AES67 Standard Device.
Power By USB Type-C 5V@300mA.
Input: AES67 Network HD audio. 4x Channels, 24-bit, sample rates up to 96 kHz (48 kHz typical). Support Dante Network controller and Dante virtual sound card.
Output: AES3 Standard digital audio. 4x Channels, 24-bit, sample rates up to 96 kHz (48 kHz typical).
The SUPERCAN AES67 to AES3 converter is a bidirectional bridge that lets you patch IP-audio networks — AES67, Dante, RAVENNA and Livewire — into legacy XLR / AES3 broadcast infrastructure without latency penalty or sample-rate mismatch. Inside the chassis, an AES67-compliant network port (SMPTE ST 2110-30 profile compatible) receives RTP audio streams clocked to PTPv2 (IEEE 1588 Precision Time Protocol v2) grandmaster sync. The streams are decoded, word-clock-aligned, then re-emitted on the AES3 side as balanced 110 Ω differential audio over XLR.
The converter supports sample rates of 44.1 kHz, 48 kHz and 96 kHz, and bit depths of 16-bit, 20-bit and 24-bit — covering broadcast-standard 48 kHz / 24-bit production as well as high-resolution 96 kHz / 24-bit mastering. End-to-end latency through both stages of conversion is under 1 ms, low enough for in-ear monitor (IEM) chains, live theater foldback, and broadcast intercom routing where round-trip delay budget is tight.
AES67, Dante and RAVENNA all derive their network clock from PTPv2. When you bring an IP-audio stream into the AES3 world, the synchronization reference has to carry through — otherwise downstream digital consoles and routers will reject the AES3 frames as clock-slipped or word-clock-misaligned. The SUPERCAN converter exposes the recovered PTPv2 clock on a BNC word-clock output, so the rest of your AES3 / S/PDIF chain can lock to the same PTPv2 grandmaster as the IP-audio side. AES3 vs AES67 sample-rate conversion (e.g. 48 kHz IP audio → 96 kHz AES3 archive) is handled on the same hardware path.
End-to-end latency through both stages of conversion is under 1 ms — low enough for in-ear monitor (IEM) chains, live theater foldback, and broadcast intercom routing where round-trip delay budget is tight.
It is a bidirectional bridge between IP-audio networks — AES67, Dante, RAVENNA and Livewire — and legacy XLR / AES3 infrastructure. The AES67-compliant network port is SMPTE ST 2110-30 profile compatible and receives RTP audio streams clocked to PTPv2 (IEEE 1588 Precision Time Protocol v2) grandmaster sync.
Sample rates of 44.1 kHz, 48 kHz and 96 kHz, and bit depths of 16-bit, 20-bit and 24-bit — covering broadcast-standard 48 kHz / 24-bit production as well as high-resolution 96 kHz / 24-bit mastering.
Four channels in and four channels out. The AES67 network input carries 4 channels at 24-bit / 48 kHz, and the AES3 side re-emits 4 channels at 24-bit / 48 kHz as balanced 110 Ω differential audio over XLR.
It exposes the recovered PTPv2 clock on a BNC word-clock output, so the rest of the AES3 / S/PDIF chain locks to the same PTPv2 grandmaster as the IP-audio side. Without that shared reference, downstream digital consoles and routers reject the AES3 frames as clock-slipped or word-clock-misaligned.
Yes. The AES67 network input supports a Dante network controller and Dante virtual sound card, so a Dante-managed system can feed the converter directly while the AES3 output stays locked to the same PTPv2 grandmaster.
From USB Type-C, at 5 V / 300 mA.
Go deeper with AES67