Scalable And Embedded Codec For Speech And Audio Signals

• US 20150302859A1
• Filed: 05/04/2015
• Published: 10/22/2015
• Est. Priority Date: 09/23/1998
• Status: Abandoned Application

• Current Assignee: Alcatel-Lucent SA (Nokia Corporation)
• Original Assignee: Alcatel-Lucent SA (Nokia Corporation)
• Inventors: Aguilar, Joseph Gerard, Campana, David A., Dunn, Robert B., McAulay, Robert J., Sun, Xiaoquin, Wang, Wei, Zopf, Robert W., Chen, Juin-Hwey (Raymond), Watkins, Craig
• Application Number:
  US14/703,261
• Publication Number:
  US 20150302859A1
• Time in Patent Office: Days
• US Class Current: 1/1
• CPC Class Codes: G10L 19/002 Dynamic bit allocation for perceptual audio coders G10L19/032
• CPC Class Codes: G10L 19/093 using sinusoidal excitation models
• CPC Class Codes: G10L 19/24 Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

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.

Claims

1. A system for embedded coding of audio signals comprising:

(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 one said first encoded data portion and said one or more secondary encoded data portions of the user-selected parametric representation.

2. The system of claim 1 further comprising:

(f) means for providing representations of the signal in each frame, which are not based on a signal model.

3. The system of claim 2 further comprising:

(g) means for selecting a specific one from the representations in (b) and (f) based on user-selected constraints.

4. The system of claim 1 wherein said means for providing parametric representations of the signal in each frame comprises a pitch detector for computing a first estimate of the pitch of a signal in each frame;

means for determining parameters of sinusoids representing the signal in each frame; and

a spectrum envelope encoder for encoding the shape of the envelope of the signal in each frame.

5. The system of claim 1 wherein said means for providing an embedded output signal comprises a bit stream assembler for providing an output bit stream containing user-specified information about parameters of at least one sinusoid in the spectrum of the input signal, and about parameters representing a spectrum envelope of the signal in each frame.

6. The system of claim 1 further comprising means for decoding the embedded output signal.

7. The system of claim 6 wherein said means for decoding operates at a sampling frequency different from a sampling frequency of the input signal.

8. The system of claim 1 wherein said means for providing an embedded output signal comprises means for assembling data packets suitable for transmission over a packet-switched network.

9. A system for processing audio signals comprising:

(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 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.

10. The system of claim 9 wherein said refined estimates comprises an estimate of the pitch.

11. The system of claim 9 wherein said refined estimates comprises an estimate of a voicing parameter for the input speech signal.

12. The system of claim 9 wherein said refined estimates comprises an estimate of a pitch onset time for an input speech signal.

13. The system of claim 9 wherein said non-linear processor computes the maximum of a correlation function of the input signal over a set of complex frequencies.

14. The system of claim 13 wherein the computation is done iteratively.

15. The system of claim 9 wherein a measure of voicing for the input signal is computed as