Methods, encoder and decoder for linear predictive encoding and decoding of sound signals upon transition between frames having different sampling rates

• US 9,852,741 B2
• Filed: 04/02/2015
• Issued: 12/26/2017
• Est. Priority Date: 04/17/2014
• Status: Active Grant

• Current Assignee: VoiceAge EVS LLC (Government of ABU Dhabi)
• Original Assignee: VoiceAge Corporation
• Inventors: Salami, Redwan, Eksler, Vaclav
• Primary Examiner(s): Cordero, Marivelisse Santiago
• Assistant Examiner(s): Harris, Keara
• Application Number:
  US14/677,672
• Publication Number:
  US 20150302861A1
• Time in Patent Office: 999 Days
• Field of Search: None
• CPC Class Codes: G10L 19/06 Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
• CPC Class Codes: G10L 19/07 Line spectrum pair [LSP] vocoders
• CPC Class Codes: G10L 19/12 the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
• CPC Class Codes: G10L 19/167 Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• CPC Class Codes: G10L 19/173 Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding
• 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
• CPC Class Codes: G10L 19/26 Pre-filtering or post-filtering
• CPC Class Codes: G10L 2019/0002 Codebook adaptations
• CPC Class Codes: G10L 2019/0004 Design or structure of the codebook
• CPC Class Codes: G10L 2019/0016 Codebook for LPC parameters
• CPC Class Codes: G10L 21/038 using band spreading techniques
• CPC Class Codes: G10L 25/06 the extracted parameters being correlation coefficients

Abstract

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

Claims

1. A method for encoding a sound signal, comprising:

producing, in response to the sound signal, parameters for encoding the sound signal during successive sound signal processing frames, wherein the sound signal encoding parameters include linear predictive (LP) filter parameters, wherein producing the LP filter parameters comprises, when switching from a first one of the frames using an internal sampling rate S1 to a second one of the frames using an internal sampling rate S2, converting the LP filter parameters from the first frame from the internal sampling rate S1 to a the internal sampling rate S2, the and wherein converting the LP filter parameters from the first frame comprises;

computing, at the internal sampling rate SI, a power spectrum of a LP synthesis filter using the LP filter parameters;

modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2;

inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S2; and

using the autocorrelations to compute the LP filter parameters at the internal sampling rate S2; and

encoding the sound signal encoding parameters into a bitstream; and

wherein modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2 comprises;

if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on a ratio between S1 and S2;

if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.

2. The method as recited in claim 1, wherein the frames are divided into subframes, and wherein the method comprises computing LP filter parameters in each subframe of a current frame by interpolating LP filter parameters of the current frame at the internal sampling rate S2 with LP filter parameters of a past frame converted from the internal sampling rate S1 to the internal sampling rate S2.

3. The method as recited in claim 2, comprising forcing the current frame to an encoding mode that does not use a history of an adaptive codebook.

4. The method as recited in claim 2, comprising forcing a LP-parameter quantizer to use a non-predictive quantization method in the current frame.

5. The method as recited in claim 1, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

6. The method as recited in claim 1, comprising:

computing the power spectrum of the LP synthesis filter at K samples;

extending the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is less than the internal sampling rate S2; and

truncating the power spectrum of the LP synthesis filter to K(S2/S1) samples when the sampling rate S1 is greater than the sampling rate S2.

7. The method as recited in claim 1, comprising computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

8. The method as recited in claim 1, comprising inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

9. The method as recited in claim 1, comprising searching a fixed codebook using a reduced number of iterations.

10. A method for decoding a sound signal, comprising:

receiving a bitstream including sound signal encoding parameters in successive sound signal processing frames, wherein the sound signal encoding parameters include linear predictive (LP) filter parameters;

decoding from the bitstream the sound signal encoding parameters including the LP filter parameters during the successive sound signal processing frames, and producing from the decoded sound signal encoding parameters an LP synthesis filter excitation signal, wherein decoding the LP filter parameters comprises, when switching from a first one of the frames using an internal sampling rate S1 to a second one of the frames using an internal sampling rate S2, converting LP filter parameters from the first frame from the internal sampling rate S1 to the internal sampling rate S2, and wherein converting the LP filter parameters from the first frame comprises;

computing, at the internal sampling rate SI, a power spectrum of a LP synthesis filter using the received LP filter parameters;

modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2;

inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S2; and

using the autocorrelations to compute the LP filter parameters at the internal sampling rate S2;

synthesizing the sound signal using LP synthesis filtering in response to the decoded LP filter parameters and the LP synthesis filter excitation signal; and

wherein modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2 comprises;

if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on a ratio between S1 and S2;

if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.

11. The method as recited in claim 10, wherein the frames are divided into subframes, and wherein the method comprises computing LP filter parameters in each subframe of a current frame by interpolating LP filter parameters of the current frame at the internal sampling rate S2 with LP filter parameters of a past frame converted from the internal sampling rate S1 to the internal sampling rate S2.

12. The method as recited in claim 10, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

13. The method as recited in claim 10, comprising:

computing the power spectrum of the LP synthesis filter at K samples;

extending the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is less than the internal sampling rate S2; and

truncating the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is greater than the internal sampling rate S2.

14. The method as recited in claim 10, comprising computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

15. The method as recited in claim 10, comprising inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

16. The method as recited in claim 10, wherein a post filtering is skipped to reduce decoding complexity.

17. A device for encoding a sound signal, comprising:

at least one processor; and

a memory coupled to the processor and comprising non-transitory instructions that when executed cause the processor to;

produce, in response to the sound signal, parameters for encoding the sound signal during successive sound signal processing frames, wherein (a) the sound signal encoding parameters include linear predictive (LP) filter parameters, (b) for producing the LP filter parameters when switching from a first one of the frames using an internal sampling rate S1 to a second one of the frames using an internal sampling rate S2, the processor is configured to convert the LP filter parameters from the first frame from the internal sampling rate S1 to the internal sampling rate S2, and (c) for converting the LP filter parameters from the first frame, the processor is configured to;

compute, at the internal sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters,

modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2,

inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S2,

use the autocorrelations to compute the LP filter parameters at the internal sampling rate S2, and

encode the sound signal encoding parameters into a bitstream; and

wherein the processor is configured to;

extend the power spectrum of the LP synthesis filter based on a ratio between S1 and S2 if S1 is less than S2; and

truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.

18. The device as recited in claim 17, wherein the frames are divided into subframes, and wherein the processor is configured to compute LP filter parameters in each subframe of a current frame by interpolating LP filter parameters of the current frame at the internal sampling rate S2 with LP filter parameters of a past frame converted from the internal sampling rate S1 to the internal sampling rate S2.

19. The device as recited in claim 17, wherein the processor is configured to:

compute the power spectrum of the LP synthesis filter at K samples;

extend the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is less than the internal sampling rate S2; and

truncate the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is greater than the internal sampling rate S2.

20. The device as recited in claim 17, wherein the processor is configured to compute the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

21. The device as recited in claim 17, wherein the processor is configured to inverse transform the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

22. A device for decoding a sound signal, comprising:

at least one processor; and

a memory coupled to the processor and comprising non-transitory instructions that when executed cause the processor to;

receive a bitstream including sound signal encoding parameters in successive sound signal processing frames, wherein the sound signal encoding parameters include linear predictive (LP) filter parameters;

decode from the bitstream the sound signal encoding parameters including the LP filter parameters during the successive sound signal processing frames, and produce from the decoded sound signal encoding parameters an LP synthesis filter excitation signal, wherein (a) for decoding the LP filter parameters when switching from a first one of the frames using an internal sampling rate S1 to a second one of the frames using an internal sampling rate S2, the processor is configured to convert the LP filter parameters from the first frame from the internal sampling rate S1 to the internal sampling rate S2, and (b) for converting the LP filter parameters from the first frame, the processor is configured to;

compute, at the internal sampling rate SI, a power spectrum of a LP synthesis filter using the received LP filter parameters,

modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2,

inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S2, and

use the autocorrelations to compute the LP filter parameters at the internal sampling rate S2, and

synthesize the sound signal using LP synthesis filtering in response to the decoded LP filter parameters and the LP synthesis filter excitation signal, and wherein the processor is configured to;

extend the power spectrum of the LP synthesis filter based on a ratio between S1 and S2 if S1 is less than S2; and

truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.

23. The device as recited in claim 22, wherein the frames are divided into subframes, and wherein the processor is configured to compute LP filter parameters in each subframe of a current frame by interpolating LP filter parameters of the current frame at the internal sampling rate S2 with LP filter parameters of a past frame converted from the internal sampling rate S1 to the internal sampling rate S2.

24. The device as recited in claim 22, wherein the processor is configured to:

compute the power spectrum of the LP synthesis filter at K samples;

extend the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is less than the internal sampling rate S2; and

truncate the power spectrum of the LP synthesis filter to K(S2/S1) samples when the internal sampling rate S1 is greater than the internal sampling rate S2.

25. The device as recited in claim 22, wherein the processor is configured to compute the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

26. The device as recited in claim 22, wherein the processor is configured to inverse transfoiiii the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.