Even the most experienced presenter can sometimes inadvertently create varying vocal levels. To assist in loudness based programming, a voice leveler can be used to pre-condition the voice component prior to combining it with the program element.
Operating in its powerful proprietary ‘Level Mode’. the perceived level of your voice sources is automatically adjusted to a pre-defined target value by our Level Magic™ algorithm. The system works with a multi-stage design.
The first stage is the Transient Processor, taking care of problems with short signal bursts, so called transients. They carry little energy and thus their contribution to the perceived loudness is very limited.
The second stage is an Adaptive Gain Control (AGC) network, leveling the overall signal energy to keep the loudness persistent. The user can adjust the overall system characteristics from relaxed to aggressive, while the algorithm automatically adapts its parameters to the incoming audio signal.
This is to guarantee the best performance under all conditions, without artifacts like pumping or breathing.
The Level Magic™ technology has proven its effectiveness and leading audio quality in over 25 000 installations worldwide.
Level Magic™ can be used to create consistency over one continuous audio stream or with switching sources. Also, with its adaptive abilities it is suitable for single channel sources, like voice or instrument inputs, as well as bus duties and multichannel outputs.
It is designed to work autonomous and unmonitored, but also allows for automated or manual parameter adjustment and control. This opens up potential use cases to a vast amount of scenarios.
Ranging from studio and live production, broadcast master busses, distribution chains, to customer head units, playback devices, public address systems and many more.
With multiple sources, a combination of instances working in Level Mode in all input channels and one loudness standard normalization instance on the summed output side can yield excellent results.
Level Magic™ at a glance:
- High performance loudness management
- Unmatched audio quality without annoying artifacts
- Ideal for stand-alone operation
- Optimized for all international loudness regulations
- Compliant with ITU-R BS.1770 (all revisions), EBU R128, ATSC A/85, ARIB TR-B32, Free TV OP-59, Portaria 354 and more
2 Channel Voice Audio Processor - D*AP4 VAP
- Dual mono or stereo voice channel configuration
- 2 channel program path in addition to voice channels
- 5 band Parametric EQ and Spectral Signature dynamic EQ
- Dynamics with upward/downward compressor and expander/gate
- 19”, 1RU device, redundant PSU
C8000 - Feature Set
- 19” frame based fully modular system (1Ru or 3RU frames)
- Wide range of audio processing functionality
- Extensive support for Dolby® decoding and encoding
- SDI interface for embedding/de-embedding of audio, Metadata transport and lip sync control
- Network based automation, control, management and measurement