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rs232 tx to XLR audio

Discussion in 'Audio' started by pieman, Mar 26, 2014.

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  1. pieman


    Mar 26, 2014
    I am a bit of a noob to electronics with some soldering experience but incrreasing I feel I need to learn a bit more about electronics

    I have a project where it seems sensible to record rs232 data along with video data so I thought why not use the not very compressed pcm audio tracks(24bit 48Khz). The inputs to these are analogue.

    I would like to take the tx from an rs232 data stream +-25volts normally maybe +-15voilts and transformer it down to +-1.736V so I can record it as anolugue->digital audio.
    I should be able to get my data in the sample rates, assuming I don't go above Nyqvist law so max 24000 bits per second?

    1) Can anyone suggest a circuit/components that could reduce the rs232 tx voltage to broadcast line level audio +-1.736V?

    2) I asume a simple amp at the other end could transform these logic levels back up to around +-15volts rs232 levels. Can anyone suggest an amplifier?

    Many Thanks
  2. OLIVE2222


    Oct 2, 2011
    log RS232 data on a audio track is not very efficient. It's closser from a digital strorage oscilloscope than a digital logger. You are going to use 24 bits where only one is needed. Program like terra term can log RS232 data in a text file. However it's possible and I guess you may have some synchronization needs to want it. You can use an IC like the MAX3232 who can handle the data both way. When supplied with 3 volts you will have 3 volts level to feed one of the XLR input. If it's still too high you can use a resistor based voltage divider. With the 48KHz sampling rate you can theoretically go up to 19200baud (assuming 1 start bit, 8 data bits and one stop bit)

  3. KrisBlueNZ

    KrisBlueNZ Sadly passed away in 2015

    Nov 28, 2011
    I don't think that will be very workable. RS-232 serial data can't reliably be fed through an AC-coupled signal path. Even if the data is continuous (no long periods of idle line), the DC average of the data will change constantly.

    Normally you would use a modem for this. There are several types of "blind" modem - modems that simply convert a data stream into an audio stream which can be recorded, played back, and demodulated by the other half of the corresponding modem. FSK is a simple modulation method that you could use, but it needs a lot more bandwidth than the rate of the data. More complicated modulation schemes make better use of bandwidth but require a cleaner signal path.

    Laying down digital information on an audio track can be done using timecode (that's SMPTE timecode, not IRIG timecode) and there's an interesting article that describes the timecode formats at LTC is the format relevant to your needs, and that article describes the "modulation" (if you can call it that) that LTC uses, which is a "biphase mark" system. It requires a minimum bandwidth of AT LEAST twice the bit rate; I would suggest 3~4 times the bit rate for good reliability. You'll also need to make sure that there's no noise reduction enabled on the audio track!

    Generating biphase mark data is fairly straightforward, but decoding it is another story! Feel free to post again if we can be of any more help.
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