SOUNDCARDS

LoLa280 is a low profile professional linear (PCM) multichannel sound card based on the PCI Express bus interface. It is a reference sound card for professional audio recording systems for courtrooms, parliaments and other conference centers, and for audio logging and archiving systems for radios or safety organizations. LoLa280 features eight balanced analog inputs and two balanced analog outputs. Thanks to its hardware digital mixer, it allows mixing of inputs before recording from eight mono devices , monitoring of a mix of inputs on the two outputs, and mixing up to 8 playback channels to the two outputs. This card can can be used with Windows or Linux operating systems, with software applications based on standard driver interfaces such as WDM Kernel streaming, DirectSound, Wasapi, and ASIO for Windows, and Alsa for Linux.

• Developed for broadcast applictions
• Low profile PCI Express
• Eight balanced analog inputs and two balanced analog outputs; +24 dBu max level
• On board mixer:
  – 8 recording channels with pre-mixing of the input signals
  – 8 playback channels mixed to the two outputs
  – mixing and monitoring of the inputs to the two outputs
  – AGC on all the input
  – Adjustable input and output analog and digital gains
• Breakout cable for audio connectivity

CONFIGURATION

Bus/Format: PCI EXPRESS TM (PCIe ® ) x1 (compatible x2, x4, x8, x16, x32)
Size: 168 mm x 69 mm x 20 mm
Power requirements (+3.3 V / +12 V): 1,2 A / 0,22 A
Operating: temp / humidity (non-condensing) 0°C / +50°C • 5% / 90%
Storage: temp / humidity (non-condensing) -5°C / +70°C • 0% / 95%

INPUTS

Analog line inputs (mono): 8 balanced
Maximum input level/impedance: +24 dBu / > 10 kOhms
Adjustable input gain (manager):
– analog: from –90 dB to +18 dB (max sensitivity: 0dbfs for +6dBu input signal)
– digital: from +18 dB to +36 dB
Other inputs: 1 standard Word Clock input (up to 192 kHz)

* maximum sensitivity: 0 dBFs for +6 dBu input

Outputs

Analog line outputs (mono): 2 balanced
Maximum output level / impedance: +24 dBu / >100 ohms
Adjustable output gain (manager) analog: from –48 dB to 0 dB. digital: from -110 dB to +12 dB
Other outputs: 1 standard Word Clock output (up to 192 kHz). Stereo headphones (20 mW in 600 ohms) on mini jack (female TRS 3,5 mm)

Connectors

External connector: 50 pin SCSI
Sampling frequencies available: Programmable from 22,05 to 192 kHz
A/D and D/A converter resolution: 24 bits
Supported audio formats: PCM (16, 24, 32 bits, and 32 bits Float)

LoLa881 and LoLa881-SRC are professional linear (PCM) multichannel sound cards based on the PCI Express bus interface. They are designed for use in any professional system running under Windows or Linux, and requiring up to four AES/EBU I/O connections. They are reference cards in radio broadcast automation for applications such as multichannel audio ingest, play-out, and production.
LoLa881 and LoLa881-SRC cards feature eight input and output channels through AES/EBU connectivity, with the possibility to synchronize on an external clock (AES11, Word clock, black burst video). LoLa881-SRC can in addition handle asynchronous AES/EBU input signals thanks to its high quality hardware sample rate converters on each AES/EBU input. LoLa cards allows for very low latency operations, and can be used under Windows or Linux operating systems, with software applications based on standard driver interfaces such as WDM Kernel streaming, DirectSound, Wasapi, and ASIO for Windows, and Alsa for Linux.

• Developed for broadcast applications
• Four AES/EBU inputs/outputs, with hardware sample rate converter on each AES/EBU input for LoLa881-SRC.
• Low latency
• Support for Linux (Alsa driver) and Windows 32-bit and 64-bit (WDM Kernel streaming, DirectSound, Wasapi, ASIO)
• Breakout cable or breakout box (BOB16AES) with XLR connectors for audio connectivity

CONFIGURATION

Bus/Format: PCI EXPRESS™ (PCIe®) slot (x1, x4, x8, x16)
Size: 168 mm x 111 mm x 20 mm
Power requirements (+3.3 V / +12 V) : 0,6A / 0,01A (LoLa881) – 0,8A / 0,01A (LoLa881-SRC)
Operating temp / humidity (non-condensing): 0°C / +50°C • 5% / 90%
Storage: temp / humidity (non-condensing): -5°C / +70°C • 0% / 95%

INPUTS

Digital inputs (stereo): 4 AES/EBU (20 kHz to 192 kHz)
Hardware sample rate converters (LoLa881-SRC): conversion ratio 16:1 à 1:16, 20 kHz to 192 kHz, Dynamic: 144dB, THD+N: -140dB
Other inputs: AES/EBU Sync* (from 20KHz to 192 kHz managed by driver, h/w 216 kHz), Word clock (up to 192 kHz), Video sync (PAL, NTSC, 32000 Hz – 192000 Hz)
AES11 synchronization: Yes

OUTPUTS

Digital outputs (stereo): 4 AES/EBU (20KHz to 192 kHz)
Other outputs: Word Clock (up to 192 kHz)

CONNECTORS

External connector(s) : 1 x 26-pin SCSI MDR

AUDIO SPECIFICATIONS

Sampling frequency: Programmable from 32 to 192 kHz
Hardware mixer: Special development required – contact Digigram
Supported audio formats: PCM (16, 24, 32 bits, Float IEEE754)

DEVELOPMENT ENVIRONMENTS

Other management: ASIO, DirectSound, Wasapi, Alsa
OS supported: Windows and Windows Server, 32-bit and 64-bit.versions, Linux
Main on-board processing features: PCM, play+record

VX1222e is the reference multichannel PCM sound card designed for operating in continuous 24/7/365 use-environments as part of professional audio systems. It can be used under Windows or Linux operating systems, with software applications based on standard driver interfaces such as WDM Kernel streaming, DirectSound, Wasapi, and ASIO for Windows, and Alsa for Linux. It offers six stereo AES/EBU outputs and one stereo AES/EBU input, and can synchronise on an external clock (AES11, Word clock, black burst video).

• Developed for the broadcast industry
• High audio quality sound card that supports balanced analog and AES/EBU audio
• Interoperable with most of the third party software applications for audio production under Windows
• Twelve balanced analog outputs and two balanced analog inputs, +24 dBu max level
• Six stereo AES/EBU outputs and one stereo AES/EBU input with hardware sample rate converter
• Synchronisation inputs: AES11, word clock, black burst video
• Adjustable input and output digital gains
• On-board 3-band parametric EQ and Maximizer effects

Drivers
Windows (32-bit / 64-bit): WASAPI, ASIO, DirectSound, Digigram proprietary “np” SDK
Linux: ALSA

Configuration

Bus/Format: PCI EXPRESS™ (PCIe®) x1 (x2, x4, x8, x16, x32 compatible)
Size: 168 mm x 111 mm x 20 mm
Power requirements (+3.3 V / +12 V): 3 A / 0.4 A
Operating: temp / humidity (non-condensing): 0°C / +50°C • 5% / 90%
Storage: temp / humidity (non-condensing): -5°C / +70°C • 0% / 95%

Inputs

Analog line inputs (mono): 2 balanced
Maximum input level/ impedance: +24 dBu/ >10 kOhms
Digital input (stereo): 1 AES/EBU with hw Sample Rate Converter, 7.5:1 to 1:8, up to 192 kHz
Programmable input gain:
– analog: from –94.5dB to +15.5 dBÄ
– digital: from –110 dB to +18 dB
Other inputs: AES/EBU Sync (up to 192 kHz), Word clock (up to 96 kHz), LTC, Video
AES11 synchronization

Outputs

Analog line outputs (mono): 12 servo-balanced
Maximum output level / impedance: +24 dBu / <100 Ohms
Digital outputs (stereo): 6 AES/EBU, up to 192 kH
Programmable output gain:
– analog: from –86 dB to +24 dB
– digital: from –110 dB to +18 d
Other outputs: Word clock (up to 96 kHz)

Connectors

Internal connectors: Inter-board Sync
External connector: 68-pin SCSI MDR
Digigram accessories available: Breakout cable or 2U 19″ Breakout Box

Audio specifications

Sampling frequencies available: Programmable from 22.05 to 192 kHz
A/D and D/A converter resolution: 24 bits
Supported audio formats: PCM (8, 16, 24 bits), Float IEEE754

Audio Performance

Frequency response (record + play): 20 Hz–20 kHz: ±0.3 dB
Channel phase difference: 20/20kHz: <0.2°/2° Dynamic range (A-weighted): – analog In: >104 dB
– analog Out: >110 dB
THD + noise 1 kHz at –1 dBfs:
– analog In: <–96 dB
– analog Out: <–96 dB
Crosstalk (Analog in or out):
– 1 kHz at 24 dBu: <–100 dB
– 15 kHz at 24 dBu: <–90 dB

Development environments

Digigram management under Windows: np SDK
Other management under Windows: Wave (PCM & MPEG Laye II), WASAPI, ASIO, DirectSound, Alsa (all PCM)
OS supported: from Windows 7 and Windows server 2003 (32-bit and 64-bit versions), Linux
Main on-board processing features (with np SDK): PCM play & rec, MPEG Layers I & II play & rec, Layer III play, Float IEEE754,direct monitoring, real-time mixing, level adjustment, panning, crossfade, punch-in/punch-out, scrubbing, time-stretching, pitch-shifting, 3-band parametric equalizer, maximizer, format and frequency conversions

VX442e features four input and output channels and offers balanced analog and AES/EBU connectivity, with the possibility to synchronize on an external clock (AES11, Word clock, black burst video). It can be used under Windows or Linux operating systems, with software applications based on standard driver interfaces such as WDM Kernel streaming, DirectSound, Wasapi, and ASIO for Windows, and Alsa for Linux.

• Developed for the broadcast industry
• High audio quality sound card that supports balanced analog and AES/EBU audio
• Interoperable with most of the third party software applications for audio production under Windows
• Four balanced analog inputs and outputs, +24 dBu max level
• Two AES/EBU inputs/outputs, with hardware sample rate converter on each AES/EBU input.
• Adjustable input and output analog and digital gains
• On-board 3-band parametric EQ and Maximizer effects

Drivers
Windows (32-bit / 64-bit): WASAPI, ASIO, DirectSound, Digigram proprietary “np” SDK
Linux: ALSA

Configuration

Bus/Format: PCI EXPRESS™ (PCIe®) x1, (x2, x4, x8, x16, x32 compatible)
Size: 67.7 mm X 106.7 mm X 17.1 mm
Power requirements (+3.3V / +12V): 2.5 A / 0.1 A
Operating: temp / humidity (non-condensing) : 0°C/+50°C • 5%/90%
Storage: temp / humidity (non-condensing) : -5°C/+70°C • 0%/95%

Inputs

Analog line inputs (mono): 4 balanced
Maximum input level/ impedance: +24 dBu / >10 kOhms
Digital inputs (stereo): 2 AES/EBU with hw Sample Rate Converters, 7.5:1 to 1:8, up to 192 kHz
Programmable input gain: analog: from –94.5dB à +15.5 dB (maximum sensitivity: 0 dBFs for –15.5 dBu input), digital: from –110 dB à +18 dB
Other inputs: AES/EBU Sync (up to 192 kHz), Word clock (up to 96 kHz), LTC, Video
AES11 synchronization

Outputs

Analog line outputs (mono): 4 servo-balanced
Maximum output level / impedance: +24 dBu / <100 Ohms
Digital outputs (stereo): 2 AES/EBU, up to 192 kHz
Programmable output gain:
– analog: from –86 dB to +24 dB
– digital: from –110 dB to +18 dB
Other outputs: Word clock (up to 96 kHz)

Connectors

Internal connectors: Inter-board Sync
External connector: 68-pin SCSI MDR

Audio specifications

Sampling frequencies available: Programmable from 8 to 192 kHz
A/D and D/A converter resolution: 24 bits
Supported audio formats: PCM (8, 16, 24 bits), Float IEEE754

Audio Performance measured at Fs=48 kHz

Frequency response (record + play): 20 Hz–20 kHz: ±0.2 dB
Channel phase difference: 20/20kHz: <0.2°/2° Dynamic range (A-weighted) : – analog In: >104 dB
– analog Out: >104 dB
THD + noise 1 kHz at –1 dBfs :
– analog In: <–97 dB
– analog Out: <–94 dB
Crosstalk (Analog in or out):
– 1 kHz at 24 dBu: <–100 dB
– 15 kHz at 24 dBu: <–85 dB

VX882e is a professional linear (PCM) multichannel sound card based on the PCI Express bus interface. It is designed for use in any professional system running under Windows or Linux. It is a reference card in radio broadcast automation for applications such as automated recording and play-out and multichannel production.
VX882e features eight input and output channels and offers balanced analog and AES/EBU connectivity, with the possibility to synchronize on an external clock (AES11, Word clock, black burst video). It can be used under Windows or Linux operating systems, with software applications based on standard driver interfaces such as WDM Kernel streaming, DirectSound, Wasapi, and ASIO for Windows, and Alsa for Linux.

• Developed for the broadcast industry
• High audio quality sound card that supports balanced analog and AES/EBU audio
• Interoperable with most of the third party software applications for audio production under Windows
• Eight balanced analog inputs and outputs, +24 dBu max level
• Four AES/EBU inputs/outputs, with hardware sample rate converter on each AES/EBU input
• Adjustable input and output analog and digital gains
• On-board 3-band parametric EQ and Maximizer effects

Drivers
Windows (32-bit / 64-bit): WASAPI, ASIO, DirectSound, Digigram proprietary “np” SDK
Linux: ALSA

Configuration

Bus/Format: PCI EXPRESS™ (PCIe®) x1, (x2, x4, x8, x16, x32 compatible)
Size: 168 mm x 111 mm x 20 mm
Power requirements (+3.3V / +12V): 2.5A / 0.5A
Operating: temp / humidity (non-condensing): 0°C/+50°C • 5%/90%
Storage: temp / humidity (non-condensing): -5°C/+70°C • 0%/95%

Inputs

Analog line inputs (mono): 8 balanced*
Maximum input level/ impedance: +24 dBu/ >10 kW
Digital inputs (stereo): 4 AES/EBU** with hw Sample Rate Converters, 7.5:1 to 1:8, up to 192 kHz
Programmable input gain: analog: from –94.5dB à +15.5 dB (maximum sensitivity: 0 dBFs for –15.5 dBu input), digital:from –110 dB à +18 dB
Other inputs: AES/EBU Sync (up to 192 kHz), Word clock (up to 96 kHz), LTC, Video
AES11 synchronization: Yes

Outputs

Analog line outputs (mono): 8 servo-balanced***
Maximum output level / impedance: +24 dBu / <100 W
Digital outputs (stereo): 4 AES/EBU**, up to 192 kHz
Programmable output gain: analog: from –86 dB to +24 dB / digital: from –110 dB to +18 dB
Other outputs: Word clock (up to 96 kHz)

Connectors

Internal connectors: Inter-board Sync
External connector: 68-pin SCSI MDR
Digigram accessories available: Breakout cable or 2U 19″ Breakout Box

Audio specifications

Sampling frequencies available: Programmable from 8 to 192 kHz
A/D and D/A converter resolution: 24 bits
Supported audio formats: PCM (8, 16, 24 bits), Float IEEE754

Audio Performance measured at Fs=48 kHz

Frequency response (record + play): 20 Hz–20 kHz: ±0.2 dB
Channel phase difference: 20/20kHz: <0.2°/2° Dynamic range (A-weighted) : Analog In: >104 dB / Analog Out: >104 dB
THD + noise 1 kHz at –1 dBfs : Analog In: <–97 dB / Analog Out: <–94 dB
Crosstalk (Analog in or out): 1 kHz at 24 dBu: <–100 dB / 15 kHz at 24 dBu: <–85 dB

Development environments

Digigram management: np SDK (PCM only)
Other management: Wave, ASIO, DirectSound,WASAPI, ALSA
OS supported: Windows 32bit and 64 bit versions, Linux
Main on-board processing features (available through the Digigram np SDK): PCM play, rec, Float IEEE754, direct monitoring, real-time mixing, level adjustment, panning, cross-fade, punch-in/punch-out, scrubbing

* can be used with unbalanced signals
** can be used as S/PDIF interface as well
*** electronically servo-balanced outputs provide automatic level adjustment to accommodate either balanced or unbalanced lines

Built on a long tradition of uncompromising equipment, CANCUN is a high-performance sound card that embeds native USB Audio 2.0 streaming, mixing and processing in an ultra-robust yet stylish casing. This professional toolbox is ideal for any Broadcast or live event application on a PC or Mac. On-the-go technicians, reporters and mobile production talents will enjoy its embedded mixer and seamless integration to their favorite editor or radio automation, particularly when coupled to audio-over-IP software on their laptop such as IQOYA V*MOTE.
Production engineers can quickly set up high-quality Record/Play on a PC or Mac using CANCUN’s LED-lighted touch panel and monitoring from zero-latency headphone mixer. Live sound engineers appreciate CANCUN’s AES/EBU and analog interfaces, which Play/Record up to 4+4 channels simultaneously. During sound system set up, operators can count on the card’s linearity and the settings control enabled by main audio measurement applications.

• Best-in-class audio performances:
  – excellent Mic preamp (+55 dB gain, -105 dB THD+N, -126 dB EIN)
  – professional analog level max +25 dBu
  – guaranteed low latency (< 4 mS)- excellent Mic preamp (+55 dB gain, -105 dB THD+N, -126 dB EIN)
  – professional analog level max +25 dBu
  – guaranteed low latency (< 4 mS)
• Zero latency embedded mixer
• A mobile toolbox:
  – simultaneous AES/EBU and analog connectivity
  – up to 8/8 I/Os
  – USB Audio Class 2.0 compliant for control from PC, Mac OS and iOS Apps –
  – stand alone A/D & D/A converter mode
• Neutrik XLR connectivity and break-out cables

CONFIGURATION

Bus/Format USB 2.0 / Compliant with the USB 2.0 audio specification
Size 254 mm x 96 mm x 36 mm
Operating: Temp / Humidity 0°C to +50°C / 0 % to 90 % (non condensing)

OVERALL AUDIO

A/D and D/A converters 24-bit / frequency: 32 kHz, 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 192 kHz
Audio formats supported PCM 8, 16, 20, and 24 bits, full duplex
Latency 3.4 mS
– analog-to-PC or PC-to-Analog (Windows 7 / 64 bits) 3.8 mS
– analog-to-MAC or MAC-to-Analog (Mac OS X 10.6.8)
Channel phase diff. (A/D and D/A) ± 0.2° (20 Hz – 20 kHz)

INPUTS

Analog inputs (mono) 2 balanced Mic/Line inputs
Digital inputs (stereo) 1 AES/EBU (AES3-2003) compliant
Switchable 48V phantom power 7 mA Max on each input**
Analog input gain
– From 0 to 55 dB by 1 dB steps,
– switchable Pad 30.0 dB
Input sensitivity 0 dBfs adjustable from -30 dBu to +25 dBu (Line) and -60 dBu to -5 dBu (Mic)
Maximum input level/impedance Line: +25 dBu / > 3.5 kOhms; Mic: -5 dBu / > 2 kOhms
Frequency response (A/D Input)
– at 48 kHz: 20 Hz–20 kHz +0/-0.5 dB
– at 96 kHz: 20 Hz–40 kHz +0/-0.6 dB
– at 192 kHz: 20 Hz–80 kHz +0/-2.0 dB S/N
Analog input, typical S/N: 108 dB unweighted and 111dBA @48 kHz THD+N
Analog input, typical < -105 dB unweighted and -107 dBA / 20 Hz-20 kHz, @48 to 192 kHz ref 1 kHz at –3 dBfs
Mic inputs E.I.N., typical < -128 dB / Zsource = 40 Ohms; Pad Off; gain 55 dB OUTPUTS Analog outputs (mono) 2 balanced Line outputs Digital outputs (stereo) 1 output, AES/EBU (AES3-2003) compliant Maximum analog level/impedance at 48 kHz: +10 dBu / 2×33 Ohms Frequency response – at 48 kHz: 10 Hz–20 kHz +0/-0.1 dB – at 96 kHz: 10 Hz–40 kHz +0/-0.3 dB – at 192 kHz: 10 Hz–80 kHz +0/-1.3 dB S/N Analog output, typical > 111 dB unweighted @48 kHz
THD+N Analog output, typical < -98 dB unweighted / 20 Hz-24 kHz @48 kHz to 192 kHz, ref 1 kHz at –3 dBfs Headphones output (stereo) – Dedicated output stage, – > 10 mW from 32 to 600 Ohms 10Hz-20 kHz +-0.1 dB;
– Dynamic range: 93 dB @32 Ohms, typical

CONNECTORS

Audio connectors One XLR female for left input channel
Sub-D 25 pts (222-Mic) for all analog and digital I/Os 6.35 mm jack for stereo headphone output
Neutrik XLR breakout cable included USB connector Standard, includes two A-type on PC side, one mini-B USB on card side

MIXER AND PROCESSING

Analog-to-AES/EBU Bridge In Stand-alone mode
AES/EBU-to-Analog Bridge In Stand-alone mode
Embedded Mixer Zero-latency headphone Mixer

ENVIRONMENTS

Operating systems: Windows 10, 8, Windows 7 (32 and 64 bits), Vista, Windows XP, Mac OS X, Linux
Management: depending on the host operating system’s implementation of the USB Audio 2.0 specification. Microsoft Windows management is provided through DirectSound, Core Audio, WASAPI third-party ASIO driver.

The PCIe LX-DANTE sound card bridges professional audio software applications to Dante networks. By supporting 128 x 128 redundant channels, and providing extremely low latency, LX-DANTE matches all the requirements for high channel density, low latency, and reliability of professional DAWs used for high quality recordings, processing, and multi-channel playout over Dante networks.

• Standard PCI Express 4x card format
• 128 x 128 redundant ch. at 44.1, 48kHz, 88.2, 96KHz
• 64 x 64 redundant channels at 176.4 or 192kHz
• Dante plug and play media networking
• AES67 compatibility (48 samples per packet, 48 kHz, SAP)
• Ultra-low latency with sub-microsecond synch.
• Drivers: ASIO (Windows), Alsa (Linux)
• Supported operating systems: Windows as of Windows 7, Windows Server as of 2008 R2, Linux as of kernel 3.10
• Seamless Dante network redundancy via the two Gbit network interfaces
• Interoperable with any other Dante-powered device, and with other AES67-compliant devices
• Can be used in Thunderbolt external chassis

Configuration
PCI Express card PCI EXPRESS™ (PCIe®) x4, ( x8, x16 compatible)
Size Length: 117mm, Height: 98.4mm, Width:18mm
Network interfaces Two Gigabit Ethernet RJ45 connectors

IP audio
IP audio transport Dante Audio over IP, AES67
Redundancy Glitch-free Dante audio redundancy using dual Ethernet networks
Clock synchronization Master or slave

Audio
Audio Channels 128 / 128 I/O channels @ up to 96kHz
64 / 64 I/O channels @ 192kHz
Supported Sample Rates 44.1, 48, 88.2, 96, 176.4 and 192kHz
Sample bit-depth 24 bit PCM Audio
Latency Round trip latency as low as 2.99ms

Software environment
OS Supported From Windows 7, and from Windows Server 2008 R2
Linux as of kernel 3.10 (see LX-Dante support page)
Drivers Windows: ASIO
Linux: Alsa

Compatibility with expansion chassis

Thunderbolt chassis OWC (Other World Computing) Mercury Helios `
Sonnet Echo Express SE II
Magma ExpressBox 1T 1 Slot

LX-IP provides the perfect gateway from mission critical applications to RAVENNA and AES67 AoIP networks. Its 64 channels, 0.5 ms latency and phase accuracy match all requirements for high density, ultra-low latency on-air or production studio workflows. Hardware-based architecture maintains performance regardless of the computational load of other applications running on the host system.
LX-IP with MADI optional interface provides a seamless migration path from legacy digital audio to AoIP and provides all MADI features described below.

• High channel count, ultra-low latency and phase-accurate audio distribution on synchronous AoIP networks
• Phase-accurate clock synchronization between digital audio and AoIP networks
• Interoperability with traditional digital audio
• Performance & stability independent of PC applications
• Mission critical reliability
• Play/record 64 audio channels between a DAW (Digital Audio Workstation) or automation application and AES67 or RAVENNA synchronous AoIP network
• Play/record 64 channels between a DAW or automation application and digital audio MADI equipment
• Provides zero-latency routing on-board between 2*64 RAVENNA/AES67 channels, 64 DAW channels and (option) 64 MADI channels
• Provides PTP Grand Master clock to synchronous AES67 or RAVENNA AoIP network. This Grand Master can be slaved to WordClock or MADI inputs
• Synchronises RAVENNA/AES67 AoIP streams on external PTP clock. The MADI output and WordClock clocks can be slaved to the external PTP clock
• 64/64 Record/Play channels on PCIe bus
• 64/64 AES67/RAVENNA I/O channels on each of the two Gigabit Ethernet interfaces
• 64/64 I/O MADI (AES10) optical channels (option)
• Ultra-low sub-millisecond round trip latency
• Fully compliant with AES67 interoperability recommendations, including Unicast SIP support
• Packet size from 128 down to 1 audio sample per RAVENNA packet
• Local clock eligible as PTP Grand Master
• Zero delay embedded routing matrix for AoIP / MADI / PC channels
• Performance maintained regardless of the computational load of applications running on the host system
• Fully configurable through Web user interface and EMBER+ protocol

Configuration

Bus/Format: PCI Express(R) x1 (compatible x1, x4, x8, x16 slots)
Size: 111.15 mm x 167.65 mm x 20 mm
Power requirements (+3.3V/+12V): 0.4 A / 0.12 A
Operating: temp / humidity (non-condensing): 0°C / +50°C • 5% / 90%
Storage: temp / humidity (non-condensing): -5°C / +70°C • 0% / 95%

Inputs and Outputs

Connectors
– 2 Gigabit Ethernet RJ45 ports for RAVENNA I/O (dual port or Primary / Back up mode)
– 1 optical connector for MADI I/O (Factory option) (multimode, 1300nm)
– 1 BNC for Word Clock In / Out
RAVENNA I/O channels: 2 banks of 64/64 I/O (Mono) channels at 44.1 kHz or 48 kHz (64/64 I/O on each Gigabit Ethernet interface)
RAVENNA packet size: From 128 down to 1 (ultra-low latency profile) audio samples per RAVENNA packet
AES67 compliance: Full compliance in all respects with AES 67
Supported audio payload formats: PCM16 / PCM24 / PCM32 / AM824 (PCM24+AES3 channel status)
PC Record/Play channels: 64/64 simultaneous Record/Play (Mono) channels to/from PC
MADI (Multichannels Audio Digital Interface) inputs and outputs:
– Optical I/O connector, 64/64 I/O (Mono) at 48 kHz sampling frequency and 32/32 I/O (Mono) at 96 kHz
– (Factory option)
Word Clock input or output
– BNC connector, Input or Output position selectable by software.
– Input: TTL , impedance selectable by jumper (75 Ohms / HighZ).
– Output: Max 5 Vpp, 75 Ohms output impedance
Clock sources:
– PTPv2 (IEEE1588-2008) from network or internal clock or Word Clock or MADI input
– Local clock eligible as GrandMaster PTP
– Local clock precision : better than 10 ppm
Sampling frequencies:
– From local clock: 44.1 kHz, 48 kHz and 96 kHz (MADI)
– From network or Word Clock: 44.1 kHz, 48 kHz and 96 kHz (MADI)
– From MADI: 44.1 kHz, 48 kHz, 96 kHz

Control and routing

Control
– HTTP (web pages from embedded server)
– EMBER+
Routing: Zero latency on-board routing matrix between RAVENNA, PC Rec/Play and optional MADI channels

Environment

Latency: Round trip time down to 0.8 ms (excluding IP network)
Supported operating systems: Windows 10, 8 and 7 32/64bits, Windows server 2003/2008, Linux
Supported drivers: ASIO, WASAPI / low latency WDM DirectSound, ALSA