Enabling Direct Monitoring of Audio-Over-IP Streams
By Aki Mäkivirta, Research and Development Director, Genelec.
Audio-over-IP technology opens new possibilities for building audio infrastructure. Standard ethernet infrastructure is becoming the medium for audio that will be able to support any resolution, sample rate, or format of professional audio data. The Genelec 8430 two-way smart active monitor enables direct monitoring of the AES67 formatted digital audio stream directly.
Using Internet Protocol (IP) network to pass the highest quality audio (and video) signals with low latency carries a great potential allowing flexible connectivity from any source device to any destination device, and the ability to pass any audio format, sample rate, and word length. This implies a change in the way professional quality level audio (and video) systems have been built up until today.
Audio-over-IP technology divides the samples of digital audio into packets and transports these packets to the destination devices over CAT5 or CAT6 cables, using IP networking protocols. Several protocols are needed, each responsible of particular details of the process. As these protocols handle data, they become capable of passing any format of audio sample data, irrespective of the sample rate, audio bandwidth, format of encoding, or any other such aspect of presentation that usually has been of great significance in terms of enabling system connectivity.
While the network itself does not limit the presentation of audio data, the significant topic in professional audio is audio-over-IP interoperability. This refers to the ability of devices from different manufacturers to confer with each other. This is where the AES67 standard comes in.
Traditionally audio interconnections accomplish interoperability with standards like the balanced XLR output and input for conventional analogue audio, and the AES/EBU interface for the conventional digital audio.
As IP networking systems must utilise a large collection of protocols and settings for each protocol in their communication, interoperability has not been a self-evident result of using IP audio networking. In fact, the numerous audio-over-IP systems on the market have contained enough small differences to render interoperability difficult to achieve.
A standard for audio-over-IP interoperability has been developed by the Audio Engineering Society and was published first in September 2013. This AES67 standard has been designed to allow interoperability between IP-based systems designed for audio transport with high dynamic range, low end-to-end latency, very high clock accuracy, and without a technical limit to the maximum number of audio channels the system can support. Examples of such systems include RAVENNA, Livewire, Q-LAN and Dante.
The AES67 is an Internet Protocol based solution and it uses protocols at or above layer three in the sever-layer Open System Interconnect (OSI) reference model. This makes AES67 fully compatible with existing cable-based IP networking hardware environments (ethernet). Audio-over-IP can transport on virtually any local area network (LAN) and benefits from the continuous increase in the speed of these networks. Today, the device connections operate at 1Gbit/s while the faster switch-to-switch connections can exceed 10Gbit/s.
Although ethernet is usually the underlying data link layer, the Internet Protocol is infrastructure-agnostic and can be used on virtually any network technology.
To achieve interoperability, AES67 addresses the following tasks in audio networking:
- Device clock synchronisation pulls all audio devices in synch across the network.
- Media clock management defines the audio clocks and how they relate a common clock reference.
- Transport moves audio data across the network with sufficient priority.
- Encoding and streaming describes the formats of audio in the packets of a stream.
- Stream description communicates the data needed to establish a connection, specifying network addressing, encoding and origination.
- Connection management communicates the data needed to establish and maintain an audio stream connection.
- Industry-first in audio-over-IP monitoring.
Six years ago, Genelec publicly demonstrated its first audio-over-IP loudspeaker system at the Integrated Systems Europe exhibition. This system was selected as ‘Pick of ISE 2010’, one of the best innovations presented at the show.
In February 2016 Genelec launched the 8430 Smart Active Monitor, the first audio monitor to allow direct monitoring of audio-over-IP signals.
The 8430 combines various connectivity options. The AES67 signal input is housed in an XLR-integrated RJ45 connector, enabling audio-over-IP signal connection. The AES67 input supports 44.1 kHz to 96 kHz sample rates, as well as 16-, 24- and 32-bit word lengths. With the accurate clock synchronisation to a network-attached Precision Time Protocol grandmaster clock source, this enables accurate monitoring of high resolution audio signals.
The 8430 also supports monitoring of professional analogue signals using a balanced XLR connector. The full 25dBu professional audio signal level range is supported.
The digital audio in an audio-over-IP stream may contain any number of channels. On an IP network, all audio streams are visible to all monitors. The physical cabling has absolutely no significance in determining what audio channel goes where in the system. The cabling in an IP network system always follows the same principle; the monitor connects to an IP switch device using one IP network cable. All devices connect via IP switches. There is nothing further to know about the cabling for audio-over-IP speakers, except that the maximum length of the cable is 100 meters from the switch device to the monitor. Often this is ample, but can be extended using another switch device. After pulling these physical cables, the connectivity resides entirely in the software configuration.
The routing of audio in the network is no longer dependent on cable routing. Connecting audio-over-IP devices is simple. You attach one cable to the network and you are done. All devices are accessible and visible through the network and when access to several configuration user interfaces is needed, all of these are available at the same computer display.
Audio-over-IP is attractive to audio infrastructure designers, because:
- More flexibility in cabling, system design and building.
- One network supports any audio format.
- Efficiency of maintenance.
- Lower overall facility-level cost.
- Flexibility of connectivity.
- Easier setup and operation.
- Less limitations to improve product features and performance.
- Long term availability of technology with improving speed of networking.
Proven Acoustical Accuracy
As part of Genelec Smart Active Monitoring (SAM) Series, the 8430 shares the same electro-acoustic design features such as Genelec MDE and DCW technologies, a flow optimised reflex port, very low distortion of the acoustic output across audible frequencies, high SPL capacity, and a wide bandwidth, uncolored output on the acoustical axis as well into off-axis directions, delivered in a compact enclosure.
Being a member of the Smart Active Monitoring series, the 8430 uses the Genelec Loudspeaker Manager (GLM 2.0) control network and software which allows adjustments of all aspects of monitor settings.
Unicast and Multicast
The typical method of defining a connection between two IP devices is called unicast. This method transports the data packets across the network so there is only one transmitter and one receiver for the packets.
IP networking also supports multicasting. Multicasting implies one transmission is addressed to several receivers while the data packets are only sent once. Audio presentations are typically collections of several channels. While it is up to the system configurator, a convenient method of using an IP-based audio sample transport is to pack samples from all the channels of one presentation into one data packet. Then, multicasting a stream of these data packets across the network allows all monitors and subwoofers to listen to this transmission and pick applicable audio channels from the data stream. Because only one audio stream is sent on the network, multicast can reduce the data processing load in the network IP switch devices, enabling higher throughput. Both multicast and unicast are defined in AES67.
8430 monitors are calibrated individually at the factory to eliminate and any unit to unit differences.
The 8430 supports the smart SAM system calibration using the automated system measurement and calibration feature built into the Genelec Loudspeaker Manager software (GLM AutoCal) and is supported by room response measurement hardware. Genelec manufactures a room measurement microphone with calibrated frequency response and omnidirectional characteristics, enabling measurements for all professional reproduction configurations – including 3D systems. The automated measurement and calibration supports more than 30 monitors and subwoofers in one room and runs in a few minutes.
Supporting spot and wide area corrections at unlimited positions in a room, the GLM AutoCal system aligns at the listening position, the times of flight for all monitors, setting monitors at equal acoustic distance and aligns the reproduction levels enabling all devices to play at the same sound level – irrespective of their distance or other acoustic influences, and finally, equalises the frequency responses compensating for acoustic influences of the room.
The 8430 materials are recyclable, and the enclosures are manufactured of already recycled aluminium. The die cast aluminum enclosure offers driver protection against mechanical damage and electronic abuse, and versatile fixing features integrated in the enclosure structure offer ease of installation and long-term reliability.
The Genelec 8430 supports the interoperability standard AES67. It is also a fully developed Smart Active Monitor, enabling fast and accurate multichannel or immersive audio system calibration with detailed and individual compensation of the acoustic influences of the listening room. Genelec 8430 is an industry-first solution for monitoring audio-over-IP streams and integrates with facilities offering modern IP networks.