Anthropomorphic coding of speech and audio: a model inversion approach
EURASIP Journal on Applied Signal Processing
On the recognition of cochlear implant-like spectrally reduced speech with MFCC and HMM-based ASR
IEEE Transactions on Audio, Speech, and Language Processing
Magnitude-Sign split quantization for bandwidth scalable wideband speech codec
PCM'05 Proceedings of the 6th Pacific-Rim conference on Advances in Multimedia Information Processing - Volume Part I
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For speech coders which fall within the class of waveform coders, the reconstructed signal approaches the original with increasing bit rate. In such coders, the distortion criterion generally operates on the speech signal or a signal obtained by adaptive linear filtering of the speech signal. To satisfy computational and delay constraints, the distortion criterion must be reduced to a very simple approximation of the auditory system. This drawback of conventional approaches motivates a new speech coding paradigm in which the coding is performed in a domain where the single-letter squared-error criterion forms an accurate representation of perception. The new paradigm requires a model of the auditory periphery which is accurate, can be be inverted with relatively low computational effort, and which represents the signal with relatively few parameters. We develop such a model of the auditory periphery and discuss its suitability for speech coding. The results indicate that the new paradigm in general and our auditory model in particular form a promising basis for the coding of both speech and audio at low bit rates.