Help me understand the concept

Discussion in 'General Electronics Chat' started by Mozee, Sep 16, 2016.

  1. Mozee

    Thread Starter Member

    Jul 23, 2016
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    FSS2PDMIE7J97GS.LARGE.jpg

    Hello,
    I would Like to understand how does this work and why did the designer use these values (Formulas is possible)
    Let me start with my own and please note I'm still learning here so bear with me please :)

    1-The first Op Amp from the left used in an Inverting input mode, I don't know why would we choose this mode if we want the input just to be amplified not changed in shape.
    2- I know that the amplification factor is Av=(R10+R11)/R9 so the input will be multiplied by Av and yields the output at pin10 but what if I want to use the non Inverting input? also why those values were used why not using1K for R9 and 1K for R10 and 10k for R11(Note a Double power source is used for -Ve voltage and +Ve voltage)
    3-What is the configuration of the second Op amp?
    4-Why R12 was used? is it really needed?

    Thanks in advance
    M.zee
     
  2. kubeek

    AAC Fanatic!

    Sep 20, 2005
    4,670
    804
    1)inverting or non-inverting the shape stays the same, only the polarity is switched. This does not matter if you do the same thing with both R and L audio channels.
    2) not sure why you´d want to use the non-inverting input, but you´d have to change the configuration into a non-inverting amplifier, and the rest would be basically identical.
    They choose R9=10k, because that is the load that is presented to device that drives this circuit, and it is kind of a standard load impedance in consumer audio.
    3) second opamp is a voltage follower with the difference that it has another voltage follower made of transistors on the output. It is used to increase available current to the load like headphones, and has gain=1. the feedback to the inverting input of the opamp makes sure that the opamp compensates for the nonlinearity of the transistors and you get the output as much as possible equal to the input.
    4)I don´t think that R12 is needed for anything.
     
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  3. ian field

    Distinguished Member

    Oct 27, 2012
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    The inverting stage doesn't change the waveform shape - it just turns it the other way up.

    There AFAICR: pros and cons with a non inverting stage - maybe the easy way is adding another inverting stage - it can be x1 gain.

    The buffer driving the complementary pair is a "trick of the trade". Normally without controlled bias the transistor pair would produce crossover distortion - particularly objectionable at low volume level. As long as the op-amp has fast slew rate, including it in the nfb loop makes it shoot through the crossover dead zone. Not exactly Hi-Fi, but its totally adequate for a test bench amplifier.
     
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  4. Mozee

    Thread Starter Member

    Jul 23, 2016
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    Thanks guys for helping me out, really appreciate it.
    @kubeek and @ian field

    I am aware of the cross-over distortion but I don't know how to fix it without offsetting the output!!! I want the output to swing about ZERO volts live up and down but if i bias the transistors, the output will be offsetted, am I right???
    What I am trying to do is to make a better design by producing a non-inverted , amplified and undistorted signal, as much as possible!!

    And please, could you tell me how to add a quote of a some portion of a post that I want to reply to?? I really don't know how yet :(
     
  5. AnalogKid

    Distinguished Member

    Aug 1, 2013
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    Besides the explanations so far, there are two problems with this circuit.

    1. The overal signal path from input to output is inverting. If youdo not know that the next thing that this circuit drives also is inverting, then the speaker phase will be incorrect with respect to the orginal signal source. That is, when a positive pressure wave hit the microphone in the original recording, that generated a voltage (usually positive). If the signal is inverted before it hits the speaker, then the speaker will produce a nevative pressure wave at that point in the audio. Not the end of the world, but poor system design.

    2. There is a deadband in the output stage between when the pin 12 is positive enough toturn on Q1 and when it is negative enough to turn on Q2. The opamp output has to zip through this dead zone, something that takes time and can be heard as a very strange kind of distortion in quiet passages. This circuit usually is the first one taught when discussing how to increase the output power of an opamp, as an introduction into all of the other, better ways. It is the most simple, but too simple for its own good.

    ak
     
    Last edited: Sep 16, 2016
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  6. AnalogKid

    Distinguished Member

    Aug 1, 2013
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    Lower right, REPLY
     
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  7. Mozee

    Thread Starter Member

    Jul 23, 2016
    40
    6

    Hello again AK,,

    I couldn't understand what you mean by your second note?! could you please explain? did you mean the cross-over distortion effect region ?
    And In your opinion, how to improve this circuit? I am actually using this design to build a wave generator and I would Like to feed the input from a smartphone which produces very precise wave forms!
    The actual designer built this but I am not happy with the inverting and the cross-over distortion at the transistors
     
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