be as cheap as a hefty analog balanced output. That's a no-gain plan, but once you do that there are all these other "features" you can add for pennies. Just being able to save mike-logs in a click could be hundreds of dollars of gofer-pay per year. On-Air lights are traditionally an extra box and wire (and not really linked to the mike that is on). I'm not sure what the cable looks like, but clearly you could mux multiple mikes onto one cable, the way my whole building is networked off a few hubs. Not the same EtherNet Metcalfe collision traffic: audio streams are fixed bitrate so they can be scheduled. In fast busy Ethernet, 50% utilization is tolerated and latency can be theoretically infinite; scheduled audio streams can give <1 sample latency at 100% use of the wire.
 
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