Opamp Single Supply - TL072

[Suraj143]

I'm making a tone control circuit for TDA7377.Going to use single supply 12V to power the pre amp tonecontrol circuit.
I'm making a baxandhal type tone control circuit.To feed the audio signal to the tone control circuit I use a TL072 buffer.I'm confused what type is best for enter stage circuit out of these two?

Thanks

 

[AnalogKid]

The left circuit will not work.

The right circuit will work. Note that the right circuit is inverting.

ak

 

[Suraj143]

The left circuit will not work.

The right circuit will work. Note that the right circuit is inverting.

ak

Thanks.I will use the right one.Is that inverting thing ok?

 

[gophert]

Thanks.I will use the right one.Is that inverting thing ok?

You'll be fine. Just make sure your output signal is less than 8v peak-to-peak.

 

[Pommie]

Is that inverting thing ok?

An inverted sine wave is still a sine wave. If it's a sawtooth wave then inverting may be a problem.
I'm assuming Audio due to tone control so there should be no problem.

Mike.

 

[Suraj143]

You'll be fine. Just make sure your output signal is less than 8v peak-to-peak.

Many thanks.I planned to power the circuit using a 12V supply.But sometimes giving from an unregulated supply (12V transformer) so it may come 17V DC to the opamps

 

[AnalogKid]

Thanks.I will use the right one.Is that inverting thing ok?

Depends.

For an intercom - sure. For critical listening or scientific work, no.

The ultimate goal of audio electronics is to produce sound into the human ear. Faithful reproduction involves both amplitudes and phases, and the biggest phase of all is the direction of the speaker cone. The microphone sees positive (compression) and negative (rarefaction) instantaneous air pressure during each half-cycle of whatever the sound is, and the speaker should mimic this. That is, at a point in the audio when the microphone diaphragm is being pushed inward by positive air pressure, the speaker cone should be moving outward to create positive air pressure. By convention, positive air pressure against the diaphragm is represented by the positive half-cycles of the audio signal, and positive half-cycles push the speaker cone outward. This is why amplifier outputs and speaker terminals are marked with some kind of polarity indication - it matters.

The air-to-microphone-to-signal-to-speaker-to-air relationship can be maintained through the entire signal chain from the microphone in a recording studio to wireless ear buds while jogging by having either zero or an even number of inverting stages. Of course no one can control everything, but each stage along the path can be controlled to be non-inverting between its ins and outs.

So it comes down to quality versus effort (and cost). How much of a "golden-eared wonder" are you (or your customers), what is the application, etc. - ?

ak

BTW, I use the term "golden-eared" only semi-sarcastically. Thanks to a couple of my past lives, I'm pretty good; but I know some people who are outright scary when it comes to audible perception. An easy example is the 1812 Overture ("The Year 1812 Solemn Overture" by Tchaikovsky). There are actual cannon shots in the musical score. The initial wave front from an explosion is compressive, and as such is quite directional. If the signal to the speaker is inverted WRT the microphone, the speaker moves inward rather than outward, creating a "blast" of low pressure - a micro-vacuum. Thing is, evacuation is not nearly as directional as compression. In a controlled listening test, a random dude off the street might not be able to say why "identical" signals A and B sound differently, but over 90% will be able to hear a difference.

 

[Nigel Goodwin]

Depends.

For an intercom - sure. For critical listening or scientific work, no.

The ultimate goal of audio electronics is to produce sound into the human ear. Faithful reproduction involves both amplitudes and phases, and the biggest phase of all is the direction of the speaker cone. The microphone sees positive (compression) and negative (rarefaction) instantaneous air pressure during each half-cycle of whatever the sound is, and the speaker should mimic this. That is, at a point in the audio when the microphone diaphragm is being pushed inward by positive air pressure, the speaker cone should be moving outward to create positive air pressure. By convention, positive air pressure against the diaphragm is represented by the positive half-cycles of the audio signal, and positive half-cycles push the speaker cone outward. This is why amplifier outputs and speaker terminals are marked with some kind of polarity indication - it matters.

The air-to-microphone-to-signal-to-speaker-to-air relationship can be maintained through the entire signal chain from the microphone in a recording studio to wireless ear buds while jogging by having either zero or an even number of inverting stages. Of course no one can control everything, but each stage along the path can be controlled to be non-inverting between its ins and outs.

So it comes down to quality versus effort (and cost). How much of a "golden-eared wonder" are you (or your customers), what is the application, etc. - ?

ak

BTW, I use the term "golden-eared" only semi-sarcastically. Thanks to a couple of my past lives, I'm pretty good; but I know some people who are outright scary when it comes to audible perception. An easy example is the 1812 Overture ("The Year 1812 Solemn Overture" by Tchaikovsky). There are actual cannon shots in the musical score. The initial wave front from an explosion is compressive, and as such is quite directional. If the signal to the speaker is inverted WRT the microphone, the speaker moves inward rather than outward, creating a "blast" of low pressure - a micro-vacuum. Thing is, evacuation is not nearly as directional as compression. In a controlled listening test, a random dude off the street might not be able to say why "identical" signals A and B sound differently, but over 90% will be able to hear a difference.

Sorry, but I think you're completely wrong- speaker terminals are marked +ve and -ve because it's vital they are connected in phase with each other, and not out of phase - not because of any desire to hopefully keep the phase correct from microphone to speaker.

Even assuming 'speaker moves inward rather than outward, creating a "blast" of low pressure' - this would only be for the first half cycle, then it's going the back other way.

I seriously doubt hardly anyone (if anyone at all?) could detect a difference in double blind tests, and certainly nothing like your claimed 90%.

 

[audioguru]

Your brain does not care about actual true phase. If your eardrum moves in or out then your brain senses only a movement, not a direction.
I can make POP with my lips by sucking air in or blowing air out. They both sound the same.
But both ears must have the same phase.

The biasing resistors for the opamp must have a well filtered voltage. An extra RC filter is needed to feed the biasing voltage divider to suppress hum from a non-regulated power supply.

The buffer is too complicated. For the high input impedance but very low output impedance do this:

 

[Suraj143]

Ok thank you everybody.For the last few hours I was searching for a better input design.I found one & make a drawing.PLease see the attachment.
1)What is the purpose of the 1 Meg resistor?
2)Is that final R8 is value ok?
3)I'm also doubt on electrolytic capatior directions..!!

 

[gophert]

The JFET input of some op amps doesn't allow any current to flow in/out. It works only on voltage. So the cap in series with the + input needs somewhere to discharge otherwise a high or low "floating" footage will develop and saturate the output. The 1M resistor keeps the + input at a reference to your virtual ground.

 

[AnalogKid]

I seriously doubt hardly anyone (if anyone at all?) could detect a difference in double blind tests, and certainly nothing like your claimed 90%.

1980's. Large mid-western land grant university, psycho-acoustics research. Soundproof and anechoic room, random subject pool selections from undergraduate, graduate, faculty, and off-campus populations. This was preliminary testing to determine which technical factors might affect the main experiment results.

ak

 

[audioguru]

1) What is the purpose of the 1 Meg resistor?

If the 1M resistor is replaced by a piece of wire then the filter capacitor C6 (it filters the bias voltage) will short the input signal to ground.

2) Is that final R8 is value ok?

100 ohms is fine to isolate the opamp output from the capacitance of a shielded output cable. The capacitance of the cable directly in the opamp output would cause the opamp to oscillate at a high frequency.

3) I'm also doubt on electrolytic capacitor directions.

You are correct because the polarity of C1, C2 and C5 are backwards. C6 is correct.
C1 will take too long to charge because its 1uF value is WAY too much to feed the 1M resistor R11. Use a 0.033 (33nF) film capacitor instead.

 

[Suraj143]

Ok thank you.I redraw some changes in the capacitor polarity, Also marked the polarity.

1. Please check the capacitor polarity.I still confused on that.
2. I want to change the values on C1 & C2. Is my new values marked in red ok?

 

[audioguru]

Your schematic shows an opamp with no part number. Somebody might copy it and use a different opamp that cannot work properly with the 1M resistor for R11.

The polarity of your capacitors is correct.
C2 has no voltage across it so its polarity does not matter. C2 is not even needed, use a piece of wire instead.
Calculate the cutoff frequency of 0.1uF (a 104 type) into the 1M resistor R11. It is 1.6Hz. Can you hear that low? Can your speakers play sounds that low?
That is why i recommended 33nF.

With C2 as 1uF then 20Hz is cut a little if the bass control is at maximum. C2 should be about 2.2uF or a piece of wire.

 

[Suraj143]

ok thanks AG.
My opamp is TL072.I'll remove the C2 & use a wire, so I can save my PCB space as well.

Thanks

 

[Suraj143]

I have finalised the schematic for this tone control.Please check the errors & unwanted parts. Thanks.

Note that, I'm going to power this from a 12V transformer.So the DC will rise around 17V.Doubt on this.Thats why I put a 100R resistor prior to feed the opamps power.
Also there is an ability to give the supply from a 12V battery (without a transformer).In this case do I still need the 100R?

 

[rjenkinsgb]

The link is not accessible, it says it is private.

The TL071/072/074 can operate at anything up to 40V single supply, so your 12 to 17V will not be a problem.

 

[Suraj143]

The link is not accessible, it says it is private.

The TL071/072/074 can operate at anything up to 40V single supply, so your 12 to 17V will not be a problem.

Sorry Here is the attachcment.Then what about the virtual ground on opamps inputs..!
Thanks

 

[rjenkinsgb]

It will just stay at around mid-supply, it's fine.
Just make sure any electrolytic caps are rated somewhat above the actual voltages they will be working at.

To possibly simplify the overall circuit, you could use the same bias divider to feed the input of your mic preamp via another 1M resistor; though I'd probably add a 0.1 polyester cap across the 47uF to ensure no possible high frequency feedthrough.

Probably not actually needed, just me preferring belt-and-braces type filtering.. It's easier to leave something out than add it in, later on.