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A novel scheme for low bitrate unified speech and audio coding - MPEG RM0

: Neuendorf, M.; Gournay, P.; Multrus, M.; Lecomte, J.; Bessette, B.; Geiger, R.; Bayer, S.; Fuchs, G.; Hilpert, J.; Rettelbach, N.; Nagel, F.; Robilliard, J.; Salami, R.; Schuller, G.; Lefebvre, R.; Grill, B.

Audio Engineering Society -AES-:
126th Audio Engineering Society Convention 2009. CD-ROM : Munich, Germany, 7 - 10 May 2009
Red Hook, NY: Curran, 2009
ISBN: 978-1-615-67166-3
Paper 7713
Audio Engineering Society (Convention) <126, 2009, Munich>
Conference Paper
Fraunhofer IDMT ()
Fraunhofer IIS ()

Coding of speech signals at low bitrates, such as 16 kbps, has to rely on an efficient speech reproduction model to achieve reasonable speech quality. However, for audio signals not fitting to the model this approach generally fails. On the other hand, generic audio codecs, designed to handle any kind of audio signal, tend to show unsatisfactory results for speech signals, especially at low bitrates. To overcome this, a process was initiated by ISO/MPEG, aiming to standardize a new codec with consistent high quality for speech, music and mixed content over a broad range of bitrates. After a formal listening test evaluating several proposals MPEG has selected the best performing codec as the reference model for the standardization process. This paper describes this codec in detail and shows that the new reference model reaches the goal of consistent high quality for all signal types.