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# compressor effect in audio signal

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#### roomy

##### Junior Member level 3
Hello,
how a clean audio signal modify by analogue components to give compressed signal effect .

thanks

you could try using a log amplifier. its an op-amp circuit. much like an inverting amplifier circuit, but replace the feedback resistor with a diode or a transistor with the base grounded.

i suppose this is what you meant my compressor effect.

roomy

### roomy

Points: 2
hi ,

do you have block diagram for this compressor effect .

thanks

I don't have a block diagram as such, but here's the schematic.

i created this using lt-spice. and i haven't included the power supply for an op-amp for the sake of clarity.

Let me explain the circuit.

An op-amp is a differential amplifer, i.e, it amplifies the difference between the non-inverting terminal(marked by +) and the inverting terminal (marked by -). Here, the + terminal is grounded. Usually the gain of an opamp is in the order of 10^ 5 (100,000),i.e, if the input signal to the opamp is 1 milli volt, the output will be 100V, in the same time duration of the original signal.

now, any element connected in the feedback path of an exceptionally high gain amplifier will behave as an inverted element. in short, a capacitor will behave as an inductor and an inductor will behave as a capacitor, i.e, the transfer function of the element in the laplace domain will be inverted. Here, in the log-amplifer, we have a transistor which is on (a pnp transistor). The transistor produces an output voltage that is a function of the exponent of the input current. So, V out is proportional to e^ ( input current). (this is not complete but this formula will do for the intuitive understanding of the circuit). Since the transistor is connected in the feedback path, it will behave in an inverted fashion. i.e, the exponent operation will become inverse exponent or natural logarithm operation.

please let me know if you want any further explanation or if you can't understand this.

( the circuit was from the book by sergio franco )

roomy

Points: 2
roomy

### roomy

Points: 2
Audio compression is a complex subject. This article: **broken link removed** from the Rane Library gives some excellent details on the priniples involved.

roomy

### roomy

Points: 2
Some modern techniques to achieve inteligent ( eficient ) audio compression consists in deenphasize the spetrum band range less used at a time. Usually other most common methods performs compression at a pre defined range only based on human spectrum response.

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roomy

### roomy

Points: 2
Thanks every one but i still confused by how a clean audio signal modify by analogue components to give compressed signal effect

Maybe you need elaborate more the question to clairify you doubt.

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Audio compression usually wants to keep an undistorted signal and primarly modify the signal envelope. A log amp would cause significant signal distortion.

Particularly the manufacturers paper linked by rogs has a good description how standard audio compressors work. You can also review the datasheets and manuals of the compressors made by rane.com. They even have schematics of some instruments.

But I agree, that the intention of your question isn't quite clear. Do you want to understand it's operation or design your own audio compressor? In both cases, you should tell more details.

Logaritmic compression was used at hardware level in some broadcast systems, not in modern storage codecs.
The target application would help sugest the best approach.

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Logaritmic compression was used at hardware level in some broadcast systems, not in modern storage codecs.
You possibly refer to the µ-law codecs used in telephone systems? In contrast to the DC logarithmic amplifiers quoted by ninju, a logarithmic compression characteristic for an audio signal must be bipolar, like y = sign(x)*log(abs(x)). And it would need an inverse decompression characteristic at the receiver.

The term "compressor effect" in contrast refers to a dynamic manipulation, that's not intended to be reversed at the receiver. Reversable dynamic compression as e.g. implemented with various Dolby processors is another topic. Interestingly, it also uses a gain variation (partly frequency selective) rather than applying a non-linear characteristic to the signal itself. The difference matters!

hi FvM

I noticed the LOG compressor used to limit amplitude in a RF AM transmiter circuit much time ago ( ~1980 ); however I did not analysed more deeply the working theory, but only confirmed this concept was applyed there, observing the circuit topolgy. The transistor was operated in a conduction region in wich gain varies with current.

That transformation is not reversable at receiver side, and the disttortion you had mentioned occurs; but did not affect so hardly speech inteligibility. The system operation were analogic an no one quatization were performed.

In fact, at modern systems ( both storage and broadcast ) that loss of quality are no more aceptable and due everithig is made digitally, reverse transform to decompress may be performed by codecs.

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i think
the compressor can be reduced the out put waveform by reduction amplifier .also this block diagram i posted can be corrected .if any comment please send to this

also this block diagram i posted can be corrected
Sorry, to blurry to identify any details.

I noticed the LOG compressor used to limit amplitude in a RF AM transmiter circuit much time ago
The compressor circuits I know (and used in some applications) are feedback AGC circuits with a transistor utilized either as variable resistor or variable gain amplifier. They produce some distortions but have still almost linear behaviour in the AC signal path.

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if you want genuine audio compression, you'd have to go in for a digital assembly. linear predictive coding is excellent if you want to compress voice, even though the output voice will be a sort of robotic voice. you'll know that a human didn't produce it. So, they modified it and brought in QCELP and RELPC, which are used in CDMA and GSM (mobile telephone technologies.).

As far as analog goes, i think any compression circuit is meant to distort the signal.By doing this, one allots more bits for the weaker parts of the signal and relatively less bits for stronger parts of the signal. Please refer to the chapter on pulse code modulation on any standard communication text book.(i referred to simon haykin). If you want to reduce the amplitude alone, you might as well use an attenuator.

If you want to build a very simple compander, i'd suggst a diode function generator. Use a circuit consisting of resistors, diodes and voltage sources which can modify the incoming signal. Please refer to the working of a triangular wave to sine wave converter. It too is performed by a diode function generator.

If you need help , pm me.

As far as analog goes, i think any compression circuit is meant to distort the signal.
I don't agree. Review the professional audio dynamic compressors, particularly those described in the link given by rogs. It's true of course, that recent instruments are mostly using DSP techniques which gives some special advantages like applying delay to the signal path. But nervertheless, high performance devices have been designed in a fully analog way for decades, and e.g. used in recording or broadcast studios. Be sure, that no audible distortions are tolerated in these applications.

But the information is at your fingertips, if you are seriously interested to learn about good old analog audio processing, you can get it.

yeah, but the term " no distortion " means same as the original signal. Even in the link that rogs suggested, since the gain changes after a certain threshold, the output signal will not be the same as the original signal and hence it can be called as distorted.

please correct me if i am wrong. To err is human.

---------- Post added at 19:22 ---------- Previous post was at 19:11 ----------

an addition to my previous post- by same as original signal , i meant uniform amplification. not where it changes half way through.

Related to audio signals, I would follow this statement
Distortion refers to any kind of deformation of a waveform, compared to an input, usually Clipping, harmonic distortion and intermodulation distortion (mixing phenomena) caused by non-linear behavior of electronic components and power supply limitations.
Distortion - Wikipedia, the free encyclopedia
In this view, dynamic processing won't be understood as distortion. The techniques you suggested, e.g. logarithmic amplifiers or diode networks are in contrast involving harmonic distortions. But I think, it's more interesting to discuss about applications and techniques than about definition of terms.

compressor can make the audio of more consistent level and eliminate inconsistencies developed by for example when the singer moving closer to and further froma microphone and also the block diagarm we can used diode as the ninju wrote above but i read about this and i found the tremolo circuit can be used as compressor

is it correct or not

thanks

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