Overview
The APT IP Codec is a cutting-edge, professional IP stereo codec geared up with unique, market-leading technology to provide next generation broadcast performance. Innovative and feature-packed, the APT IP Codec elevates the quality of reliable, cost-effective, and accurate temporal IP audio transmission to a level never before achieved.
The proven SureStream redundancy and distributed intelligence of ScriptEasy is extended by the new compression format APTmpX for composite/MPX signals and the highly precise synchronization of IP streams with APT SynchroStream.
The APT IP Codec is perfectly equipped for individual FM feed as well as multi- and single-frequency broadcasting. It is suitable for mission critical applications and provides extensive control and monitoring functions to manage your audio, data and network conditions, as well as other equipment located at the transmitter site.
With the APT IP Codec, you know you will enjoy the rock-solid performance upon which APT has earned the trust of countless broadcasters worldwide.
Features
+10 Years Experience: Our team of engineers has extensive experience optimizing our algorithm for redundant streaming, making SureStream synonymous with reliable transmission in lossy IP networks.
Low Latency - Low Costs: SureStream enables the broadcaster to turn imperfect, but much cheaper services, into true broadcast-grade, low-latency IP connections.
Scalability and Flexibility: SureStream is the most flexible and scalable solution for content transmission protection, able to combine multiple paths from any combination of MPLS, Satellite, Microwave, xDSL and/or Cellular (4G/5G), creating a unified super robust connection to get your audio from point A to B.
Stable Latency: The GPS-based SynchroStream eliminates variable latencies of an IP network within unprecedented narrow limits. For program transmissions, a temporal synchronized connection appears like a synchronous link.
Complete Control Over Target Latency: SynchroStream requires a single setting on the IP Encoder to define the target latency to each Decoder at a transmitter site. Only one setting is required, and all Decoders are synchronized; accurate and stable to the millisecond. Fine-tuning is done at the decoder(s) in the array.
Synchronized FM Modulation: Temporal fine-tuning is the key to optimal geographic positioning of overlapping modulations of FM carriers. SynchroStream enables modulation control with the uniquely fine granularity of < 50 meters in terrain.
Compressed Composite/MPX: APTmpX is the world’s first and only non-perceptual MPX/composite algorithm to save network bandwidth. It protects the sonic signature generated by the station’s sound processor settings.
Low Bitrate, Low Delay: An APTmpX sample is decodable on its own. Packet losses in the network have as little effect as those of base-band audio samples. Low delay transmission is inherent and with a bandwidth requirement of <300, <400, <600 or < 900 kbps, non-dedicated IP connections can be used.
APT codecs offer the reliability, quality, and redundancy required by broadcasters for applications such as Studio-Transmitter links, studio networking and remote broadcasting/OBs. With both stereo and multi-channel audio platforms available, APT units can be deployed as a simple STL, a large-scale broadcast audio network or an audio distribution installation for commentary or in-store music.
Remote studio codecs
Studio to transmitter link (STL)
Secure IP transport
Specifications
AUDIO
- Asymmetric Audo: Independent audio modes for sent and receive, Tx and Rx clock domains and auto-detection, Reply-to-Sender, NAT traversal mode
- Analog I/O: Electronically balanced, capacitive isolated XLR for Left/Right, Imp. Hi/Lo and 600 Ω, level adjustment in 0.1 dBu steps
- Digital Audio I/O: AES-3, 24 Bit, transformer balanced, Imp. 110 Ω, XLR-Connectors
AUDIO FORMATS
- Multi Algorithm Suite: Eapt-X 16/24 bit, lin. PCM 16/24 bit, MPEG2/4 AAC LC/LD/ELD, HE-AACv1/2, MPEG1/2 L1/2, OPUS
- MPX Formats (AES192): optional, lin. MPX 16/24 bit APTmpX @ 300/400/600 & 900 kbp
STREAMING MODES
- Stream Types: Multiple stereo Audio, UDP and RTP forwarding, Reply-to-Sender, NAT traversal mode
- SIP Modes: Peer-to-peer & SIP-Server mode, multiple SIP user accounts, sym. and asymm. SIP profiles
- Unit Clock Modes: Asymmetric, master, slave, NTP-based & high precision GPS (optional)
- Jitter Buffer: 2 - 5000 ms with packet re-sequenzer
- QoS DiffServ: (RFC 2474) per stream
- Redundant Streaming: SureStream Option, multi-stream packet-by-packet redundnacy
- Backup Feature: SD Card for audio file storage
MONITORING & ALARMS:
- Adjustable Silence Detectors (Inputs & Outputs), Event Logs, Alarm Relays. SNMP Traps / Noti
PHYSICAL INTERFACES
- Audio on XLR: L / R analog In-Outputs digital (AES 3) In-Output, ext. AES11 reference Input
- Headphone: 1/4” (6.3 mm) Jack Socket (front)
- AUX Data: D9-way connector
- GPIO: D15-way connectors
- Network: 2x RJ45
- GPS Clock: (otional) 2x BNC (10 MHz, 1 PPS)
- AC Power: 1 + 1 (optinal) IEC type
- DC Power: 1 + 1 (optinal) Power D3-way connectorfications