Scalable and embedded codec for speech and audio signals
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Abstract
A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.
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Citations
50 Claims
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21. A system for embedded coding of audio signals comprising:
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(a) a frame extractor for dividing an input signal into a plurality of signal frames corresponding to successive time intervals;
(b) means for providing parametric representations of the signal in each frame, said parametric representations being based on a signal model;
(c) means for providing a first encoded data portion corresponding to a user-specified parametric representation, which first encoded data portion contains information sufficient to reconstruct a representation of the input signal;
(d) means for providing one or more secondary encoded data portions of the user-selected parametric representation; and
(e) means for providing an embedded output signal based at least on said first encoded data portion and said one or more secondary encoded data portions of the user-selected parametric representation. - View Dependent Claims (22, 23, 24, 25, 26, 27, 28)
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29. A method for multistage vector quantization of signals comprising:
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(a) passing an input signal through a first stage of a multistage vector quantizer having a predetermined set of codebook vectors, each vector corresponding to a Voronoi cell, to obtain error vectors corresponding to differences between a codebook vector and an input signal vector falling within a Voronoi cell;
(b) determining probability density functions (pdfs) for the error vectors in at least two Voronoi cells;
(c) transforming error vectors using a transformation based on the pdfs determined for said at least two Voronoi cells; and
(d) passing transformed error vectors through at least a second stage of the multistage vector quantizer to provide a quantized output signal. - View Dependent Claims (30, 31, 32, 33, 34, 35, 36, 37)
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38. A system for processing audio signals comprising;
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(a) a frame extractor for dividing an input audio signal into a plurality of signal frames corresponding to successive time intervals;
(b) a frame mode classifier for determining if the signal in a frame is in a transition state;
(c) a processor for extracting parameters of the signal in a frame receiving input from said classifier, wherein for frames the signal of which is determined to be in said transition state said extracted parameters include phase information; and
(d) a multi-mode coder in which extracted parameters of the signal in a frame are processed in at least two distinct paths dependent on whether the frame signal is determined to be in a transition state. - View Dependent Claims (39, 40, 41, 42, 43)
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44. A system for processing audio signals comprising:
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(a) a frame extractor for dividing an input signal into a plurality of signal frames corresponding to successive time intervals;
(b) means for providing a parametric representation of the signal in each frame, said parametric representation being based on a signal model;
(c) a non-linear processor for providing refined estimates of parameters of the parametric representation of the signal in each frame; and
(d) means for encoding said refined parameter estimates. - View Dependent Claims (45, 46, 47, 48, 49, 50)
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Specification