Very simple Audio compressor for FM transmitter

[ronsimpson]

I know better but.....I am going to cause trouble.
1) Audio Guru please look at the first link again. You have said many times that it is slow. Please explain. The attack and release time is adjustable and seems to be pretty fast. (if set to fast) I can build and compare to your circuit that seems to work in about 1mS. I do have yours working in LT spice with a MOSFET.
https://www.electro-tech-online.com...o-compressor-for-fm-transmitter.139249/page-3
2)I understand you want head room on your stereo. This is FM and there is no headroom. 100% modulation is all you are allowed. A peak limiter is very important to reduce distortion and to not transmit outside your band.
3)The first post has peak limiting and compression. It also has per-emphises for FM and all the other options do not because they were not built for broadcast FM.
4)Much of what Audio Guru is complaining about is two fold:
4a) Slow attack time.
4b) Driving too hard into the limiter. If the station only had 3db or 6db of compression things would be fine. They are probably running 20db of compression, and slow attack time.
The peak limiting is to keep the transmitter below 100%. The compression is to make the music louder and be above the noise level. Every radio station (and TV) that I know of have a peak limiter. I cant think of one with out a little compression. Many HAN transmitters have (even home made) have limiters.
Also I can't think of using one of these compressors with out a meter to show how hard the limiter is working. A radio station where you can hear the effects of the limiter/compressor does not have a good engineer. The sails department and the boss will say "turn it up". They want louder! I know I have been there many times. All these limiter/compressors shown so far are sheep. You can't hit them hard or you will hear bad effects. On the other hand, not having a limiter/compressor causes bad effects.

There is a balancing act between too little and too much audio processing. You and I would not allow the audio to be driven hard into a limiter/compressor. We probably agree that 100% modulation is all there is. Above 100% modulation things go very wrong.

The first circuit is the only one built for FM. What can we do to evaluate it, change it, make it or come up with a better circuit that is good for FM? (not stereo)

 

[audioguru]

The first link is to the PiRa compressor/limiter that is broadcast quality but it does not spec its distortion. They used to sell "illegal power" FM transmitters but now they are only in their archives.

Willen built my low power toy FM transmitter but says its max output is a little less than radio stations and peaks cause distortion so he wants a compressor/limiter for it. He doesn't need broadcast quality so I linked to one with very good spec's and it uses common parts.

 

[Willen]

MrAl- I am not more experienced to do as you said. So can you post the circuit by editing it? (by remving 2252 RMS detector chip)? Will it act as a good limiter?

Ronsimpson and audioguru-

If I made any limiter, simply can't I add 51k+100nf in series for Asian pre-emphasis (around 50uS)?

May be canadaelk has no THAT2252 RMS detector chips, so have to build just using THAT 2180 VCA chip. Datasheet has a limiter circuit but it uses 2252 chip too, so I am facing trouble to make a limiter using just only 2180 chip. May be MrAl has solution.

 

[ronsimpson]

If I made any limiter, simply can't I add 51k+100nf in series for Asian pre-emphasis (around 50uS)?

Yes you need to add the pre-emphasis before the limiter and them add de-emphasis after. Then the transmitter add it back in and the receiver takes it out.

There is pre-emphasis in the transmitter so........another way:
>Add pre to the limiter and remove it from the transmitter.

I don't know how to help you. No budget, hard to get parts, long distance,................
I know your situation. When I was a child in Canada, parts were one month away. My budget was small.

 

[Willen]

Yes you need to add the pre-emphasis before the limiter and them add de-emphasis after.

another hard calculation then. Cant I remove pre-emphasis from FM tx and use before limiter for simple?

I don't know how to help you. No budget, hard to get parts, long distance,................
I know your situation. When I was a child in Canada, parts were one month away. My budget was small.

Yeah, feeling like to cry like a baby... Components in USA are extremly expensive for Asian people. May be $50 is not mor for Americans but here a normal man can earn Rs.5,000 within a month which is equal to $50!

I don't know how to help you.

Now I need a good limiter circuit made by few opamps and 2180 without 2252 but good performance.

 

[audioguru]

If I made any limiter, simply can't I add 51k+100nf in series for Asian pre-emphasis (around 50uS)?

I don't know which limiter circuit you are talking about. The PiRa, Harry's and my Mod4 circuits use a parallel capacitor and resistor.

51k and 100nF make 5100us, not 50us. For 50us then you need 50k and 1nF. It must be fed from a low impedance source and feed a low impedance.

 

[Willen]

I don't know which limiter circuit you are talking about. The PiRa, Harry's and my Mod4 circuits use a parallel capacitor and resistor.

51k and 100nF make 5100us, not 50us. For 50us then you need 50k and 1nF. It must be fed from a low impedance source and feed a low impedance.

I am talking any limiter made by 2180 chip without its another chip 2252. (didn't get exact circuit till now)

Oops yes, I made mistake, I was trying to say 51k+1nF in series before limiter.

 

[audioguru]

I am talking any limiter made by 2180 chip without its another chip 2252. (didn't get exact circuit till now)

You need an average level or a peak level detector circuit, preferably with full-wave rectification. The datasheet for the LM3915 has some full-wave peak detector circuits.

(Pre-emphasis) Oops yes, I made mistake, I was trying to say 51k+1nF in series before limiter.

That is not pre-emphasis. Pre-emphasis boosts high frequencies above 50us which is 3200Hz. So 3.2kHz is boosted +3dB, 6.4kHz is boosted +6dB and 12.8kHz is boosted +12dB. Your series RC does not boost high frequencies like that. Instead it cuts low frequencies so the highs have a flat frequency response and 3.2kHz is at -3dB, 1.6kHz is at -6dB, 800Hz is at -12dB, 400Hz is at -18dB, 200Hz is at -24dB, 100Hz is at -30dB etc. (no bass).

Instead the 51k resistor and 1nF capacitor are in parallel so the resistor forms an attenuator for low frequencies and the capacitor passes more and more at high frequencies.

 

[Willen]

wow......... I am getting 2252 RMS detector chip with 2180 VCA!!! I got my all problem of component has been solved!! Thank you dear CanandaElk!!!!!

Audioguru- I will ask for good pre-emphasis showing schematic diagram then will do as you said! It's really Mid-night here, woke up at 1am and didn't sleep till now, have to sleep.

Audioguru, have to tried the limiter/compressure tiny software? Is it working you found?

 

[audioguru]

Audioguru, have to tried the limiter/compressure tiny software? Is it working you found?

I downloaded it and WINRAR unzipped it. Then I downloaded Java Runtime Environment but I cannot open the limiter/compression software.

 

[Willen]

Hi after so... long time,

A member of ETO and Canadian engineer/designer 'canadaelk' sent so me parts related to a compressor circuit (attached below) and he sent some different part than a real schematic, suggesting me to change the main IC. due to my poor mind on electronics, I could not decide what to do for better result. Even some basic bias factor made me lot more confusing! So i tried to contact to him in inbox, but I feel he use the ETO very less and reply very less. According to him I need to replace the IC SSM2013 VCA with THAT2155 because he sent me THAT2155 IC instead of SSM2013 and said to replace. Circuit has designed with SSM2013 IC- see Compressor GIF copy.gif.

To do that myself I downloaded some related 'design notes' from THAT corporation. They are showing some VCA based on THAT2155 but different bias schematics like 'Ec+ driven' and 'Ec- driven' made me confused me LOT! I am worrying so want to get a better suggestion to submit my this one hobby project.

In the compressor schematics below, named as 'Compressor GIF copy.gif' I marked three connections A, B and C. And in the THAT2155 based VCA (I found at THAT's design note), named as THAT2155 VCA bias with ec+ and ec- dr.gif there are two circuit with ec+ and ec- driven. and there also I marked A, B and C connections. Can I delete SSM2013 stage and connect these A,B,C connection marked at VCA schematic? Which one I need to choose among Ec+ and Ec- driven?

(THAT design note137.pdf of THAT has nice guide to replace the IC THAT2155 with another one and has some schematics too. Due to its completed thing, it is not being helpful to me. Sorry for waking up again!

 

[audioguru]

Hi Willen,
I HATE the sound produced from a single compressor. It senses all frequencies then compresses all frequencies. Then a loud bass beast causes compression of the background music and vocals that sounds awful because then their levels are pumping up and down. My local TV station uses too much compression for the news announcers then before they begin speaking the gain is maximum causing their first breath of air to BLAST at full volume before the compressor turns down the gain for the voice. Any loud sound kills the voices for a moment when the compressor turns down the gain before the compressor returns the gain to normal.

A half-decent compressor circuit has a more than one compressor (one for each frequency band) then a loud sound on one band is compressed without compressing the other bands. Dolby noise reduction and DBX did that many years ago.
Guess what? The DBX company became the THAT company.

I play music on my sound systems without using a compressor. But some of the music is already compressed. Are you recording music that has a very wide range of loudness?

 

[canadaelk]

Willen:
I had answered the questions you asked last fall. My mail did not bounce back, so I assumed you got my answer. I do not keep copies of these things, please pm me again and I will respond.

Audioguru:
You are correct in post #72. However, there is no point in confusing Willen on the finer points of compression technology, when all he wants is to learn some basics. Once he has built one and learned how they work, I am sure he will get himself to HP, BP and LP filters. E

 

[Willen]

Hi Audioguru,
Your sentences are so much interesting! And little funny to hear about TV announcers' blasting voices with compressor and VCA . For me, I want to insert music and want to compress music and want to transmit and want to see its amazing changes! I will accept even if I listened awful. I feel so fun to do any project and the project is amazing for me because the related parts are rare for me. Really thanks to 'canadaelk'. I am excited to listen modified sound or anything else. As I said before, I am from remote place so I won"t get such ICs and parts anymore here. So feel to anything if I got parts. It's fun to me.

Hi Elk,
I don't know what happened with your PM reply. You can check your inbox in this ETO. Yes I got your mail and few pages guides too with schematic and related THAT parts. You explained their so nicely but you didn't wrote in detail about to replace SSM2013 with THTA2155. So I searched THAT2155 schematic myself around THAT design niotes and showed here. Can you read my actual problem carefully at post #71 and reply me, here or on PM? There is very old message there in your inbox, for same problem. Thank you so much for returning back. Feeling that I found a lost person

 

[BobW]

I realize that I'm arriving late to the game here, and hope that this doesn't muddy the waters, but the discussion of substituting a MOSFET for the JFET in the FET attenuator circuit is very interesting. The fact that the US5808516 patent drawing in post #12 appears to be essentially the same as the JFET attenuator version implies that this is possible. However it exhibits the same problems, because the 1/2 Vds signal is fed back through a capacitor and is subject to the slow attack problem. A different method of applying the 1/2 Vds to the gate which avoids the capacitor was given in an article by the late Prof. Marshall Leach:
https://users.ece.gatech.edu/mleach/papers/limiter.pdf
It has been designed with his typical thoroughness, and is well worth a read. I suspect that a MOSFET could replace the JFET in this circuit with some bias modifications to produce good results.
Prof. Leach used four of these limiters as part of a complete broadcast quality FM audio processor. Also well worth a read:
https://users.ece.gatech.edu/mleach/papers/fmlimiter.pdf

 

[audioguru]

My local TV station uses too much compression for the news announcers. When the show begins the compressor senses no sound levels so it forces the gain to be maximum. Then when the announcer takes that first breath of air it BLASTS as loud as it can before the compressor reduces the gain down to normal. The compressor used on some radio stations causes the same effect to singers where each breath of air is much too loud.

Prof. Leach says that his audio limiter (not a compressor) rarely exceeds 6dB of limiting the peaks. It prevents the FM transmitter from producing excessive deviation which might cause interference to an adjacent station.
An audio compressor compresses the dynamic range of the audio much more than only 6dB so that loud sound levels are reduced and low sound levels are boosted.

Audio is supposed to have a wide dynamic range from low levels to loud levels.

 

[MrAl]

Hi,

I suppose you could change the dynamic limiting by changing a couple resistor values.

But these kinds of circuits limit another thing besides the audio signal: they limit us to analog technology which can not be as effective as digital because audio is casual in real time whereas digital can be made to look like it can see into the future which means it can predict exactly what the incoming audio will be in the future and thus adjust the gain for the required attenuation. The sequence of sounds will still appear natural to the listener.
Of course it dosnt really see into the future, but because it can delay the audio it can also delay the attenuation and during that delay make the necessary adjustments. This would work very well with content that is only audio, but with audio and picture content we'd need to delay the picture by the same amount. Normally this would not hurt anything but it would not work very well for a live performance.

 

[audioguru]

I worked with high quality digital telephone conferencing products. The digital compressor was so slow that it was obtrusive and very obvious. It operated in real time without having a delay allowing it to "see into the future". Half-decent analog compressors work much better. Compressors were used because some people scream into their phone and other people whisper.

With a delay then the circuit can even censor out profanity or certain phrases from being heard.

 

[ronsimpson]

I worked with high quality digital telephone conferencing products.

the circuit can even censor out profanity or certain phrases from being heard.

That's what I need. A audio processor that will censor what I am saying.
"YOU @!%#$^*%^%$% USE YOUR F'N TURNING SIGNAL" becomes "What a sweet ride you have. Have a good day."

 

[MrAl]

Hello,

AudioGuru:
Think about it. The main problem with compressors is they dont know when that long silence or low level audio is going to become a very loud burst, and they dont know when that very long loud sequence is going to becomes more quiet time. If they could know that, they could determine what the gain should be for that segment of audio BEFORE it gets output to the speaker. So DSP would solve one of the biggest problems with compressors.
There's always the chance that somebody uses an analog delay line, like the old 'bucket brigade' chip that used to be around.
The designer has to know what they are doing too, and that's a lot to ask for these days