Audiobus How A2B Enables More Complex Data and Audio Systems

Further developments of the A²B bus open up new possibilities for the transmission of data and audio signals. Intelligent buildings, halls, rooms and smart homes benefit from this. The article shows application examples and how the technology behind them simplifies wiring with little hardware and software effort.

Loudspeaker systems are required in many public buildings, factories or offices. These consist of microphones, amplifiers, loudspeakers and accessories. Such systems replace traditional alarm systems, for example. Music transmission, voice alarms, broadcasting and intercom systems can be integrated into these systems.

Ethernet networks are used for simple audio and control devices. In addition, 100 V cables are also used. However, these cables only transmit a single audio signal in one direction. Data communication cannot be added.

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This is where the Audiobus A²B comes into its own, connecting components such as loudspeakers, microphones, microphone units, control panels and sensors with simple daisy-chain cabling. The digital audio bus works bidirectionally with a high bandwidth. It transmits control signals as well as clock information and power via a 2-wire UTP cable (unshielded twisted pair cable) over a length of up to 98 feet between the individual nodes and up to 984 feet over the entire bus length.

A²B has proven itself in both automotive and communication applications and can be used for many commercial and industrial purposes.

When connecting an A²B transceiver to a device, users have 64 bidirectional audio channels (32 downstream and 32 upstream) that can be coupled with I²C/SPI/GPIO communication for peripheral devices at the endpoint. No more complex time slot stacks need to be developed or used.

The time-synchronized high-speed bus, which transmits I²S/TDM together with I²C, SPI and GPIO signals at a data rate of 50 Mbit/s, offers a latency between any two nodes of 50 μs. Up to 17 nodes (including the main node) can be connected to the bus via a UTP cable.

Communication takes place from node to node, which means that each node can send 32 audio channels to every other node if 48 kHz and 16 bits are selected. The data size can be set for different requirements, with the data rate adjustable between 1.5 and 192 kHz. Lower bandwidths and data rates result in more available channels.

Power Supply Integrated in the Bus

Another advantage is the power supply integrated into the bus, which can supply power of up to 50 W over the entire bus. This means that nodes can be supplied with phantom power via the bus without the need for an additional local power supply. Speaker nodes with low output power can also be supplied via the A²B bus. However, external power supplies are required for higher power consumption.

Figure 1 illustrates the flexibility of the A²B bus, which simplifies the design and configuration of the system. As the transceiver already contains many blocks and interfaces, it can often be operated without a separate microcontroller.

As shown, the simplest node is a microphone array with the AD2437 transceiver. Up to four PDM (pulse density modulation) microphones can be connected. Such microphone arrays localize around noise sources, perform noise suppression or detect the direction from which a noise is coming. In this case, it is possible to extract the audio source from the adjacent background noise.

Due to the very low latency of the bus, this array does not necessarily have to be present on one node. It can easily be distributed to different nodes at different locations within a room. Power is supplied via the bus, as the node has a very low power consumption. This makes the solution very compact and easy to install. In an example application, the node had a size of 1.38 inches × 0.75 inches, including cabling and bus power supply.

Connecting A Class D Or Audio Amplifier

For more complex requirements, a separate Class D or other power amplifier can be connected to the AD2437 via the I²S output. An audio codec can also be integrated. As audio must not normally be interrupted, the I²C interface is used to set up the power amplifier or codec in parallel.

This configuration can be used to create a simple, bus-powered intercom system. The AD2437 also has GPIOs, some of which are equipped with pulse width modulation (PWM). These can be used as keyport inputs for interaction with the host. The host receives an interrupt signal when a button is pressed, after which communication can be established.

The PWM output is used to indicate whether communication is active, to control LEDs that indicate the connection status and to display any other information required. This simple but effective function does not require a separate microcontroller, which reduces the software effort for the entire system.

Intercom systems with a graphical user interface (GUI) use a microcontroller that receives data via the SPI interface. The maximum speed of the SPI bus is10 MBit/s.

External power supply units are sufficient for pure loudspeaker nodes that are located in ceilings or other parts of buildings, for example, and require a higher output power. The bus does not have to be powered at the host node, but can be powered at any point within any intermediate node in order to reduce the current load in the lines.

Applications of the Audio Bus in Care

There are numerous other areas of application in which the functions of the A²B audio bus offer a decisive advantage over previously used technologies.

A nurse call/notification system is based on an Ethernet configuration. In this case, a room controller is connected via Ethernet, but the connection to the patient terminals is made via the A²B bus. In this configuration, all necessary audio, data and power signals are transmitted to 16 beds in a room via a UTP cable.

Each terminal can easily be equipped with a small microcontroller to provide patients with different audio channels. A display shows the selected channel, time and alarm status. An alarm button can be connected directly to one of the GPIO ports to trigger an interrupt on the room controller. If patients are restricted in their freedom of movement, a selected end device can also transmit microphone signals immediately to the room controller. A selected keyword triggers an alarm.

The system configuration can be changed easily. Terminals are added and removed from the network via the host's plug-and-play stack. To maintain the daisy chain configuration on remote terminals, a small adapter can bridge the missing node. If communication is interrupted, the diagnostic function reports an error.

Beamforming: Distribute Audio Signal to Several Rooms

Counter intercom systems also benefit from the very low latency and fully synchronous behavior of the A²B bus. Multiple microphones can be placed in different positions to achieve clear channel separation of the person speaking in front of the counter using beamforming. All background noise from neighboring people is suppressed.

In this way, crystal-clear conversation can be ensured between any acoustic barriers such as glass walls. Such solutions are typically used for counters, isolation areas in hospitals or clean room facilities. Similarly, intercom systems in prisons can transmit all microphone conversations to a host system and many audio channels, such as different radio channels, to each cell.

Another example of the use of beamforming is conference systems where many microphones are placed in a meeting room. Speech-to-text systems require a clear separation of the audio signals of different speakers. Due to the low latency, all microphone channels can be transmitted simultaneously to the host controller or the DSP, which calculates the different beam positions.

An alternative would be a table microphone or headset for each seat that is connected to an interpreting system. Here, all participants receive personalized audio data that is automatically translated into their native language. By using a lower audio bandwidth, a very high number of channels is supported.

This means that all microphone signals are available at the host and can be easily prioritized, either to the signal with the highest volume or in a system-controlled manner where the microphone is only enabled for selected people. In this case, all terminals are supplied with power via the bus without the need for a local power supply at the tables.

Audio distribution to many rooms is also desirable in home automation systems. This way, users can hear the doorbell anywhere in the house, even if they are listening to their favorite hi-fi playlist in the bathroom. The ability to stream audio to anywhere in the house is also beneficial, as is microphone presence to control the house with voice commands. Compared to a wireless connection via WLAN, a wired connection is more reliable and reduces the load on wireless data traffic.

A²B is also suitable for professional audio systems, sound recording studios and live stage installations and enables a simple audio connection with existing cable standards such as CAT.-5 or XLR.

Functionality of the A2B Bus

The A²B bus sends superframes via the bus in a 48 kHz clock. The data is transmitted 1,024 times faster, resulting in a transmission frequency of 49.152 MHz on the bus. The superframe consists of two parts: an upstream and a downstream, whereby the start is initiated by a sync control frame and a sync response frame. All I²S/TDM data, I²C data, GPIO and interrupt information is inserted into the downstream/upstream slots.

The host generates the clock signal and all nodes synchronize with this clock signal. This ensures that the system is synchronized at all times. A preamble in the synchronization control frame ensures that all nodes are synchronized and can pass this clock on to peripheral devices. This has the advantage that the entire audio chain does not require any additional clocks, local oscillators or asynchronous sample rate converters.

At system startup, the host processor, which is directly connected to the first transceiver chip, is set as the main node. The host provides a stable signal of 48 kHz and the transceiver synchronizes its phase locked loop (PLL) to this signal. After the main node has been set up, the sub-nodes are activated one after the other.

Setting Up the System With SigmaStudio+

The SigmaStudio+ software supports the entire system setup, including the configuration of audio channels and nodes. It is a graphical programming, diagnostic and tuning tool that allows the developer to create a graphical user interface for an A²B network by integrating peripherals such as audio codecs, Class D amplifiers, etc. Software stacks are available for Linux along with a plug-and-play stack that allows nodes to be added or removed from the bus on the fly.

A²B is supported by evaluation modules for main or sub-nodes with XLR or RJ45 connectors including power transmission.

Conclusion: Further improvements to the new A²B transceivers, such as a longer cable range and more bus power, open up a wide range of possibilities for different applications, especially when connecting multiple audio channels to control data via simple cabling.

If the devices connected to the bus are a mixture of simple and complex nodes, the bus enables cost-effective hardware implementation even for less complex devices. Even in non-audio applications, such as precisely synchronized sensor networks, A²B can provide significant simplification. (kr)

*Roland Prager works as a Staff Field Applications Engineer at Analog Devices in Wilmington / USA.

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