Audio encoder and decoder

• US 8,938,387 B2
• Filed: 05/28/2013
• Issued: 01/20/2015
• Est. Priority Date: 01/04/2008
• Status: Active Grant

• Current Assignee: Dolby International AB (Dolby Laboratories Incorporated)
• Original Assignee: Dolby Laboratories Licensing Corporation (Dolby Laboratories Incorporated)
• Inventors: Carlsson, Pontus, Samuelsson, Leif Jonas, Schug, Michael, Hedelin, Per
• Primary Examiner(s): Godbold, Douglas
• Application Number:
  US13/903,173
• Publication Number:
  US 20130282383A1
• Time in Patent Office: 602 Days
• Field of Search: 704200-203, 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 (LP) unit for filtering an audio signal based on a LP filter, the LP unit operating on a first frame length of the audio signal;

an adaptive length transformation unit for transforming a frame of the audio signal into a transform domain, the transformation being a Modified Discrete Cosine Transform (MDCT) operating on a variable second frame length;

a quantization unit for quantizing a MDCT-domain signal;

a gain curve generation unit for generating MDCT-domain gain curves based on magnitude responses of the LP filter; and

a mapping unit for mapping LP parameters to corresponding frames of the MDCT-domain signal;

wherein the mapping unit interpolates LP polynomials generated on a rate corresponding to the first frame length so as to match frames of the MDCT-domain signal generated on a rate corresponding to the second frame length for usage as perceptual weights in the quantization unit.

2. Audio coding system of claim 1, comprising:

a window sequence control unit for determining, for a block of the audio signal, the second frame lengths for overlapping MDCT windows.

3. Audio coding system according to claim 1, comprising a perceptual modeling unit that modifies a characteristic of the LP filter by chirping and/or tilting an LPC polynomial generated by the linear prediction unit for an LPC frame.

4. Audio coding system according to claim 1, comprising:

a frequency splitting unit for splitting the audio signal into a lowband component and a highband component; and

a highband encoder for encoding the highband component,

wherein the lowband component is input to the linear prediction unit and the transformation unit, and wherein the highband encoder is a spectral band replication encoder.

5. Audio coding system of claim 4, wherein the frequency splitting unit comprises a quadrature mirror filter bank and a quadrature mirror filter synthesis unit configured to downsample the audio signal.

6. Audio coding system of claim 5, wherein the boundary between the lowband and the highband is variable and the frequency splitting unit determines the cross-over frequency based on audio signal properties and/or encoder bandwidth requirements.

7. Audio coding system of claim 4, wherein the boundary between the lowband and the highband is variable and the frequency splitting unit determines the cross-over frequency based on audio signal properties and/or encoder bandwidth requirements.

8. Audio coding system according to claim 1, wherein the MDCT-domain gain curves are applied to MDCT-domain data.

9. Audio coding system according to claim 1, comprising:

a scalefactor estimation unit for estimating scalefactors to control the quantization noise of the quantization unit.

10. Audio coding system of claim 9, wherein the scalefactors are determined based on the MDCT-domain gain curves.

11. Audio coding system according to claim 1, comprising a parametric stereo unit for calculating a parametric stereo representation of left and right input channels.

12. Audio decoder comprising:

a de-quantization unit for reconstructing quantized MDCT lines received in an input bitstream;

an adaptive length Inverse MDCT transformation unit for inversely transforming a transform domain signal into a time domain signal, the inverse MDCT transformation unit operating on a variable frame length;

a gain curve generation unit for generating MDCT-domain gain curves based on magnitude responses of linear prediction filters, wherein parameters for the linear prediction filters are received in the bitstream and correspond to a first frame length; and

a mapping unit for mapping LP parameters to corresponding frames of the MDCT-domain signal;

wherein the mapping unit interpolates LP polynomials generated on a rate corresponding to the first frame length so as to match frames of the transform domain signal generated on a rate corresponding to the second frame length for usage as perceptual weights in the de-quantization unit.

13. Audio encoding method comprising the steps:

performing a linear prediction (LP) analysis on an audio signal, the LP analysis operating on a first frame length and generating LP parameters;

transforming a frame of the audio signal into a Modified Discrete Cosine Transform (MDCT)-domain, the MDCT operating on a variable second frame length;

quantizing a MDCT-domain signal;

generating MDCT-domain gain curves based on magnitude responses of the generated LP filters; and

mapping LP parameters to corresponding frames of the MDCT-domain signal;

wherein mapping comprises interpolating LP polynomials generated on a rate corresponding to the first frame length so as to match frames of the MDCT-domain signal generated on a rate corresponding to the second frame length for usage as perceptual weights for quantizing the MDCT-domain signal.

14. Audio decoding method comprising the steps:

reconstructing quantized MDCT lines received in an input bitstream;

inverse Modified Discrete Cosine-transforming (MDCT) a transform domain signal into a time domain signal, the inverse MDCT operating on a variable frame length;

generating MDCT-domain gain curves based on magnitude responses of linear prediction filters, wherein parameters for the linear prediction filters are received in the bitstream and correspond to a first frame length; and

mapping LP parameters to corresponding frames of the MDCT-domain signal;

wherein mapping comprises interpolating LP polynomials generated on a rate corresponding to the first frame length so as to match frames of the MDCT-domain signal generated on a rate corresponding to the second frame length for usage as perceptual weights for quantizing the MDCT-domain signal.