Linear Acoustic’s APTO technology offers a suite of features, including dialog-based normalization.
At IBC Linear Acoustic will spotlight its APTO technology, including the APTO.file and APTO.stream applications, which are designed to help broadcasters automate their file-based operations with a single toolkit that supports the entire broadcast chain, from production to playout.
The company said file-based and real-time workflows require a unique approach. Sound quality must be preserved, while loudness levels must be maintained within comfortable ranges, and the overall average level must meet the recommended target. Monophonic, stereophonic and multichannel audio assets need to be assessed and aligned properly, even when downscaling is applied.
Linear Acoustic's APTO architecture relies on sophisticated, optimized processing that is capable of properly defining the ideal amount of required adaptation for each distribution platform. This approach results in artifact-free audio. It includes both file-based and real-time processing that allows the content owner, the distributor or the broadcaster to adapt any content, anytime, and in any operational scenario to provide the viewer with the best possible listening experience. It supports mono to 7.1 audio formats and implements all international loudness recommendations, including dialog-based normalization.
The algorithm architecture includes a Loudness Analysis block, which leverages real-time loudness assessments of the sound source based on psychoacoustic models that analyze (in real time) all main aspects involved in human hearing: that is, frequency, intensity, duration and sound source direction. It extracts the loudness components of the audio asset and intelligently adapts them in order to produce a comfortable listening experience with consistent average levels.
The Loudness Analysis block, then provides the data to the Loudness Adaptation block. The adaptation is based on computational algorithms that aim to reduce the loudness modulation in real-time in order to maintain the foreground sounds within the listening “comfort zone.” The amount of adaptation is defined in real-time according to the incoming level and predetermined profiles, which include universal and genre-oriented modules as well as dialog-centric or agnostic adaptation.
The current profile lists for APTO.file include movie adaptation (theatrical mixes to broadcast), broadcast, radio (consistent radio levels), streaming requirements, news dialog consistency, FICAM audio requirements and inflight entertainment (loudness and dialog consistency for inflight entertainment).
The applications for APTO.stream include EBU-R128, ATSC/A85, FreeTV OP59, ARIB TR-B32, general broadcast/TV broadcast, radio broadcast, streaming requirements, home-cinema adaptation, enhance listening in gaming and improving earphone listening.
Depending on source level, distribution platform, and the target loudness values, APTO will detect the specific amount of loudness correction required by acting on the micro- and macro-dynamic modulations in the signal. The result is comfortable listening with consistent average levels and increased dialog intelligibility, especially on devices where that aspect might be critical.
Once the source audio is adapted, it is again measured and aligned according to the international recommendation standards, such as ITU-R.BS1770-4, or according to additional dialog detection measurements.
APTO can be integrated in an automated workflow, or accessed through control panel that sets the amount of targeted adaptation. This feature is available as an end-user device application that allows customization of the listening experience to taste. By handling loudness metadata, including Loudness Status, Program Level, Dialog Level, Loudness Range, Real-Time Loudness Level and Program Duration, the APTO ecosystem provides full control of any audio transmission while complying with international recommendations.
You might also like...
Digital audio interfaces were developed as a way of avoiding generation loss between devices.
The recursive filter has the advantage of using less hardware, but is more complex to understand.
Once the basic requirements for reproducing sound were in place, the most significant next step was to reproduce to some extent the spatial attributes of sound. Stereophony, using two channels, was the first successful system.
Having looked at how microphones are supposed to work, here we see that what happens in practice isn’t quite the same because the ideal and the actual are somewhat different.
There are two approaches to digital filtering. One is to implement the impulse response directly. The other is to use recursion. Here we look at the direct implementation.