ATSC 3.0 Delivered
Linear Acoustic AMS Authoring and Monitoring System is a comprehensive solution for real-time authoring, rendering, and monitoring of advanced audio programs for the ATSC 3.0 Digital Television System. The audio system in ATSC 3.0 provides listeners with a personalized, immersive audio experience using Next Generation Audio (NGA) technologies, including MPEG-H. AMS simultaneously delivers advanced audio for ATSC 3.0 broadcasts and 5.1-/2-ch audio for ATSC 1.0 broadcasts.
The AMS web interface controls the interactive features of MPEG-H offered to the viewer. Using smart metadata and Linear Acoustic APTO loudness control, AMS facilitates easy audio mixing and authoring operations, even in the most demanding production scenarios. Individual audio elements are combined with user specified metadata to create immersive and personalized audio programs for the viewer. The AMS web interface allows the authoring engineer to easily control the interactive features of MPEG-H offered to the viewer, and build presets for different listening experiences. This enables viewers to personalize their sound experience for optimal playback, from mobile devices to immersive home theatres.
The system provides a web interface for the following functions:
Authoring Made Easy
Authoring an MPEG-H audio stream using the web interface is as simple as assigning the 15 available mono inputs to the 10 available channel groups. From defining music beds, identifying independent objects, or grouping objects to be switched by the user, full control is available.
Comprehensive Real-Time Monitoring
Linear Acoustic AMS can be used in production to actively author and monitor metadata, or in other parts of the broadcast chain to monitor and validate audio streams containing a control track which were authored upstream. In both scenarios, AMS is capable of outputting 15-channels of discrete audio with a metadata control track, a 5.1-channel rendered output, 2-channel rendered output, and a dedicated monitoring output, all simultaneously.
Loudness Adaptation by Linear Acoustic APTO™
AMS is equipped with Linear Acoustic APTO, the state-of-the-art loudness adaptation technology designed to carefully control audio levels in a way that preserves the transients, sonic image and artistic intent of the source, while ensuring loudness consistency and compliance for any desired target.
The xNodes xFactor
The Telos Alliance xNode family of AoIP devices offer optional analog, AES/EBU, GPIO logic, and SDI I/O, all with full Livewire+ AES67 Audio over IP support.
Telos Alliance SDI xNodes can de-embed 3G/HD/SD-SDI inputs, extracting up to 16 channels of audio to the Livewire+ AES67 port. The audio can be re-embedded into the SDI output stream with full video delay compensation for each SDI input, ensuring that audio video synchronization is maintained.
Using the web interface, up to 36 outputs may be configured, including 16 channels of authored audio plus control channels, up to 12 channels for monitoring, and two additional sets of output channels for rendered broadcast outputs.