IP Library Granted Patent US 10,672,408
Granted Patent B2
US 10,672,408 · App. 15/752,699 · Granted Jun 2, 2020

Audio decoder and decoding method

Inventors: Dirk Jeroen Breebaart (Ultimo, AU); David Matthew Cooper (Carlton, AU); Leif Jonas Samuelsson (Sundbyberg, SE)
Assignees: Dolby Laboratories Licensing Corporation; Dolby International AB
G10L19/0212G10L19/008G10L19/0204H04S7/308H04R2460/03H04S2400/01H04S2420/01H04S2420/03H04S2420/07
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Quick Facts
Patent No.
US 10,672,408
App. No.
15/752,699
Granted
Jun 2, 2020
Kind
B2
Abstract

A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Claims (51)

1. A method for encoding a second presentation of audio channels or objects as an encoded audio signal, the method comprising the steps of:

(a) providing base signals, said base signals representing a first presentation of the audio channels or objects;

(b) providing transformation parameters for transforming the base signals of said first presentation into output signals of said second presentation; said transformation parameters including at least high frequency transformation parameters specified for a higher frequency band and low frequency transformation parameters specified for a lower frequency band, with the low frequency transformation parameters including a set of multi-tap convolution matrix parameters for convolving low frequency components of the base signals with the low frequency transformation parameters to produce convolved low frequency components and the high frequency transformation parameters including a set of parameters of a stateless matrix for multiplying high frequency components of the base signals with the high frequency transformation parameters to produce multiplied high frequency components; the first presentation being for loudspeaker playback and the second presentation being for headphone playback, or vice versa; and

(c) combining said base signals and said transformation parameters to form said encoded audio signal.

2. The method of claim 1 wherein said multi-tap convolution matrix parameters are indicative of a finite impulse response (FIR) filter, include at least one coefficient that is complex valued, and/or are utilized to process a low-frequency band.

3. The method of claim 1 wherein said base signals are divided up into a series of temporal segments, and transformation parameters are provided for each temporal segment.

4. The method of claim 1 , wherein

providing the base signals comprises determining the base signals from the audio channels or objects using first rendering parameters;

the method comprises determining desired output signals for the second presentation from the audio channels or objects using second rendering parameters; and

providing the transformation parameters comprises determining the transformation parameters by minimizing a deviation of the output signals from the desired output signals.

5. The method of claim 4 , wherein determining the transformation parameters comprises

determining sub-band-domain base signals for a number B of frequency bands using an encoder filter bank;

determining sub-band-domain desired output signals for the B frequency bands using the encoder filter bank; and

determining a same set of multi-tap convolution matrix parameters for at least two adjacent frequency bands of the B frequency bands.

6. The method of claim 5 , wherein

the encoder filter bank comprises a hybrid filter bank which provides low frequency bands of the B frequency bands having a higher frequency resolution than high frequency bands of the B frequency bands; and

the at least two adjacent frequency bands are low frequency bands.

7. The method of claim 6 , wherein determining the transformation parameters comprises determining a same real-valued transformation parameter for at least two adjacent high frequency bands.

8. The method of claim 1 wherein

the high frequency transformation parameters do not modify a signal phase of the base signals, and

the low frequency transformation parameters do modify the signal phase of the base signal.

9. The method of claim 1 wherein

said high frequency transformation parameters include high frequency audio matrix coefficients for matrix manipulation of a high frequency portion of said base signals, and

wherein for a medium frequency portion of the high frequency portion of said base signals, the matrix manipulation includes complex-valued transformation parameters.

10. A computer readable non-transitory storage medium including program instructions for the operation of a computer in accordance with the method of claim 1 .

11. A decoder for decoding an encoded audio signal, the encoded audio signal including:

a first presentation including audio base signals for reproduction of the encoded audio signal in a first audio presentation format; and

transformation parameters, for transforming said audio base signals in said first presentation format, into output signals of a second presentation format, said transformation parameters comprising high frequency transformation parameters specified for a higher frequency band and low frequency transformation parameters specified for a lower frequency band, with said low frequency transformation parameters including multi tap convolution matrix parameters and the high frequency transformation parameters including a set of parameters of a stateless matrix, the first presentation format being for loudspeaker playback and the second presentation format being for headphone playback, or vice versa,

the decoder including:

first separation unit for separating the audio base signals, and the transformation parameters,

a first matrix multiplication unit for applying said multi tap convolution matrix parameters to low frequency components of the audio base signals; to apply a convolution to the low frequency components, producing convolved low frequency components;

a second matrix multiplication unit for applying said high frequency transformation parameters to high frequency components of the audio base signals to produce scalar high frequency components; and

an output filter bank for combining said convolved low frequency components and said scalar high frequency components to produce a time domain output signal of said second presentation format.

12. The decoder of claim 11 wherein said first matrix multiplication unit modifies a phase of the low frequency components of the audio base signals.

13. The decoder of claim 11 wherein said multi tap convolution matrix transformation parameters are complex valued, one or more of said high frequency transformation parameters is complex-valued, and/or one or more of said high frequency transformation parameters is real-valued.

14. The decoder of claim 11 , further comprising filters for separating the audio base signals into said low frequency components and said high frequency components.

15. A method of decoding an encoded audio signal, the encoded audio signal including:

a first presentation including audio base signals for reproduction of the encoded audio signal in a first audio presentation format; and

transformation parameters, for transforming said audio base signals in said first presentation format, into output signals of a second presentation format, said transformation parameters comprising high frequency transformation parameters specified for a higher frequency band and low frequency transformation parameters specified for a lower frequency band, with said low frequency transformation parameters including multi tap convolution matrix parameters and the high frequency transformation parameters including a set of parameters of a stateless matrix, the first presentation format being for loudspeaker playback and the second presentation format being for headphone playback, or vice versa, the method including the steps of:

convolving low frequency components of the audio base signals with the low frequency transformation parameters to produce convolved low frequency components;

multiplying high frequency components of the audio base signals with the high frequency transformation parameters to produce multiplied high frequency components;

combining said convolved low frequency components and said multiplied high frequency components to produce output audio signal frequency components for the second presentation format.

16. The method of claim 15 , wherein said encoded audio signal comprises multiple temporal segments, said method further includes the steps of:

interpolating transformation parameters of multiple temporal segments of the encoded audio signal to produce interpolated transformation parameters, including interpolated low frequency transformation parameters; and

convolving multiple temporal segments of the low frequency components of the audio base signals with the interpolated low frequency transformation parameters to produce multiple temporal segments of said convolved low frequency components.

17. The method as claimed in either claim 16 wherein said interpolating utilizes an overlap and add method of the multiple sets of intermediate convolved low frequency components.

18. The method of claim 15 wherein the transformation parameters of said encoded audio signal are time varying, and said convolving low frequency components of the audio base signals includes the steps of:

convolving the low frequency components of the audio base signals with the low frequency transformation parameters for multiple temporal segments to produce multiple sets of intermediate convolved low frequency components; and

interpolating the multiple sets of intermediate convolved low frequency components to produce said convolved low frequency components.

19. The method of claim 15 , further comprising filtering the audio base signals into said low frequency components and said high frequency components.

20. A computer readable non-transitory storage medium including program instructions for the operation of a computer in accordance with the method of claim 15 .

Assignments (1)
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Mar 28, 2018
From: BREEBAART, DIRK JEROEN; COOPER, DAVID MATTHEW; SAMUELSSON, LEIF JONAS
To: DOLBY LABORATORIES LICENSING CORPORATION; DOLBY INTERNATIONAL AB
Reel/Frame 045760/0917 →
Priority Claims (1)
EP 15189008 · Oct 8, 2015 · regional
Continuity (2)
Provisional Application 62209742 · Aug 25, 2015
Related Publication 20180233156A1 · Aug 16, 2018
Cited By (1)
US 12,646,322