Use Merlin for high fidelity bidirectional IP audio and communications from the studio to a wide range of Tieline remote codecs or smartphones using Report-IT.
Now supports the Opus Algorithm, Wi-Fi, ISDN and POTS!
Merlin is the newest addition to Tieline’s powerful family of audio codecs and is designed to deliver rock solid, high fidelity IP audio over an extensive range of public and private IP networks. It also supports connections over transports including IP, ISDN and POTS.
The 1RU Merlin rack mount IP codec delivers high quality bidirectional stereo and full duplex communications for point-to-point and remote broadcast connections. It is ideal for studio and remote truck installations and delivers:
- High fidelity point-to-point stereo, or
- Point-to-point stereo plus a separate bidirectional IFB channel, or
- Up to two mono connections to different Tieline IP codecs, or smartphones using Report-IT.
- High reliability over IP networks without Quality of Service
- Support for ISDN or POTS network connections
- Powerful audio and data routing
- Simple local or remote command and control
- Recallable connection programs
Reliability and Connectivity
Merlin will connect to all Tieline IP broadcast codecs, including G3, Bridge-IT and Genie codecs and Report-IT Enterprise smartphone applications.
It has dual 1 Gigabit (10/100/1000) onboard Ethernet ports, dual redundant power supplies, audio silence detection and IP network backup solutions to ensure you’re always on the air. An automated network failure detection feature provides switching to a backup IP LAN connection. You can also fail over to backup ISDN or POTS connections as required using optional ISDN or POTS modules.
Merlin is specifically designed to deliver a wide variety of point-to-point and remote broadcast connections over the full suite of wired and wireless IP networks, including:
- Public Wide Area Networks (WANs)
- Private Local Area Networks (LANs)
- 3G and 4G wireless Networks #
- Wi-Fi and WiMAX Networks #
SmartStream: Rock Solid Audio using Inexpensive IP Links
As an IP audio pioneer, Tieline was the first major codec manufacturer to integrate SmartStream features such as Automated Jitter Buffer management, Forward Error Correction (FEC) and error concealment techniques in all IP codecs. These features dynamically respond to variable conditions over unmanaged IP networks like the public internet to ensure reliable streaming.
Now SmartStream PLUS has revolutionised IP broadcasting by delivering the rock solid and reliable STL-grade audio quality you would expect over a T1/E1 link by using inexpensive unmanaged IP networks like the internet for remotes.
Save Money Now with SmartStream PLUS Redundant Streaming
Some other manufacturers charge thousands of dollars for IP management software like SmartStream PLUS as if it’s an optional extra. However Tieline believes high performance and STL-grade rock-solid reliability is an essential part of each and every broadcast, and delivers its renowned SmartStream PLUS as standard in Merlin codecs.
Tieline can show you how an investment in new codecs will pay for itself in just a few months by using SmartStream PLUS over inexpensive IP links to transport STL-grade, high fidelity audio at a fraction of the cost of synchronous leased lines. With SmartStream PLUS:
- Stream simultaneous redundant data streams and deliver seamless redundancy by switching back and forth, without loss of audio, from the nominated primary data link to the backup link if one fails and then subsequently recovers. Note: Use IP links from two different IP network providers for optimal redundancy over mission critical STL connections.
- When multiple redundant audio streams are sent, the decoding codec automatically reconstructs audio into a perfect single stream on a first packet arrived basis to ensure audio integrity.
Flexibility and Interoperability
|Tieline G5 ISDN module|
Many broadcast networks are transitioning to IP but still require connectivity over ISDN or POTS. Tieline Merlin codecs deliver flexibility by allowing you to connect over ISDN, POTS or IP networks as required. Simply purchase optional ISDN G5 or POTS G5 modules, insert them in the codec and you are ready to connect over ISDN, POTS or IP on demand.
Merlin supports both IPv4 and IPv6 (Dual Stack) protocols, so broadcasters can be assured that they have made a smart investment for the future as new networks and technologies emerge over time.
Connect over IP with any SIP-enabled IP codec brand that supports the EBU N/ACIP tech 3326 standard, as well as VoIP telephone devices supporting G.711 and G.722. Merlin is also compliant with I3P EBU334 requirements for interoperability of AoIP production intercoms.
Merlin is capable of 16-24bit 22kHz low latency linear PCM audio at up to 48kHz sampling and features aptX® Enhanced, LC-AAC, HE-AAC v1 and v2, AAC-LD, AAC-ELDv1 and v2, Opus, MPEG II, MPEG Layer-3, Tieline Music and MusicPLUS, G.722 and G.711 algorithms.# Requires 3rd party device * Feature not available in the first release.
- Robust DSP-based architecture designed for rock solid, mission critical broadcast audio.
- 24 Bit 48kHz audio sampling.
- Analog and Digital (AES3) inputs and outputs.
- Bidirectional Mono, Stereo or dual mono connections.
- Bidirectional Mono or Stereo plus a separate full-duplex IFB circuit.
- Uncompressed PCM audio plus the low-delay, cascade resiliant aptX® Enhanced algorithm.
- Other popular algorithms including LC-AAC, HE-AAC v1 and v2, AAC-LD, AAC-ELDv1 and v2, Opus, MPEG II, MPEG Layer-3, Tieline Music and MusicPLUS, G.722 and G.711.
- Dual 1 Gigabit (10/100/1000) Ethernet ports support automatic switching for redundancy.
- Connect over Wi-Fi using USB Wi-Fi dongles.
- Primary and backup audio over IP, ISDN* and POTS*.
- Automated failover to backup connections.
- SmartStream PLUS dual redundant IP streaming:
- Fuse-IP network bonding (data aggregation) of Ethernet ports or Ethernet and Wi-Fi.
- SIP (SDP) compatible with EBU N/ACIP 3326 for interoperability with other codec manufacturers.
- Deterministic SIP answering and call routing: Configure up to 6 SIP accounts and connect up to 2 SIP interfaces.
- Compatibility over ISDN with Worldcast, Prodys, Mayah, CDQ Prima, AETA, AEQ, Orban and more.
- Compatibility over ISDN using aptX Enhanced encoding with Worldcast, Prodys Prontonet and Mayah.
- Selectable fixed or automatic jitter buffer: 5 automated settings available to manage network connectivity and avoid dropouts (includes configurable min and max buffer settings).
- Configurable Forward Error Correction (FEC).
- Highly advanced error concealment strategies.
- Real-time IP connection packet statistics.
- Real-time POTS line quality and IP link quality displayed.
- Support for Line Hunt dialing/answering and caller IDs using Tieline IP.
- Full hardware front panel interface including navigation, LCD display, PPM meters and dialing key pad.
- User-friendly HTML5 Toolbox Web-GUI enables codec configuration and control remotely over WANs.
- Toolbox Scheduler supports loading/unloading, connecting and disconnecting programs.
- Auto switching, dual redundant internal AC power supplies.
- Automated failover to backup connections.
- IPv4 & IPv6 compatible and ready.
- Asymmetric algorithmic encode/decode.
- Low latency in-band RS-232 aux data channel.
- Configurable software rules engine via web-GUI for Control Port (4 inputs/outputs) functions, plus WheatNet-IP LIO compatibility.
- Upgrade firmware from USB sticks.
- Backup and restore configuration via USB sticks.
- Automated firmware upgrade notifications via Web-GUI.
- Support for multiple languages: English, Spanish, Portuguese and French.
- Backward compatible with all Tieline IP codecs.
- User name and password protection of codec and Toolbox Web-GUI.
- Support for SSL Security Certificate installation.
- SIP Whitelists and Blacklists allow filtering of SIP URIs and User Agents.
|Analog Audio Inputs||2 x Female XLR line inputs|
|Analog Audio Outputs||2 x Male XLR|
|AES3 In||1 x female XLR (Channel 1 in; shared with Ch1 analog input)|
|AES3 Out||1 x male XLR|
|Auxiliary Input||1 x 6.35mm (1/4″) Mic/Line level Jack on rear panel|
|Headphones Out/Aux Out||1 x 6.35mm (1/4″) Jack on rear panel and 1 x 6.35mm (1/4″) Jack on the front panel|
|Control Port In/Out||Four relay inputs and four opto-isolated outputs for machine control via a DB15 connector.|
|Audio Input Impedance||High Impedance > 5K ohm|
|Output Impedance||<50 ohm Balanced|
|Clipping Level||+22dBu (input and outputs)|
|A/D & D/A Converters||24 bit|
|Frequency Response at 48kHz||20Hz to 22kHz|
|THD and Noise (Analog)||<0.0035% at +16dBu or -89dBu unweighted|
|THD and Noise (Digital)||<0.000056%|
|Analog Signal To Noise Ratio||>98.5dB at +22dBu, unweighted|
|IP Sample Frequencies||16kHz, 32kHz, 44.1kHz, 48kHz|
|IP||Tieline Music, Tieline MusicPLUS, G.711, G.722, MPEG-1 Layer 2, MP3, LC-AAC, HE-AAC and HE-AACv2, AAC-LD, AAC-ELD, Opus, 16/24 bit aptX Enhanced|
|IP (uncompressed)||Linear PCM16/24 bit 48kHz sampling|
|Data and Control Interfaces|
|USB||USB 2.0 Host port on the front panel supports Wi-Fi using a USB Wi-Fi dongle and USB stick firmware upgrades|
|LAN||2 x 10/100/1000 RJ45 connectors|
|Advanced Networking||VLAN tagging (IEEE 802.1Q,802.1p)|
|Serial||RS232 up to 115kbps with or without CTS/RTS flow control via female DB9 connector, can be used as a proprietary data channel|
|Protocols Supported||Tieline, DHCP, SNMP, DNS, HTTP, IGMP, ICMP, VLAN, IPv4/v6, FEC, SIP/SDP (EBU N/ACIP Tech 3326 compliant), RTP, I3P EBU3347 compliant|
|ISDN via module||Optional via module slot|
|POTS via module||Optional via module slot|
|Front Panel Interfaces|
|Display||256 x 64 monochrome LCD|
|Keypad||21 button keypad|
|Navigation||5 button keypad|
|Size||1U x 19″ Rackmount|
|Dimensions||19” x 1.75” x 13.5” [482mm (W) x 44mm (H) x 343mm (D) including rear connectors]|
|Power Consumption||Dual AC 100-240V IEC power inlets; 1A – 50-60Hz|
|Operating Temp.||0°C to 45°C (32°F to 113°F)|
|Humidity Operating Range||20% ≤RH ≤70% (0 to 35°C/32°F to 95°F), non-condensing|
MPEG Layer-3 audio coding technology licensed from Fraunhofer IIS and Thomson Licensing.