Sunday, October 2, 10:45 am — 12:15 pm (Rm 409B)
Scott Norcross, Dolby Laboratories - San Francisco, CA, USA
P25-1 An Efficient Algorithm for Clipping Detection and Declipping Audio—Christopher Laguna, Georgia Institute of Technology - Atlanta, GA, USA; Alexander Lerch, Georgia Institute of Technology - Atlanta, GA, USA
Convention Paper 9682 (Purchase now)
P25-2 A Two-Pass Algorithm for Automatic Loudness Correction—Alexey Lukin, iZotope, Inc. - Cambridge, MA, USA; Russell McClellan, iZotope, Inc. - Cambridge, MA, USA; Aaron Wishnick, iZotope - Cambridge, MA, USA
Loudness standards for broadcast audio, such as BS.1770, establish target values for the integrated loudness, true peak level, and short-term loudness of a record. Compliance with these three targets can be challenging when the dynamic range of a record is high, so software for automatic loudness correction is important for speeding up the workflow of post-production engineers. This work reviews existing software implementations of automatic loudness correction and proposes a new algorithm that provides efficient simultaneous correction of all three targets.
Convention Paper 9683 (Purchase now)
P25-3 A Low Computational Complexity Beamforming Scheme Concatenated with Noise Cancellation—Jin Xie, Marvell Technology Group Ltd. - Santa Clara, CA, USA; Sungyub Daniel Yoo, Marvell Technology Group Ltd. - Santa Clara, CA, USA; Kapil Jain, Marvell Technology Group Ltd. - Santa Clara, CA, USA
In this paper we present a microphone beamforming algorithm. This algorithm has been implemented in Marvell’s proprietary digital signal processor embedded in Marvell’s audio codec chip. This beamforming algorithm features (1) easy to implement; (2) sound source localization (SSL) and sound source tracking, (3) single in single out frequency domain noise cancellation. Lab tests show that the performance is better than the reference existing codec.
Convention Paper 9684 (Purchase now)