Abstract
We describe novel speech/audio coding technique designed to operate at medium bit-rates. Unlike classical state-of-the-art coders that are based on short-term spectra, our approach uses relatively long temporal segments of audio signal in critical-band-sized sub-bands. We apply auto-regressive model to approximate Hilbert envelopes in frequency sub-bands. Residual signals (Hilbert carriers) are demodulated and thresholding functions are applied in spectral domain. The Hilbert envelopes and carriers are quantized and transmitted to the decoder. Our experiments focused on designing speech/audio coder to provide broadcast radio-like quality audio around 15 − 25kbps. Obtained objective quality measures, carried out on standard speech recordings, were compared to the state-of-the-art 3GPP-AMR speech coding system.
This work was partially supported by grants from ICSI Berkeley, USA; the Swiss National Center of Competence in Research (NCCR) on “Inter active Multi-modal Information Management (IM)2”; managed by the IDIAP Research Institute on behalf of the Swiss Federal Authorities, and by the European Commission 6th Framework DIRAC Integrated Project.
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Motlicek, P., Hermansky, H., Ganapathy, S., Garudadri, H. (2007). Non-uniform Speech/Audio Coding Exploiting Predictability of Temporal Evolution of Spectral Envelopes. In: Matoušek, V., Mautner, P. (eds) Text, Speech and Dialogue. TSD 2007. Lecture Notes in Computer Science(), vol 4629. Springer, Berlin, Heidelberg. https://doi.org/10.1007/978-3-540-74628-7_46
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DOI: https://doi.org/10.1007/978-3-540-74628-7_46
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