Audio encoder and decoder

• US 8,924,201 B2
• Filed: 05/24/2013
• Issued: 12/30/2014
• Est. Priority Date: 01/04/2008
• Status: Active Grant

• Current Assignee: Dolby International AB (Dolby Laboratories Incorporated)
• Original Assignee: Dolby International AB (Dolby Laboratories Incorporated)
• Inventors: Hedelin, Per, Carlsson, Pontus, Samuelsson, Leif Jonas, Schug, Michael
• Primary Examiner(s): Godbold, Douglas
• Application Number:
  US13/901,960
• Publication Number:
  US 20130282382A1
• Time in Patent Office: 585 Days
• Field of Search: 704200-201, 704/219, 704500-504
• US Class Current: 704/219
• CPC Class Codes: G10L 19/008 Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
• CPC Class Codes: G10L 19/032 Quantisation or dequantisation of spectral components
• CPC Class Codes: G10L 19/035 Scalar quantisation
• CPC Class Codes: G10L 19/26 Pre-filtering or post-filtering

Abstract

The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Claims

1. Audio coding system comprising:

a linear prediction unit for filtering an input signal based on an adaptive filter;

a transformation unit for transforming a frame of the filtered input signal into a transform domain; and

a quantization unit for quantizing the transform domain signal;

wherein the quantization unit decides, based on input signal characteristics to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer.

2. Audio coding system according to claim 1, wherein the model in the model-based quantizer is adaptive and variable over time.

3. Audio coding system according to claim 1, wherein the quantization unit decides how to encode the transform domain signal based on the frame size applied by the transformation unit.

4. Audio coding system according to claim 1, wherein the quantization unit comprises a frame size comparator and is configured to encode a transform domain signal for a frame with a frame size smaller than a threshold value by means of a model-based entropy constrained quantization.

5. Audio coding system according to claim 1, comprising a quantization step size control unit for determining the quantization step sizes of components of the transform domain signal based on linear prediction and long term prediction parameters.

6. Audio coding system of claim 5, wherein the quantization step size is determined frequency depending, and the quantization step size control unit determines the quantization step sizes based on at least one of:

the polynomial of the adaptive filter, a coding rate control parameter, a long term prediction gain value, and an input signal variance.

7. Audio coding system according to claim 1, wherein the quantization unit comprises uniform scalar quantizers for quantizing the transform domain signal components, each scalar quantizer applying a uniform quantization, based on a probability model, to a MDCT line.

8. Audio coding system according to claim 7, wherein the quantization unit comprises a random offset insertion unit for inserting a random offset into the uniform scalar quantizers, the random offset insertion unit configured to determine the random offset based on an optimization of a quantization distortion.

9. Audio coding system according to claim 7, wherein the quantization unit comprises an arithmetic encoder for encoding quantization indices generated by the uniform scalar quantizers.

10. Audio coding system according to claim 7, wherein the quantization unit comprises a residual quantizer for quantizing a residual quantization signal resulting from the uniform scalar quantizers.

11. Audio coding system according to claim 7, wherein the quantization unit uses minimum mean squared error and/or center point quantization reconstruction points.

12. Audio coding system according to claim 7, wherein the quantization unit comprises a dynamic reconstruction point unit that determines a quantization reconstruction point based on an interpolation between a probability model center point and a minimum mean squared error point.

13. Audio coding system according to claim 7, wherein the quantization unit applies a perceptual weighting in the transform domain when determining the quantization distortion, the perceptual weights being derived from linear prediction parameters.

14. Audio decoder comprising:

a model-based de-quantization unit for de-quantizing a frame of an input bitstream;

an inverse transformation unit for inversely transforming a transform domain signal; and

a linear prediction unit for filtering the inversely transformed transform domain signal;

wherein the de-quantization unit comprises a non-model based and a model based de-quantizer.

15. Audio decoder of claim 14, wherein the de-quantization unit decides a de-quantization strategy based on control data for the frame.

16. Audio decoder of claim 15, wherein the de-quantization control data is received with the bitstream or derived from received data.

17. Audio decoder of claim 14, wherein the de-quantization unit decides the de-quantization strategy based on the transform size of the frame.

18. Audio decoder of claim 14, wherein the de-quantization unit comprises adaptive reconstruction points.

19. Audio decoder of claim 18, wherein the de-quantization unit comprises uniform scalar de-quantizers that are configured to use two de-quantization reconstruction points per quantization interval, in particular a midpoint and a MMSE reconstruction point.

20. Audio decoder of claim 14, wherein the de-quantization unit comprises at least one adaptive probability model.

21. Audio decoder of claim 14, wherein the de-quantization unit uses a model based quantizer in combination with arithmetic coding.

22. Audio decoder of claim 14, wherein the de-quantization unit is configured to adapt the de-quantization as a function of the transmitted signal characteristics.

23. Audio decoding method comprising the steps:

de-quantizing a frame of an input bitstream;

inversely transforming a transform domain signal; and

linear prediction filtering the inversely transformed transform domain signal;

wherein the de-quantization is using a non-model and a model-based quantizer and wherein the method is performed by one or more computing devices.