I have been designing Analogue Audio Intercoms and now want to start with Digital Intercoms. I now have the ability to produce PCM or IIS data but I do not know how can I transfer that over RS485 so that I can run the Intercom system? Idea is to send RS485 Audio Data following with some Intercom Specific Commands to control the system.

Any help or pointer would be appreciated. Ideal solution would be full duplex solution over 2 wires.

 

RS485 is just a physical wire transmission standard. Some sort of Time division multiplexed (TDM) format will be need to mix multiple data streams into a data line(s) that can be bit-clocked over a RS485 link. There isn't a real standard for this TDM interface like there are standards for inter-chip interfaces.

https://gab.wallawalla.edu/~larry.aamodt/engr432/cirrus_logic_TDM_AN301.pdf
https://patents.google.com/patent/US5483528A/en

 

In digital audio products RS-485 transmitters and receivers are used for digital audio data all the time.

 

I would recommend S/PDIF for the following reasons:
automatic clock recovery
encodes more than 1 channel
encodes low-speed datastream
ICs already available (WM8804/WM8805)
Industry standard data format (also called AES3)
Data can be sent over RS485 hardware

 

In digital audio products RS-485 transmitters and receivers are used for digital audio data all the time.

Yes, but that's the easy part. AES3 TDM protocol sounds like a good framework to start with.
http://www.ijscience.org/download/IJS-2-12-143-146.pdf

 

AES3 TDM protocol

So, are you suggesting PCM data signal will go into a chip which converts the signal into AES or S/PDIF first then transfer it over RS485?

 

So, are you suggesting PCM data signal will go into a chip which converts the signal into AES or S/PDIF first then transfer it over RS485?

Yes, or you could transformer couple it, because S/PDIF is Manchester coded so has no DC offset.

 

Is there some specific reason you want to use RS-485 to transfer the information?

 

Is there some specific reason you want to use RS-485 to transfer the information?

It is something which I have used a lot and it can transfer data over long cables which is what is the nature of the intercom, I make.

 

So, are you suggesting PCM data signal will go into a chip which converts the signal into AES or S/PDIF first then transfer it over RS485?

Yes, this link I post before details a possible system example: http://www.ijscience.org/download/IJS-2-12-143-146.pdf

https://www.cirrus.com/products/cs8406/
The CS8406 is a digital audio transmitter that enables consumer and professional audio products to exchange 192 kHz S/PDIF and AES/EBU audio data. The CS8406 accepts and encodes audio and digital data, which is then multiplexed, encoded and driven onto a cable/optical transmission interface.

https://www.cirrus.com/products/cs8416/
The CS8416 is a digital audio receiver that supports sample rates up to 192 kHz. It enables consumer and professional audio products to exchange S/PDIF and AES/EBU audio data. An 8:2 input multiplexer allows for up to eight channels of digital audio input data.

4. System Test
Using two devices that shown as Figure 6 to test the transmission distance, during the test we use two
terminals to connect as transmitter and receiver. The result of the experiment is: without electrical
level transition, transmit with coaxial line, transmission distance is longer than 5m;using 1mmPOF
optical fiber for transmission, the distance is longer than 50m;when RS485 electrical level is
transmitted by twisted-pair, the distance is shorter than 40m. So no matter adopting optical fiber or
twisted-pair can both satisfy the need of long distance transmission of audio signal 7 on multi-load
optical measuring vehicle. Otherwise the sampling rate of audio signal in the system is 32KHz, it can
recover the voice of the talkers, meet the need of intercom system at the same time

To guarantee the audio receive format isI2S. According to protocol AES3, the data flow
output rate is 64*32KHz=2.048Mb/s.

This speed is what will limit RS485 range. If I had the task today I would likely use a 32-bit controller that has SPI with audio protocol functionality instead of the encoder/receiver chip of the link design to reduce the Rs485 bit-rate to something more in line with intercom requirements.

https://ww1.microchip.com/downloads/en/DeviceDoc/61106G.pdf

23.4 AUDIO PROTOCOL INTERFACE MODE
The SPI module can be interfaced to most codec devices available today to provide PIC32 microcontroller-based audio solutions. The SPI module provides support to the audio protocol functionality via four standard I/O pins. The four pins that make up the audio protocol interface modes are: • SDIx: Serial Data Input for receiving sample digital audio data (ADCDAT) • SDOx: Serial Data Output for transmitting digital audio data (DACDAT) • SCKx: Serial Clock, also known as bit clock (BCLK) • SSx: Left/Right Channel Clock (LRCK) BCLK provides the clock required to drive the data out or into the module, while LRCK provides the
synchronization of the frame based on the protocol mode selected.

 

How do you avoid acoustical feedback howling in your full duplex intercom? Usually half-duplex is used with a push-to-talk switch or sound-levels do the switching.
Years ago I worked with a full duplex intercom that made a "model" of the rooms and long distance path using the shhhhhh of pink noise. Then it used a digital echo-canceller circuit to cancel feedback and echo sounds. The model got messed up if many people in the room were added subtracted or opened or closed a door during the intercom call.

 

How do you avoid acoustical feedback howling in your full duplex intercom? Usually half-duplex is used with a push-to-talk switch or sound-levels do the switching.
Years ago I worked with a full duplex intercom that made a "model" of the rooms and long distance path using the shhhhhh of pink noise. Then it used a digital echo-canceller circuit to cancel feedback and echo sounds. The model got messed up if many people in the room were added subtracted or opened or closed a door during the intercom call.

https://timscott.net/hrblogging/the-larsen-effect/
Dual mic's is one old analog solution. One for background audio only (shielded in some way from the voice mic) and the other for both background and local speaking. Subtract for local mic audio only.
There are commercial products like my old feedback destroyer I use in the home theater for bass equalization in manual parametric EQ mode instead of feedback suppression.


In the Digital realm there are a few more ways.
https://ww1.microchip.com/downloads/en//softwarelibrary/dspic dsc acoustic echo cancel/70134f.pdf

 

The digital acoustic echo canceller is what I worked with 25 years ago.