add digital signal to analog audio signal - Page 1
Forum section: electronics / projects

author: Hieronymus
author_rank: Newbie
timestamp: April 27, 2016, 02:03:52 pm
content: Hi,I have a good project in mind with one technical challenge I have no experience with. I hope you guys can help me get started or share some ideas.In short I'm looking for ways to add a (simple) digital signal to an existing analog audio signal.I've build an intercom system I use for live video registration. It is a typical partyline system and is compatible with various professional systems. Over a normal microphone cable (XLR) an audio signal and power is transmitted. Each 'station' at the camera's has a headset on which the user can hear everybody and his own voice is mixed in with the bus signal. This enables front and back communication between director and camera positions. One this bus-signal everybody listens to the same signal, thus you cannot talk to individual stations.Most videomixers also allow for tally output. It's a simple signal that indicates which camera is 'hot' and which is set to preview (next to go 'hot'). Most existing solutions require another cable to distribute this signal (or expensive professional solutions that add video, intercom, CCU, audio and tally to one).I would like to investigate the options to add this simple digital signal to the audio signal and distribute it via the intercom. This means that each station needs to get somekind of an address (set via dipswitch) to only listen to tally signals for that station. Data rates are low 5 bits for an address will allow for 32 stations and 2 bits data would suffice (a camera can normally be not selected, on preview, on air or both). It's easy to understand that slightly bigger data words would make the system more flexibel as a cue-light system in theaters as well.Speed is not much of an issue, since I would find a reaction time within one video frame (1/25th of a second acceptable).So there are various cheap digital signals available like IR that meets these requirements. It would be easy to make something with a arduino to encode or decode this.My problem is that I have no ideas how to investigate if it is possible to add this signal on the audio and what to do to check if it will propagate properly through the existing system.Hope you guys like to think about this and maybe point me in the right direction.I'll be glad to answer any questions.

author: borjam
author_rank: Unknown
timestamp: April 27, 2016, 02:27:59 pm
content: Depending on your particular system, subtones might work.Probably it will have some kind of high pass filtering at the microphones, attenuating frequencies below 100 Hz or something like that, which carry no useable voice energy. Generally the filtering will be performed only at the audio inputs (microphones) but probably the rest of the system will happily transport signals with a frequency lower than the mentioned cut.This system is frequently employed, for instance, to activate ham radio repeaters.

author: Fungus
author_rank: Super Contributor
timestamp: April 27, 2016, 02:28:56 pm
content: Does it have to be inaudible?Quote from: Hieronymus on April 27, 2016, 02:03:52 pmI'd say that's a fairly big issue, eg. it rules out telephone DTMF beeps.

author: Fungus
author_rank: Super Contributor
timestamp: April 27, 2016, 02:33:09 pm
content: Quote from: borjam on April 27, 2016, 02:27:59 pmIf you have to send 8 bits of data in 1/25th of a second then sub 100Hz won't do it.

author: Fungus
author_rank: Super Contributor
timestamp: April 27, 2016, 02:43:52 pm
content: I'm thinking of how data was stored on old fashioned cassette tapes. You send pulses down the line, a narrow pulse is a zero, a wider pulse is a 1.There's no reason it won't work but it definitely won't be inaudible and you'll need a few extra bits for startup and some sort of checksum to stop it triggering on normal audio.

author: SeanB
author_rank: Super Contributor
timestamp: April 27, 2016, 02:51:55 pm
content: Use 38kHz IR transmitter and receiver units, like the Holtek 12E serieshttp://www.futurlec.com/Others/HT12E.shtmlThat way you send a 38kHz signal down the audio lines, which will survive the cable, and which is easy to extract with a high pass filter at each end to decode it. Gives you 255 address locations and a 4 bit data section for your use, plus it is robust and reasonably good at noise rejection on the input.

author: Jeroen3
author_rank: Super Contributor
timestamp: April 27, 2016, 02:54:32 pm
content: Use RDS.https://en.wikipedia.org/wiki/Radio_Data_SystemIt modulates stuff on the signal beyond human ears.

author: Kilrah
author_rank: Unknown
timestamp: April 27, 2016, 02:58:26 pm
content: My understanding is he wants this signal to be able to go through common audio paths, where any higher frequencies than usual audio would be cut out or at least attenuated by unknown amounts making the thing basically impossible unless it's allowed for it to be audible.

author: borjam
author_rank: Unknown
timestamp: April 27, 2016, 03:03:04 pm
content: Quote from: Kilrah on April 27, 2016, 02:58:26 pmThat's why I proposed low frequencies. In order to gain usable bit rate you can try several modulation schemes such as more than two tones FSK.Maybe frequencies above 5 KHz will work as well, I wouldn't be surprised if the intercom system filtered used a band pass filter between 150 and 5000 Hz. Probably at the microphones.Time to experiment anyway. I would begin by finding out the actual pass band of the system and the microphones/headphones.

author: Hieronymus
author_rank: Newbie
timestamp: April 27, 2016, 03:12:59 pm
content: Hi,thanks for the replies.I'm not sure what the highest or lowest frequency is that will propagate through the existing system.Any way how I can test this "easily"? The audio quality is very good, so I assume it's not that narrowbanded.My main question is: how do I add that digital signal to my audio? what are the concerns/trade-offs. Is this the same if I choose a low signal (like the 100Hz or DTMF signals) or a high signal (38kHz)?I'll post the schematics tomorrow to give you a better idea.

author: Fungus
author_rank: Super Contributor
timestamp: April 27, 2016, 03:45:26 pm
content: Quote from: Hieronymus on April 27, 2016, 03:12:59 pmThrough a resistor.You have to figure out the right value for the resistor (or use a potentiometer) but it's usually that simple.If you want to do it really professionally and avoid impedance matching issues (ie. choosing the wrong resistor) you can feed your two audio signals into op-amps then mix the op-amp outputs via. resistors. This ensures the signals being mixed are similar to each other.ie. A two-channel version of this:http://www.circuitstoday.com/multi-channel-audio-mixer-using-lm3900(throw away the two channels you don't want).

author: Jeroen3
author_rank: Super Contributor
timestamp: April 27, 2016, 03:53:37 pm
content: Quote from: Hieronymus on April 27, 2016, 03:12:59 pmWith a frequency generator (could be an App) and a spectrum analyzer (could also be an App).But if audio is very good, and it's digitally processed, it will be above 16 kHz samplerate. Allowing an theoretical maximum tone of 8 KHz.Another approach would be to use pulse dialing.https://en.wikipedia.org/wiki/Pulse_dialingBut this is very audible.

author: Fungus
author_rank: Super Contributor
timestamp: April 27, 2016, 03:58:31 pm
content: Quote from: Jeroen3 on April 27, 2016, 03:53:37 pmI think it will be difficult to make it inaudible with these constraints.If anybody is listening then just make a pleasing sound like a DTMF tone instead of a horrible annoying noise. Problem solved.