APM6803+T

Audio Processing System (APS)
At the core of the APM6803+ lies a powerful and flexible signal processing engine which consists of two high end Digital Signal Processers (DSP) as well as custom built hardware to process multiple channels of audio. Functions are added purely through software keys with no plug-ins required, making the system easy to expand and future proof against increasing requirements. The diagram above shows all of the possible processing blocks. Availability of each block will depend on processor utilization. The chart below lists the requirements for each function. The numbers reflect the utilization for a single DSP. There are two DSPs on the APM6803+.

Virtual Stream Interface
A virtual stream interface greatly simplifies configuration, hiding the complexity of the underlying system and making it possible to create a high-level, processing-intensive signal flow with just a few mouse clicks. Managing audio at the “stream” level, rather than having to manage individual channels, makes configuration easier, quicker and less error-prone. Virtual streams unleash the full processing power of the APM6803+ without bogging down the user with complicated routing and configuration menus.

Flexible, Dynamic Licensing
The Audio Processing System is highly configurable and can support a wide range of Dolby* and DTS processing blocks. Software key licenses are required to enable individual Dolby* encoders and decoders. For DTS functions, a flexible token system has been designed to allow users to change the type of processing without purchasing additional licenses. All licenses are field upgradeable, enabling users to start with a minimal setup and grow the system as requirements dictate.

Intelligent Metadata Handler
One of the key topics in modern broadcast audio management is loudness. Broadcasters spend significant effort to correct the loudness of program material at ingest so that they can provide a consistent audio level to their listeners, while maintaining the full dynamic range of the signal. Metadata is often used to assist in this effort, but in practice, situations inevitably arise that require intervention, such as incorrect/ missing Dialnorm, live content or content that has not yet been processed and set to the correct loudness. The APM6803+ easily supports agile, fixed and hybrid workflows, and goes one step further to include features that provide automatic assistance for handling unknown or incorrectly processed audio. The APM6803+ protects the output against incorrect loudness levels whilst maintaining the full dynamic range of the original content wherever possible.

Automation Control
Predefined loudness control settings are available for instant recall by automation, allowing users to create simple, effective automation profiles. Additional user-defined presets are available for maximum flexibility. The APM6803+ has a large number of GPI ports available for automation control.

Real Time Loudness Control
DTS Neural Loudness Control incorporates complex real-time loudness correction system which uses critical band based measurement for the most accurate modeling of human loudness perception. By detecting spectral and density differences, inter-channel relationships and temporal overlaps within the audio signal, NLC provides transparent loudness correction which exceeds ITU-R BS.1770 requirements.

Automatic Audio/Video Alignment
As you build your audio system, the APM6803+ keeps track of the audio delay through the audio processing system, and will add the corresponding delay in the video path to ensure lipsync is preserved. Lipsync errors outside the APM6803+ can easily be corrected by manually adjusting either video or audio delays.