The Journey To Full Digitalization
Why go digital
Establishing a fully digital connection between the processor and the amplifier or active speaker system yields a range of significant advantages that enhance the overall audio experience. One key benefit is a notably reduced noise floor. By transmitting audio signals digitally, the inherent noise associated with analog connections is minimized, leading to a cleaner and more pristine audio output.
Furthermore, restricting the number of digital-to-analog (D-A) and analog-to-digital (A-D) conversions in the audio path results in fewer errors or distortions introduced to the audio signal. This preservation of the digital audio stream from the processor to the amplifier or active speaker system ensures a fidelity true to the recording, complete with the full dynamic range. This approach allows listeners to experience music or movie soundtracks as intended by the artist.
From Point-to-Point to Network Digital Audio Standards
Over the years, various types of digital methods for the transmission of audio have been invented.
Using balanced XLR interconnects, AES/EBU (AES3) is a two-channel standard for digital connections. It is an open standard that has been well established in the audio industry. AES/EBU connections are point-to-point with fixed routing. In these systems, each channel is linked to the signal path so the wiring patching design needs to be done carefully and if any changes should be made, it requires heavy hardware matrixing to achieve it when many channels are involved.
Over time, multiple solutions have been developed using the ethernet, and more specifically network technologies to carry Digital audio as offering increased bandwidth, thus allowing for more information to be sent with a reduced amount of cables. This also increases the number of audio channels that can be delivered with minimal cabling. Unlike traditional digital audio point-to-point wiring, digital signals transmitted over ethernet are also more efficient and flexible to handle, allowing point-to-multipoint transmission. It increases design possibilities in terms of functional capability and room installation topology system design options. The routing of audio within the system is not fixed, like with analog or digital point-to-point audio cabling, but software defined with streams of audio channels being broadcasted. With this, AoIP enables system design options not possible with traditional audio.
Network Digital Audio Standards
During the 1990s, technology primarily found its application in broadcasting and sports events due to its capability to transmit signals over long distances. However, over the past decade, a noticeable trend has emerged among high-end consumer speaker and amplifier manufacturers who are integrating Audio over Internet Protocol (AoIP) into their products. Various proprietary systems have been introduced to transport high-quality audio over IP, utilizing different protocols like Dante, LiveWire, or AVB. These protocols are widely adopted in professional audio and are featured in enabled speakers from various manufacturers. The challenge lies in the lack of interoperability among these protocols, leading to device incompatibility and user inconvenience.
To address these multiple issues in existing digital transmission systems, the AES67 standard was introduced.
Challenges in designing a processor for AoIP applications
StormAudio’s digital solutions first began with AES/EBU and AVB options for the ISP Elite platform. AVB (Audio Video Bridging) is a network protocol designed to run over ethernet that was developed by the IEEE (Institute of Electrical and Electronics Engineers). This connection allows routing of audio and video with speed and accuracy, but requires specific hardware elements in routers, which limits the practical use cases.
Later, motivated by the need to address this demand for a more versatile AoIP processor in the market, StormAudio embarked on an extensive research journey to find a suitable AES67 solution for enabling it on the same platform, as well to provide the most flexible options for installers and customers. This process involved a meticulous two-year survey of various digital implementations in the industry and an additional year on the development phase to create a solution.
An impetus behind the integration of AES67 is the employment within the digital cinema market, specifically with the Dolby Atmos Connect that uses AES67. Another is the interoperability of the standard, especially with Dante that has become one of the most popular connection standards in audio equipment across the professional and consumer industries. But AES67 alone does not offer a straightforward method for “device discovery” within the network.
Consequently, other available options had to be analyzed for device discovery. Ravenna invented a discovery layer that allows for easy identification of devices within a network for linking and patching streams. We chose the most suitable solution integrating Ravenna coming from Merging Technology, one of the most reliable on the market. This made StormAudio’s ISP digital solution unique on the market, being able to manage 32 channels input and output over AES67.
Advanced Clocking and Versatility for Digital Applications
Our team faced another significant challenge: making our new hardware module capable of accommodating a fully asynchronous clocking structure. As the audio network would have to handle different sampling rates, it is a challenge to ensure the correct interface between them. The primary objective was to enable different sampling rates between the network stream and the ISP’s internal sampling rate. It resulted in a game-changing achievement, as StormAudio’s digital solutions can now seamlessly support both 48kHz and 96kHz network sampling rates.
To achieve this feat, a new ASRC (Asynchronous Sampling Rate Conversion) DSP library had to be developed and specifically tailored to handle multiple sampling rates. As a result, the ISP platform proudly stands as the only one on the market that offers complete asynchronous clocking for different sampling rates on both the network inputs/outputs and internal processing, independently. This intricate and sophisticated mechanism design ensures high sound quality and utmost stability.
Harness the Power of Digital Audio Technology: Elevating Sound Quality and Control with ISP Evo
To streamline and keep digital installations cost effective, the ISP Evo was introduced. Based on the award-winning ISP platform, the ISP Evo is the first fully digital processor with 32 channel I/O AoIP on the market, and delivers all the signature StormAudio qualities and features users have come to expect, including great sound, functionality and stability.
Although the ISP Evo comes in a more space-efficient compact form, the device still stays true to our StormAudio distinctive attributes: high-quality, modular, and flexible. Embodying StormAudio’s proprietary modular platform, it offers the choice between a 20 – or 32 – channel configuration on AES or AoIP networks.
The benefits of ISP Evo are evident in its ability to optimize sound quality, offer precise control, and provide the utmost flexibility in configuring complex audio configurations. The ISP Evo guarantees exceptional audio quality as it enables digital audio systems thereby effectively reducing noise and interference typically found in analog systems. It positions professionals as frontrunners in the industry by providing cutting-edge audio solutions, incorporating advanced digital signal processing. Its compatibility with AES67/Dante and AES/EBU protocols enables smooth integration into current setups, both for input and output, granting the flexibility to create custom-designed audio systems for residential properties.
