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

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

sampling the sound signal during successive sound signal processing frames;

producing, in response to the sampled sound signal, parameters for encoding the sound signal during the successive frames, wherein the sound signal encoding parameters include linear predictive (LP) filter parameters, wherein producing the LP filter parameters comprises, upon switching from a first one of the frames using an internal sampling rate S 1 to a second one of the frames using an internal sampling rate S 2 , converting LP filter parameters from the first frame from the internal sampling rate S 1 to the internal sampling rate S 2 , and wherein converting the LP filter parameters from the first frame comprises:

computing, at the internal sampling rate S 1 , 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 S 1 to the internal sampling rate S 2 ;

inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S 2 ; and

using the autocorrelations to compute the LP filter parameters at the internal sampling rate S 2 ; and

encoding the sound signal encoding parameters into a bitstream;

wherein the frames are divided into subframes, and

wherein the method further 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 S 2 with LP filter parameters of a past frame converted from the internal sampling rate S 1 to the internal sampling rate S 2 .

2. The method as recited in claim 1 , wherein modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S 1 to the internal sampling rate S 2 comprises:

if S 1 is less than S 2 , extending the power spectrum of the LP synthesis filter based on a ratio between S 1 and S 2 ; and

if S 1 is larger than S 2 , truncating the power spectrum of the LP synthesis filter based on the ratio between S 1 and S 2 .

3. The method as recited in claim 1 , further 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 1 , further 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 , further comprising:

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

extending the power spectrum of the LP synthesis filter to K(S 2 /S 1 ) samples when the internal sampling rate S 1 is less than the internal sampling rate S 2 ; and

truncating the power spectrum of the LP synthesis filter to K(S 2 /S 1 ) samples when the sampling rate S 1 is greater than the sampling rate S 2 .

7. The method as recited in claim 1 , further 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 , further 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 , further 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, upon switching from a first one of the frames using an internal sampling rate S 1 to a second one of the frames using an internal sampling rate S 2 , converting the LP filter parameters from the first frame from the internal sampling rate S 1 to the internal sampling rate S 2 , and wherein converting the LP filter parameters from the first frame comprises:

computing, at the internal sampling rate S 1 , 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 S 1 to the internal sampling rate S 2 ;

inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S 2 ; and

using the autocorrelations to compute the LP filter parameters at the internal sampling rate S 2 ; and

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

wherein the frames are divided into subframes, and

wherein the method further 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 S 2 with LP filter parameters of a past frame converted from the internal sampling rate S 1 to the internal sampling rate S 2 .

11. The method as recited in claim 10 , wherein modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S 1 to the internal sampling rate S 2 comprises:

if S 1 is less than S 2 , extending the power spectrum of the LP synthesis filter based on a ratio between S 1 and S 2 ;

if S 1 is larger than S 2 , truncating the power spectrum of the LP synthesis filter based on the ratio between S 1 and S 2 .

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 , further comprising:

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

extending the power spectrum of the LP synthesis filter to K(S 2 /S 1 ) samples when the internal sampling rate S 1 is less than the internal sampling rate S 2 ; and

truncating the power spectrum of the LP synthesis filter to K(S 2 /S 1 ) samples when the internal sampling rate S 1 is greater than the internal sampling rate S 2 .

14. The method as recited in claim 10 , further 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 , further 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 upon switching from a first one of the frames using an internal sampling rate S 1 to a second one of the frames using an internal sampling rate S 2 , the processor is configured to convert the LP filter parameters from the first frame from the internal sampling rate S 1 to the internal sampling rate S 2 , and (c) for converting the LP filter parameters from the first frame, the processor is configured to:

compute, at the internal sampling rate S 1 , 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 S 1 to the internal sampling rate S 2 ,

inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S 2 , and

use the autocorrelations to compute the LP filter parameters at the internal sampling rate S 2 , and

encode the sound signal encoding parameters into a bitstream;

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 S 2 with LP filter parameters of a past frame converted from the internal sampling rate S 1 to the internal sampling rate S 2 .

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

extend the power spectrum of the LP synthesis filter based on a ratio between S 1 and S 2 if S 1 is less than S 2 ; and

truncate the power spectrum of the LP synthesis filter based on the ratio between S 1 and S 2 if S 1 is larger than S 2 .

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(S 2 /S 1 ) samples when the internal sampling rate S 1 is less than the internal sampling rate S 2 ; and

truncate the power spectrum of the LP synthesis filter to K(S 2 /S 1 ) samples when the internal sampling rate S 1 is greater than the internal sampling rate S 2 .

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 upon switching from a first one of the frames using an internal sampling rate S 1 to a second one of the frames using an internal sampling rate S 2 , the processor is configured to convert the LP filter parameters from the first frame from the internal sampling rate S 1 to the internal sampling rate S 2 , and (b) for converting the LP filter parameters from the first frame, the processor is configured to:

compute, at the internal sampling rate S 1 , 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 S 1 to the internal sampling rate S 2 ,

inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S 2 , and

use the autocorrelations to compute the LP filter parameters at the internal sampling rate S 2 , and

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

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 S 2 with LP filter parameters of a past frame converted from the internal sampling rate S 1 to the internal sampling rate S 2 .

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

extend the power spectrum of the LP synthesis filter based on a ratio between S 1 and S 2 if S 1 is less than S 2 ; and

truncate the power spectrum of the LP synthesis filter based on the ratio between S 1 and S 2 if S 1 is larger than S 2 .

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(S 2 /S 1 ) samples when the internal sampling rate S 1 is less than the internal sampling rate S 2 ; and

truncate the power spectrum of the LP synthesis filter to K(S 2 /S 1 ) samples when the internal sampling rate S 1 is greater than the internal sampling rate S 2 .

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 25 , wherein the processor is configured to inverse transform the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.