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walid

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Posts posted by walid

  1. Hi, I see that we limit the problem in two things:
    (1) (a) Grounding problem and I must check this. Can anybody tell me the fastest and best method to check this, what instrument may I use and what acceptable reading should I looking for, how to decide that ground is OK or need to strengthening it.
    (b) Tnk2k said: “If it is a big problem and you can not identify the source in a reasonable time, use isolation transformers to break the ground loops.” What type of transformers can I use and where to connect it?
    (2) Existence of power cables very close to signal cables, this is difficult to make them far from each other, can I use a low pass filter to pass hum (50Hz sig.) to ground, or what you comment.
    Thank you all very much, Walid Farra.
    ???

  2. please I need more info.....
    why hum increases when the main AC power is taken from our generator.
    It is true, there is power lines in parallel to signal lines in the same place, is this one of the causes producing hum.
    I notice that when toutching the receiver  outside body you feel some electricity, is this related to hum.
    thanks

  3. Hi, thank you all for your responds, first of all I want to tell you that the hum in audio is 50Hz and in video is two bars scanning the monitor screen from top to bottom or from bottom to top, the scanning is very slowly.

    Mr. Bjorn said that: “Your equipment is connected so it produces a ground loop and thus causing hum, that's my contribution”
    I think this ok but please tell me how an equipment may be connected so I produces a ground loop? And how we overcome this?
    Thanks, 
    Walid.
    :o

  4. In Palestinian satellite channel when a signal come to the continuity studio from outside, there is a hum on it (hum in video and audio), and this is clearly obvious when we take the main electric power from our generator, but the internal signals from the studio (cameras and VTRs ....) are very good and have not any hum. Sometimes the hum is only on audio especially when we take the sig. from Microwave. Please help me to solve this problem.
    Thanks.

  5. I read this:
    "The ideal parameters are never fully realized but they present a very convenient method for the preliminary analysis of circuitry. So important are these ideal definitions that they are repeated here. The ideal amplifier possesses
    1. Infinite gain
    2. Infinite input impedance
    3. Infinite bandwidth
    4. Zero output impedance
    From these definitions two important theorems are developed.
    1. No current flows into or out of the input terminals.
    2. When negative feedback is applied, the differential input voltage is reduced to zero."
    My question is about the last line, that is what the meaning of the differential input voltage?

  6. Hi audioguru
    Your answer above needs more explanation in the following points:
    (1) I'm sure that I read some where that the tel. designers make part of MIC sig. fed back to speaker network and they do so to prevent the user of tel. from shouting. This is true, if you can't hear yourself you may think that there are some troubles in the device. On the other hand you tell me the inverse that they design circuits to eliminate this by negative feedback. Where is the truth?
    (2) I understand you when you say: "The transmitted mic signal is reflected back to its speaker at the telephone line" I know that this is due to impedance mismatching, but please tell me where this negative feedback circuit in the companion diagram?
    (3) You said: "I think C5 cuts high frequencies from being transmitted." tell me please, The MIC sig come from MIC and through C7 to B of Q4 for amplification. When and where C5 takes place in cutting high frequencies from being transmitted?
    (4) You said:" The mic and speaker signals are usually switched one-at-a-time in speakerphones to avoid local and telephone line feedback". Is this done inside tel. device or in the central office? 
    Thanks.

  7. No No No you all not understand me at all, you all say absolute talk, the net is full of such talk, I ask obviousness question that:
    I need a theoretical information about designing an impedance matching circuit using a common base transistor configuration. The transistor is PNP small signal type, collector is i/p and Emitter is o/p.
    In other words, I want someone to tell me that take a type xxxxx PNP transistor and connect it like Fig.xx (attachment) then calculate resistor & capacitors values according to equation xx, OR simply advice me to go to link www.XXXXXX.com where I can find my answer.
    If anyone can't do so, don't tire yourself, leave it to the suitable person.
    I'm very sorry for saying this, but I must say this.
    Thanks for all of you who try to help me.  ;D

  8. As shown in the schematic, the MIC signal travel from MIC to C7 (0.04uF) to the base of Q4 for pre amplification.
    I read somewhere that part of MIC sig is fed back to the speaker driver (Q5) so that the user can hear himself, if they don't do this the user shouting.
    Now is this part of  MIC sig passthrough:
    (1) C5 directly without amplification to R20 then to base of Q5, if not why C5 placed here?
    (2) Collector of Q5 (after amplified) then into R20?
    (3) Both (1) & (2).
    Thank you.

  9. Very good, this is what i looking for, floating i understand it, but please tell me:
    (1) What happen when it turned on.
    (2) What the difference between a PNP transistor's collector, and NPN one.

    (3)Are you mean that the MIC acts as a resistor and the voltage drop across it =2.4Vdc 

    thanks

  10. MIC has 2 pins one grounded (the -ve) and the other connected through a resistor to the power supply, here in our circuit, the +ve MIC terminal is connected through R17 + R15 to 7.4 Vdc. To MUTE this MIC, the +ve terminal of it should connected to Ground, if it connected to 2.4Vdc as shown in the diagram, i see that equivalent to replace the 7.4Vdc by (7.4-2.4 = 5Vdc), so i think that this pin (#13) gives two outputs:
    (a) When dialing ====> Ground to mute audio network
    (b) When no dialing ===> 2.4vdc to compensate the 7.4v to 5v.
    is this true ?????????????

    Secondly: you said: "Maybe it is an open-drain P-channel Mosfet that conducts    during dialing ....."
              I read about something like this in many datasheets but they said about open collector, please what this mean?
    thanks

  11. hi, i'm comig back ..... i love you all
    my question is : All DTMF ICs used in telephone circuits have pin called
    MUTE (pin 13), in HM9102 IC datasheet i read the following "This output is
    an inverter normally at low state when there is no key entry. During
    outdialing it changes to high state and is used to mute the speech network."
    Assuming that low state = ground and high state = VDD = 4.3V, and looking to
    Fig.3 i think that the inverse must be done, that is:"normally at high state
    when there is no key entry. During outdialing it changes to low state and
    is used to mute the speech network".
    when it in low state, the two terminals of MIC are grounded so no voicesig
    canout from MIC, thismust be done during dialing. is thisture?
    thanks. 

    post-2833-14279142296915_thumb.jpg

  12. Hi Mr. audioguru, When I say "Please I want a direct answers" I don't mean a brief answers, in short you must know that you and this site is the only source of information for me.
    Now lets go....
    Question 1: Why always PNP and not NPN?
    Your answer: To make a positive supply voltage. An NPN would make a negative supply.

  13. From all of this discussion with Mr. audioguru about telephone diodes I can tell others that in general any diode in conduction state(forward bias) acts as a very small resistance and it can pass a small signal in two way direction, in other word look you can look at this diode as a small segment copper wire. isn't true Mr. audioguru.
    thank you very much for reaching this big result....

  14. From all of this discussion with Mr. audioguru about make/break ratio i conclude:
    The make/break ratio is the ratio of the time that rotary dial pulsing makes the phone's connection to the time it breaks the connection. Europe has a different standard than North America. Setting M/B pin either high or low selects the ratio, 50:50 or 67:33. these ON - OFF (high - low or 1 - 0) swquence are given from some pin remarked as PO (out put pulse) or sometimes as DP (dialing pulse), when po (or dp) out 1 the telephone connect to line, where if po = 0 the telephone disconnected. For example if you press number 2, then PO (or DP) output sequence is 1010, and if you press number 4, then PO (or DP) output sequence is 10101010. You can do this maually (as audioguru tell me) by pressing and release the hook-switch by hand, for dialind number 3 press and release the hook-switch 3 time but you must do this acurately.
    If M/B ratio = 50:50 this mean the on time = the off time, the only thing that i can not tell you about the value 50: 50 i guss 50 is the time period in mellisecond of ON or OFF.
    thanks to audioguru.


  15. %7Boption%7D
    I have more than 10 telephones schematic diagrams one of them is a very simple {from: circuit ideas} see Fig.1 and I note that all of them use a small signal PNP transistor (T1 in Fig.1 and Fig.2).
    1) Why always PNP and not NPN?
    2) Why it always connected as a common-base configuration?
    3) How T1 deals with the two audio sig. (Caller and user)?
    Please I want a direct answer and it is good idea if give me a link about this subject to read then back to ask about what I could not understand.
    Thank you and I respect you.

    post-2833-14279142290123_thumb.jpg

    post-2833-14279142290229_thumb.jpg

  16. OK, this is a good answer, what I understood is when I dial number say 3 I hear teck - teck - teck -  which mean that telephone connect and disconnect three times that is M/B = (time of 1 teck) / (time of one -). Now the new question is if I live in USA and M/B ratio there is say 67:33 and I built a telephone device has a DTMF IC (say UM9102) and I connect pin 10 to VDD, that is choose a M/B ratio = 1/2 (not 67:33), is this telephone enable to call?
    Secondly: is it true that M/B ratio takes place only in the pulse mode and
    not in the tone mode?

  17. Hi Mr. audioguru, I'm coming back, I know I annoying you with my foolish
    questions but i expect patient, so lets go back to our discussion:
    Believe me i agree with all your saying, but your answers are not pour
    directly at what i ask, please tell me directly if true or false and tell
    me where i'm true and where I'm false, ==>
    please look at Fig.2, if a certain amount of DC current continuously pass
    through the diodes in the direction from line to telephone (not vice versa)
    you will agree with me that only D1 & D4 are forward biased continuously, OK.
    Now if we put a signal like that in Fig.2 in the location shown, tell me
    exactly through any diodes it will pass and how.
    sorry for my rambling and hope you can understand this with my bad grammar
    Thanx a million for the information....
    NOTE: I want to tell you that it takes more than an hour to prepare this
    small letter for sending.

    post-2833-1427914228985_thumb.jpg

  18. Hi Mr. audioguru, thank you very much for your response, I agree with you, there are two of the diodes are forward-biased all the time and I'm sure that rectification of the audio doesn't happen, but how??? Let us examine what I think (it may be wrong!). Look at fig.1 if we put a sinusoidal sig of 1V peak between T & R (Caller audio), and if we assume that there is no 8.3 VDC between T & R, then I expect to receive a rectified sig of 0.4V peak between P1 & P2. BUT since there is 8.3 VDC D1 & D4 are already conduct so a full sinusoidal 1V peak appear between P1 & P2 "I think this is true, please comment".

    post-2833-1427914228812_thumb.jpg

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