A novel scheme for low bitrate unified speech and audio coding - MPEG RM0. - In: 126th Audio Engineering Society convention 2009, (2009), S. 1142-1154
Unified speech and audio coding scheme for high quality at low bitrates. - In: IEEE International Conference on Acoustics, Speech and Signal Processing, 2009, ISBN 978-1-4244-2353-8, (2009), S. 1-4
http://dx.doi.org/10.1109/ICASSP.2009.4959505
An error robust ultra low delay audio coder using an MA prediction model. - In: IEEE International Conference on Acoustics, Speech and Signal Processing, 2009, ISBN 978-1-4244-2353-8, (2009), S. 5-8
http://dx.doi.org/10.1109/ICASSP.2009.4959506
A parametric instrument codec for very low bitrates. - In: 125th Audio Engineering Society convention 2008, (2008), S. 427-433
Graceful degradation for digital radio mondiale (DRM). - In: 125th Audio Engineering Society convention 2008, (2008), S. 589-595
A fast feature extraction system on compressed audio data. - In: 124th Audio Engineering Society convention 2008, (2008), S. 1383-1390
Subband conversion for feature extraction from compressed audio. - In: IEEE International Conference on Acoustics, Speech and Signal Processing, 2008, ISBN 978-1-4244-1483-3, (2008), S. 217-220
http://dx.doi.org/10.1109/ICASSP.2008.4517585
An evaluation of pre-processing algorithms for rhythmic pattern analysis. - In: 125th Audio Engineering Society convention 2008, (2008), S. 581-588
Reduced rate ultra low delay audio coder using multistage vector quantization. - In: Conference record of the Forty-First Asilomar Conference on Signals, Systems and Computers, 2007, ISBN 978-1-4244-2109-1, (2007), S. 2080-2084
Communication applications are usually delay restricted, especially for the instance of musicians playing over the Internet. This requires a one-way delay of maximum 25 msec and also a high audio quality is desired at feasible bit rates. The ultra low delay (ULD) audio coding structure is well suited to this application and we investigate further the application of multistage vector quantization (MSVQ) to reach a bit rate range below 64 Kb/s, in a scalable manner. Results at 32 Kb/s and 64 Kb/s show that the trained codebook MSVQ performs best, better than KLT normalization followed by a simulated Gaussian MSVQ or simulated Gaussian MSVQ alone. The results also show that there is only a weak dependence on the training data, and that we indeed converge to the perceptual quality of our previous ULD coder at 64 Kb/s.
https://doi.org/10.1109/ACSSC.2007.4487604
Low delay filterbanks for enhanced low delay audio coding. - In: IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2007, (2007), S. 235-238
Low delay perceptual audio coding has recently gained wide acceptance for high quality communication. While common schemes are based on the well-known Modified Discrete Cosine Transform (MDCT) filterbank, this paper describes novel coding algorithms that, for the first time, make use of dedicated low delay filterbanks, thus achieving improved coding efficiency while maintaining or even reducing the low codec delay. The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) coder currently under development within ISO/MPEG combines a traditional perceptual audio coding scheme with spectral band replication (SBR), both running in a delay-optimized fashion by using low delay filterbanks.
https://doi.org/10.1109/ASPAA.2007.4392985