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

TECHNICAL FIELD

0001The present disclosure relates to the field of sound coding. More specifically, the present disclosure relates to methods, an encoder and a decoder for linear predictive encoding and decoding of sound signals upon transition between frames having different sampling rates.

BACKGROUND

0002The demand for efficient digital wideband speech/audio encoding techniques with a good subjective quality/bit rate trade-off is increasing for numerous applications such as audio/video teleconferencing, multimedia, and wireless applications, as well as Internet and packet network applications. Until recently, telephone bandwidths in the range of 200-3400 Hz were mainly used in speech coding applications. However, there is an increasing demand for wideband speech applications in order to increase the intelligibility and naturalness of the speech signals. A bandwidth in the range 50-7000 Hz was found sufficient for delivering a face-to-face speech quality. For audio signals, this range gives an acceptable audio quality, but is still lower than the CD (Compact Disk) quality which operates in the range 20-20000 Hz.

0003A speech encoder converts a speech signal into a digital bit stream that is transmitted over a communication channel (or stored in a storage medium). The speech signal is digitized (sampled and quantized with usually 16-bits per sample) and the speech encoder has the role of representing these digital samples with a smaller number of bits while maintaining a good subjective speech quality. The speech decoder or synthesizer operates on the transmitted or stored bit stream and converts it back to a sound signal.

0004One of the best available techniques capable of achieving a good quality/bit rate trade-off is the so-called CELP (Code Excited Linear Prediction) technique. According to this technique, the sampled speech signal is processed in successive blocks of <i>L</i> samples usually called <i>frames</i> where <i>L</i> is some predetermined number (corresponding to 10-30 ms of speech). In CELP, an LP (Linear Prediction) synthesis filter is computed and transmitted every frame. The <i>L</i>-sample frame is further divided into smaller blocks called <i>subframes</i> of <i>N</i> samples, where <i>L</i>=<i>kN</i> and <i>k</i> is the number of subframes in a frame (<i>N</i> usually corresponds to 4-10 ms of speech). An excitation signal is determined in each subframe, which usually comprises two components: one from the past excitation (also called pitch contribution or adaptive codebook) and the other from an innovative codebook (also called fixed codebook). This excitation signal is transmitted and used at the decoder as the input of the LP synthesis filter in order to obtain the synthesized speech.

0005To synthesize speech according to the CELP technique, each block of <i>N</i> samples is synthesized by filtering an appropriate codevector from the innovative codebook through time-varying filters modeling the spectral characteristics of the speech signal. These filters comprise a pitch synthesis filter (usually implemented as an adaptive codebook containing the past excitation signal) and an LP synthesis filter. At the encoder end, the synthesis output is computed for all, or a subset, of the codevectors from the innovative codebook (codebook search). The retained innovative codevector is the one producing the synthesis output closest to the original speech signal according to a perceptually weighted distortion measure. This perceptual weighting is performed using a so-called perceptual weighting filter, which is usually derived from the LP synthesis filter.

0006In LP-based coders such as CELP, an LP filter is computed then quantized and transmitted once per frame. However, in order to insure smooth evolution of the LP synthesis filter, the filter parameters are interpolated in each subframe, based on the LP parameters from the past frame. The LP filter parameters are not suitable for quantization due to filter stability issues. Another LP representation more efficient for quantization and interpolation is usually used. A commonly used LP parameter representation is the line spectral frequency (LSF) domain.

0007In wideband coding the sound signal is sampled at 16000 samples per second and the encoded bandwidth extended up to 7 kHz. However, at low bit rate wideband coding (below 16 kbit/s) it is usually more efficient to down-sample the input signal to a slightly lower rate, and apply the CELP model to a lower bandwidth, then use bandwidth extension at the decoder to generate the signal up to 7 kHz. This is due to the fact that CELP models lower frequencies with high energy better than higher frequency. So it is more efficient to focus the model on the lower bandwidth at low bit rates. AMR-WB standard (Reference [1]) is such a coding example, where the input signal is down-sampled to 12800 samples per second, and the CELP encodes the signal up to 6.4 kHz. At the decoder bandwidth extension is used to generate a signal from 6.4 to 7 kHz. However, at bit rates higher than 16 kbit/s it is more efficient to use CELP to encode the signal up to 7 kHz, since there are enough bits to represent the entire bandwidth.

0008Most recent coders are multi-rate coders covering a wide range of bit rates to enable flexibility in different application scenarios. Again AMR-WB is such an example, where the encoder operates at bit rates from 6.6 to 23.85 kbit/s. In multi-rate coders the codec should be able to switch between different bit rates on a frame basis without introducing switching artefacts. In AMR-WB this is easily achieved since all the rates use CELP at 12.8 kHz internal sampling rate. However, in a recent coder using 12.8 kHz sampling at bit rates below 16 kbit/s and 16 kHz sampling at bit rates higher than 16 kbits/s, the issues related to switching the bit rate between frames using different sampling rates need to be addressed. The main issues are in the LP filter transition, and in the memory of the synthesis filter and adaptive codebook.

0009Therefore there remains a need for efficient methods for switching LP-based codecs between two bit rates with different internal sampling rates.

SUMMARY

0010According to the present disclosure, there is provided a method implemented in a sound signal encoder for converting linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal 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.

0011According to the present disclosure, there is also provided a method implemented in a sound signal decoder for converting received linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the received 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.

0012According to the present disclosure, there is also provided a device for use in a sound signal encoder for converting linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2. The device comprises a processor configured to: <ul id="ul0001" list-style="bullet"><li>compute, at the sampling rate S1, a power spectrum of a LP synthesis filter using the received LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2,</li><li>inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the sampling rate S2, and</li><li>use the autocorrelations to compute the LP filter parameters at the sampling rate S2.</li></ul>

0013The present disclosure further relates to a device for use in a sound signal decoder for converting received linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2. The device comprises a processor configured to: <ul id="ul0002" list-style="bullet"><li>compute, at the sampling rate S1, a power spectrum of a LP synthesis filter using the received LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2,</li><li>inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the sampling rate S2, and</li><li>use the autocorrelations to compute the LP filter parameters at the sampling rate S2.</li></ul>

0014The foregoing and other objects, advantages and features of the present disclosure will become more apparent upon reading of the following non-restrictive description of an illustrative embodiment thereof, given by way of example only with reference to the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

0015In the appended drawings: <ul id="ul0003" list-style="none"><li><figref idref="f0001">Figure 1</figref> is a schematic block diagram of a sound communication system depicting an example of use of sound encoding and decoding;</li><li><figref idref="f0002">Figure 2</figref> is a schematic block diagram illustrating the structure of a CELP-based encoder and decoder, part of the sound communication system of <figref idref="f0001">Figure 1</figref>;</li><li><figref idref="f0003">Figure 3</figref> illustrates an example of framing and interpolation of LP parameters;</li><li><figref idref="f0004">Figure 4</figref> is a block diagram illustrating an embodiment for converting the LP filter parameters between two different sampling rates; and</li><li><figref idref="f0005">Figure 5</figref> is a simplified block diagram of an example configuration of hardware components forming the encoder and/or decoder of <figref idref="f0001">Figures 1</figref> and <figref idref="f0002">2</figref>.</li></ul>

DETAILED DESCRIPTION

0016The non-restrictive illustrative embodiment of the present disclosure is concerned with a method and a device for efficient switching, in an LP-based codec, between frames using different internal sampling rates. The switching method and device can be used with any sound signals, including speech and audio signals. The switching between 16 kHz and 12.8 kHz internal sampling rates is given by way of example, however, the switching method and device can also be applied to other sampling rates.

0017<figref idref="f0001">Figure 1</figref> is a schematic block diagram of a sound communication system depicting an example of use of sound encoding and decoding. A sound communication system 100 supports transmission and reproduction of a sound signal across a communication channel 101. The communication channel 101 may comprise, for example, a wire, optical or fibre link. Alternatively, the communication channel 101 may comprise at least in part a radio frequency link. The radio frequency link often supports multiple, simultaneous speech communications requiring shared bandwidth resources such as may be found with cellular telephony. Although not shown, the communication channel 101 may be replaced by a storage device in a single device embodiment of the communication system 101 that records and stores the encoded sound signal for later playback.

0018Still referring to <figref idref="f0001">Figure 1</figref>, for example a microphone 102 produces an original analog sound signal 103 that is supplied to an analog-to-digital (A/D) converter 104 for converting it into an original digital sound signal 105. The original digital sound signal 105 may also be recorded and supplied from a storage device (not shown). A sound encoder 106 encodes the original digital sound signal 105 thereby producing a set of encoding parameters 107 that are coded into a binary form and delivered to an optional channel encoder 108. The optional channel encoder 108, when present, adds redundancy to the binary representation of the coding parameters before transmitting them over the communication channel 101. On the receiver side, an optional channel decoder 109 utilizes the above mentioned redundant information in a digital bit stream 111 to detect and correct channel errors that may have occurred during the transmission over the communication channel 101, producing received encoding parameters 112. A sound decoder 110 converts the received encoding parameters 112 for creating a synthesized digital sound signal 113. The synthesized digital sound signal 113 reconstructed in the sound decoder 110 is converted to a synthesized analog sound signal 114 in a digital-to-analog (D/A) converter 115 and played back in a loudspeaker unit 116. Alternatively, the synthesized digital sound signal 113 may also be supplied to and recorded in a storage device (not shown).

0019<figref idref="f0002">Figure 2</figref> is a schematic block diagram illustrating the structure of a CELP-based encoder and decoder, part of the sound communication system of <figref idref="f0001">Figure 1</figref>. As illustrated in <figref idref="f0002">Figure 2</figref>, a sound codec comprises two basic parts: the sound encoder 106 and the sound decoder 110 both introduced in the foregoing description of <figref idref="f0001">Figure 1</figref>. The encoder 106 is supplied with the original digital sound signal 105, determines the encoding parameters 107, described herein below, representing the original analog sound signal 103. These parameters 107 are encoded into the digital bit stream 111 that is transmitted using a communication channel, for example the communication channel 101 of <figref idref="f0001">Figure 1</figref>, to the decoder 110. The sound decoder 110 reconstructs the synthesized digital sound signal 113 to be as similar as possible to the original digital sound signal 105.

0020Presently, the most widespread speech coding techniques are based on Linear Prediction (LP), in particular CELP. In LP-based coding, the synthesized digital sound signal 113 is produced by filtering an excitation 214 through a LP synthesis filter 216 having a transfer function 1/<i>A</i>(<i>z</i>). In CELP, the excitation 214 is typically composed of two parts: a first-stage, adaptive-codebook contribution 222 selected from an adaptive codebook 218 and amplified by an adaptive-codebook gain <i>g_p</i> 226 and a second-stage, fixed-codebook contribution 224 selected from a fixed codebook 220 and amplified by a fixed-codebook gain <i>g_c</i> 228. Generally speaking, the adaptive codebook contribution 222 models the periodic part of the excitation and the fixed codebook contribution 214 is added to model the evolution of the sound signal.

0021The sound signal is processed by frames of typically 20 ms and the LP filter parameters are transmitted once per frame. In CELP, the frame is further divided in several subframes to encode the excitation. The subframe length is typically 5 ms.

0022CELP uses a principle called Analysis-by-Synthesis where possible decoder outputs are tried (synthesized) already during the coding process at the encoder 106 and then compared to the original digital sound signal 105. The encoder 106 thus includes elements similar to those of the decoder 110. These elements includes an adaptive codebook contribution 250 selected from an adaptive codebook 242 that supplies a past excitation signal <i>v(n)</i> convolved with the impulse response of a weighted synthesis filter <i>H(z)</i> (see 238) (cascade of the LP synthesis filter <i>1</i>/<i>A(z)</i> and the perceptual weighting filter <i>W(z)),</i> the result <i>y₁(n)</i> of which is amplified by an adaptive-codebook gain <i>g_p</i> 240. Also included is a fixed codebook contribution 252 selected from a fixed codebook 244 that supplies an innovative codevector <i>c_k(n)</i> convolved with the impulse response of the weighted synthesis filter <i>H(z)</i> (see 246), the result <i>y₂(n)</i> of which is amplified by a fixed codebook gain <i>g_c</i> 248.

0023The encoder 106 also comprises a perceptual weighting filter <i>W(z)</i> 233 and a provider 234 of a zero-input response of the cascade (<i>H(z)</i>) of the LP synthesis filter <i>1</i>/<i>A(z)</i> and the perceptual weighting filter <i>W(z).</i> Subtractors 236, 254 and 256 respectively subtract the zero-input response, the adaptive codebook contribution 250 and the fixed codebook contribution 252 from the original digital sound signal 105 filtered by the perceptual weighting filter 233 to provide a mean-squared error 232 between the original digital sound signal 105 and the synthesized digital sound signal 113.

0024The codebook search minimizes the mean-squared error 232 between the original digital sound signal 105 and the synthesized digital sound signal 113 in a perceptually weighted domain, where discrete time index <i>n</i> = 0, 1, ..., <i>N</i>-1, and N is the length of the subframe. The perceptual weighting filter <i>W(z)</i> exploits the frequency masking effect and typically is derived from a LP filter <i>A(z).</i>

0025An example of the perceptual weighting filter <i>W(z)</i> for WB (wideband, bandwidth of 50 - 7000 Hz) signals can be found in Reference [1].

0026Since the memory of the LP synthesis filter 1/<i>A(z)</i> and the weighting filter <i>W</i>(<i>z</i>) is independent from the searched codevectors, this memory can be subtracted from the original digital sound signal 105 prior to the fixed codebook search. Filtering of the candidate codevectors can then be done by means of a convolution with the impulse response of the cascade of the filters 1/A(z) and <i>W</i>(<i>z</i>), represented by <i>H</i>(<i>z)</i> in <figref idref="f0002">Figure 2</figref>.

0027The digital bit stream 111 transmitted from the encoder 106 to the decoder 110 contains typically the following parameters 107: quantized parameters of the LP filter <i>A</i>(<i>z</i>)<i>,</i> indices of the adaptive codebook 242 and of the fixed codebook 244, and the gains <i>g_p</i> 240 and <i>g_c</i> 248 of the adaptive codebook 242 and of the fixed codebook 244.

<i>Converting LP filter parameters when switching at frame boundaries with different sampling rates</i>

0028In LP-based coding the LP filter A(z) is determined once per frame, and then interpolated for each subframe. <figref idref="f0003">Figure 3</figref> illustrates an example of framing and interpolation of LP parameters. In this example, a present frame is divided into four subframes SF1, SF2, SF3 and SF4, and the LP analysis window is centered at the last subframe SF4. Thus the LP parameters resulting from LP analysis in the present frame, F1, are used as is in the last subframe, that is SF4 = F1. For the first three subframes SF1, SF2 and SF3, the LP parameters are obtained by interpolating the parameters in the present frame, F1, and a previous frame, F0. That is: <maths id="math0001"><math display="block"><mi>SF</mi><mn>1</mn><mo>=</mo><mn>0.75</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>0</mn><mo>+</mo><mn>0.25</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>1</mn><mo>;</mo></math><img file="EP4336500A2_D0001.tif" /></maths><maths id="math0002"><math display="block"><mi>SF</mi><mn>2</mn><mo>=</mo><mn>0.5</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>0</mn><mo>+</mo><mn>0.5</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>1</mn><mo>;</mo></math><img file="EP4336500A2_D0002.tif" /></maths><maths id="math0003"><math display="block"><mi>SF</mi><mn>3</mn><mo>=</mo><mn>0.25</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>0</mn><mo>+</mo><mn>0.75</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>1</mn></math><img file="EP4336500A2_D0003.tif" /></maths><maths id="math0004"><math display="block"><mi>SF</mi><mn>4</mn><mo>=</mo><mi mathvariant="normal">F</mi><mn>1</mn><mo>.</mo></math><img file="EP4336500A2_D0004.tif" /></maths>

0029Other interpolation examples may alternatively be used depending on the LP analysis window shape, length and position. In another embodiment, the coder switches between 12.8 kHz and 16 kHz internal sampling rates, where 4 subframes per frame are used at 12.8 kHz and 5 subframes per frame are used at 16 kHz, and where the LP parameters are also quantized in the middle of the present frame (Fm). In this other embodiment, LP parameter interpolation for a 12.8 kHz frame is given by: <maths id="math0005"><math display="block"><mi>SF</mi><mn>1</mn><mo>=</mo><mn>0.5</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>0</mn><mo>+</mo><mn>0.5</mn><mspace width="1ex" /><mi>Fm</mi><mo>;</mo></math><img file="EP4336500A2_D0005.tif" /></maths><maths id="math0006"><math display="block"><mi>SF</mi><mn>2</mn><mo>=</mo><mi>Fm</mi><mo>;</mo></math><img file="EP4336500A2_D0006.tif" /></maths><maths id="math0007"><math display="block"><mi>SF</mi><mn>3</mn><mo>=</mo><mn>0.5</mn><mspace width="1ex" /><mi>Fm</mi><mo>+</mo><mn>0.5</mn><mspace width="1ex" /><mi>F1</mi><mo>;</mo></math><img file="EP4336500A2_D0007.tif" /></maths><maths id="math0008"><math display="block"><mi>SF</mi><mn>4</mn><mo>=</mo><mi mathvariant="normal">F</mi><mn>1</mn><mo>.</mo></math><img file="EP4336500A2_D0008.tif" /></maths>

0030For a 16 kHz sampling, the interpolation is given by: <maths id="math0009"><math display="block"><mi>SF</mi><mn>1</mn><mo>=</mo><mn>0.55</mn><mspace width="1ex" /><mi>F0</mi><mo>+</mo><mn>0.45</mn><mspace width="1ex" /><mi>Fm</mi><mo>;</mo></math><img file="EP4336500A2_D0009.tif" /></maths><maths id="math0010"><math display="block"><mi>SF</mi><mn>2</mn><mo>=</mo><mn>0.15</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>0</mn><mo>+</mo><mn>0.85</mn><mspace width="1ex" /><mi>Fm</mi><mo>;</mo></math><img file="EP4336500A2_D0010.tif" /></maths><maths id="math0011"><math display="block"><mi>SF</mi><mn>3</mn><mo>=</mo><mn>0.75</mn><mspace width="1ex" /><mi>Fm</mi><mo>+</mo><mn>0.25</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>1</mn><mo>;</mo></math><img file="EP4336500A2_D0011.tif" /></maths><maths id="math0012"><math display="block"><mi>SF4</mi><mo>=</mo><mn>0.35</mn><mspace width="1ex" /><mi>Fm</mi><mo>+</mo><mn>0.65</mn><mspace width="1ex" /><mi mathvariant="normal">F</mi><mn>1</mn><mo>;</mo></math><img file="EP4336500A2_D0012.tif" /></maths><maths id="math0013"><math display="block"><mi>SF</mi><mn>5</mn><mo>=</mo><mi mathvariant="normal">F</mi><mn>1</mn><mo>.</mo></math><img file="EP4336500A2_D0013.tif" /></maths>

0031LP analysis results in computing the parameters of the LP synthesis filter using: <maths id="math0014" num="(1)"><math display="block"><mfrac><mn>1</mn><mrow><mi>A</mi><mfenced><mi>z</mi></mfenced></mrow></mfrac><mo>=</mo><mfrac><mn>1</mn><mrow><mn>1</mn><mo>+</mo><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><msup><mi>z</mi><mrow><mo>−</mo><mi>i</mi></mrow></msup></mrow></mstyle></mrow></mfrac><mo>=</mo><mfrac><mn>1</mn><mrow><mn>1</mn><mo>+</mo><msub><mi>a</mi><mn>1</mn></msub><msup><mi>z</mi><mrow><mo>−</mo><mn>1</mn></mrow></msup><mo>+</mo><msub><mi>a</mi><mn>2</mn></msub><msup><mi>z</mi><mrow><mo>−</mo><mn>2</mn></mrow></msup><mo>+</mo><mo>…</mo><mo>+</mo><msub><mi>a</mi><mi>M</mi></msub><msup><mi>z</mi><mrow><mo>−</mo><mi>M</mi></mrow></msup></mrow></mfrac></math><img file="EP4336500A2_D0014.tif" /></maths> where <i>a_i</i> , <i>i</i> = 1,...,<i>M</i>, are LP filter parameters and M is the filter order.

0032The LP filter parameters are transformed to another domain for quantization and interpolation purposes. Other LP parameter representations commonly used are reflection coefficients, log-area ratios, immitance spectrum pairs (used in AMR-WB; Reference [1]), and line spectrum pairs, which are also called line spectrum frequencies (LSF). In this illustrative embodiment, the line spectrum frequency representation is used. An example of a method that can be used to convert the LP parameters to LSF parameters and vice versa can be found in Reference [2]. The interpolation example in the previous paragraph is applied to the LSF parameters, which can be in the frequency domain in the range between 0 and Fs/2 (where Fs is the sampling frequency), or in the scaled frequency domain between 0 and π, or in the cosine domain (cosine of scaled frequency).

0033As described above, different internal sampling rates may be used at different bit rates to improve quality in multi-rate LP-based coding. In this illustrative embodiment, a multi-rate CELP wideband coder is used where an internal sampling rate of 12.8 kHz is used at lower bit rates and an internal sampling rate of 16 kHz at higher bit rates. At a 12.8 kHz sampling rate, the LSFs cover the bandwidth from 0 to 6.4 kHz, while at a 16 kHz sampling rate they cover the range from 0 to 8 kHz. When switching the bit rate between two frames where the internal sampling rate is different, some issues are addressed to insure seamless switching. These issues include the interpolation of LP filter parameters and the memories of the synthesis filter and the adaptive codebook, which are at different sampling rates.

0034The present disclosure introduces a method for efficient interpolation of LP parameters between two frames at different internal sampling rates. By way of example, the switching between 12.8 kHz and 16 kHz sampling rates is considered. The disclosed techniques are however not limited to these particular sampling rates and may apply to other internal sampling rates.

0035Let's assume that the encoder is switching from a frame F1 with internal sampling rate S1 to a frame F2 with internal sampling rate S2. The LP parameters in the first frame are denoted LSF1 _S1 and the LP parameters at the second frame are denoted LSF2_S2. In order to update the LP parameters in each subframe of frame F2, the LP parameters LSF1 and LSF2 are interpolated. In order to perform the interpolation, the filters have to be set at the same sampling rate. This requires performing LP analysis of frame F1 at sampling rate S2. To avoid transmitting the LP filter twice at the two sampling rates in frame F1, the LP analysis at sampling rate S2 can be performed on the past synthesis signal which is available at both encoder and decoder. This approach involves re-sampling the past synthesis signal from rate S1 to rate S2, and performing complete LP analysis, this operation being repeated at the decoder, which is usually computationally demanding.

0036Alternative method and devices are disclosed herein for converting LP synthesis filter parameters LSF1 from sampling rate S1 to sampling rate S2 without the need to re-sample the past synthesis and perform complete LP analysis. The method, used at encoding and/or at decoding, comprises computing the power spectrum of the LP synthesis filter at rate S1; modifying the power spectrum to convert it from rate S1 to rate S2; converting the modified power spectrum back to the time domain to obtain the filter autocorrelation at rate S2; and finally use the autocorrelation to compute LP filter parameters at rate S2.

0037In at least some embodiments, modifying the power spectrum to convert it from rate S1 to rate S2 comprises the following operations: If S1 is larger than S2, modifying the power spectrum comprises truncating the K-sample power spectrum down to K(S2/S1) samples, that is, removing K(S1-S2)/S1 samples.

0038On the other hand, if S1 is smaller than S2, then modifying the power spectrum comprises extending the K-sample power spectrum up to K(S2/S1) samples, that is, adding K(S2-S1)/S1 samples.

0039Computing the LP filter at rate S2 from the autocorrelations can be done using the Levinson-Durbin algorithm (see Reference [1]). Once the LP filter is converted to rate S2, the LP filter parameters are transformed to the interpolation domain, which is an LSF domain in this illustrative embodiment.

0040The procedure described above is summarized in <figref idref="f0004">Figure 4</figref>, which is a block diagram illustrating an embodiment for converting the LP filter parameters between two different sampling rates.

0041Sequence 300 of operations shows that a simple method for the computation of the power spectrum of the LP synthesis filter 1/A(z) is to evaluate the frequency response of the filter at K frequencies from 0 to 2<i>π</i> .

0042The frequency response of the synthesis filter is given by <maths id="math0015" num="(2)"><math display="block"><mfrac><mn>1</mn><mrow><mi>A</mi><mfenced><mi>ω</mi></mfenced></mrow></mfrac><mo>=</mo><mfrac><mn>1</mn><mrow><mn>1</mn><mo>+</mo><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><msup><mi>e</mi><mrow><mo>−</mo><mi mathvariant="italic">jωi</mi></mrow></msup></mrow></mstyle></mrow></mfrac><mo>=</mo><mfrac><mn>1</mn><mrow><mn>1</mn><mo>+</mo><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><mspace width="1ex" /><mi mathvariant="italic">cos</mi><mfenced><mi mathvariant="italic">ωi</mi></mfenced><mo>+</mo><mi>j</mi><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><mspace width="1ex" /><mi>sin</mi><mfenced><mi mathvariant="italic">ωi</mi></mfenced></mrow></mstyle></mrow></mstyle></mrow></mfrac></math><img file="EP4336500A2_D0015.tif" /></maths> and the power spectrum of the synthesis filter is calculated as an energy of the frequency response of the synthesis filter, given by <maths id="math0016" num="(3)"><math display="block"><mi>P</mi><mfenced><mi>ω</mi></mfenced><mo>=</mo><mfrac><mn>1</mn><msup><mfenced open="|" close="|"><mi>A</mi><mfenced><mi>ω</mi></mfenced></mfenced><mn>2</mn></msup></mfrac><mo>=</mo><mfrac><mn>1</mn><mrow><msup><mfenced><mn>1</mn><mo>+</mo><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><mspace width="1ex" /><mi>cos</mi><mfenced><mi mathvariant="italic">ωi</mi></mfenced></mrow></mstyle></mfenced><mn>2</mn></msup><mo>+</mo><msup><mfenced><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><mspace width="1ex" /><mi>sin</mi><mfenced><mi mathvariant="italic">ωi</mi></mfenced></mrow></mstyle></mfenced><mn>2</mn></msup></mrow></mfrac></math><img file="EP4336500A2_D0016.tif" /></maths>

0043Initially, the LP filter is at a rate equal to S1 (operation 310). A <i>K-</i>sample (i.e. discrete) power spectrum of the LP synthesis filter is computed (operation 320) by sampling the frequency range from 0 to 2<i>π</i> . That is <maths id="math0017" num="(4)"><math display="block"><mtable><mtr><mtd><mrow><mi>P</mi><mfenced><mi>k</mi></mfenced><mo>=</mo><mfrac><mn>1</mn><mrow><msup><mfenced><mn>1</mn><mo>+</mo><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><mspace width="1ex" /><mi>cos</mi><mfenced><mfrac><mrow><mn>2</mn><mi mathvariant="italic">πik</mi></mrow><mi>K</mi></mfrac></mfenced></mrow></mstyle></mfenced><mn>2</mn></msup><mo>+</mo><msup><mfenced><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>i</mi><mo>=</mo><mn>1</mn></mrow><mi>M</mi></munderover><mrow><msub><mi>a</mi><mi>i</mi></msub><mspace width="1ex" /><mi>sin</mi><mfenced><mfrac><mrow><mn>2</mn><mi mathvariant="italic">πik</mi></mrow><mi>K</mi></mfrac></mfenced></mrow></mstyle></mfenced><mn>2</mn></msup></mrow></mfrac><mo>,</mo></mrow></mtd><mtd><mrow><mi>k</mi><mo>=</mo><mn>0</mn><mo>,</mo><mo>…</mo><mo>,</mo><mi>K</mi><mo>−</mo><mn>1</mn></mrow></mtd></mtr></mtable></math><img file="EP4336500A2_D0017.tif" /></maths>

0044Note that it is possible to reduce operational complexity by computing <i>P(k)</i> only for <i>k</i> = 0,...,<i>K</i> / 2 since the power spectrum from <i>π</i> to 2<i>π</i> is a mirror of that from 0 to <i>π</i> .

0045A test (operation 330) determines which of the following cases apply. In a first case, the sampling rate S1 is larger than the sampling rate S2, and the power spectrum for frame F1 is truncated (operation 340) such that the new number of samples is <i>K</i>(<i>S2</i> / <i>S</i>1) .

0046In more details, when S1 is larger than S2, the length of the truncated power spectrum is <i>K</i>₂ = <i>K</i>(<i>S2</i> / <i>S</i>1) samples. Since the power spectrum is truncated, it is computed from <i>k =</i>0,...,<i>K</i>₂ 12. Since the power spectrum is symmetric around <i>K</i>₂ /2, then it is assumed that <maths id="math0018"><math display="block"><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>+</mo><mi>k</mi></mfenced><mo>=</mo><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mi>k</mi></mfenced><mo>,</mo><mspace width="1ex" /><mi>from</mi><mspace width="1ex" /><mi>k</mi><mo>=</mo><mn>1</mn><mo>,</mo><mo>…</mo><mo>,</mo><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mn>1</mn></math><img file="EP4336500A2_D0018.tif" /></maths>

0047The Fourier Transform of the autocorrelations of a signal gives the power spectrum of that signal. Thus, applying inverse Fourier Transform to the truncated power spectrum results in the autocorrelations of the impulse response of the synthesis filter at sampling rate S2.

0048The Inverse Discrete Fourier Transform (IDFT) of the truncated power spectrum is given by <maths id="math0019" num="(5)"><math display="block"><mi>R</mi><mfenced><mi>i</mi></mfenced><mo>=</mo><mfrac><mn>1</mn><msub><mi>K</mi><mn>2</mn></msub></mfrac><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>k</mi><mo>=</mo><mn>0</mn></mrow><mrow><msub><mi>K</mi><mn>2</mn></msub><mo>−</mo><mn>1</mn></mrow></munderover><mrow><mi>P</mi><mfenced><mi>k</mi></mfenced><msup><mi>e</mi><mrow><mi>j</mi><mn>2</mn><mi mathvariant="italic">πik</mi><mo>/</mo><msub><mi>K</mi><mn>2</mn></msub></mrow></msup></mrow></mstyle></math><img file="EP4336500A2_D0019.tif" /></maths>

0049Since the filter order is <i>M,</i> then the IDFT may be computed only for <i>i</i> = 0,...,<i>M</i>. Further, since the power spectrum is real and symmetric, then the IDFT of the power spectrum is also real and symmetric. Given the symmetry of the power spectrum, and that only M+1 correlations are needed, the inverse transform of the power spectrum can be given as <maths id="math0020" num="(6)"><math display="block"><mi>R</mi><mfenced><mi>i</mi></mfenced><mo>=</mo><mfrac><mn>1</mn><msub><mi>K</mi><mn>2</mn></msub></mfrac><mfenced><mi>P</mi><mfenced><mn>0</mn></mfenced><mo>+</mo><msup><mfenced><mo>−</mo><mn>1</mn></mfenced><mi>i</mi></msup><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn></mfenced><mo>+</mo><mn>2</mn><msup><mfenced><mo>−</mo><mn>1</mn></mfenced><mi>i</mi></msup><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>k</mi><mo>=</mo><mn>1</mn></mrow><mrow><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mn>1</mn></mrow></munderover><mrow><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mi>k</mi></mfenced><mi>cos</mi><mfenced><mn>2</mn><mi mathvariant="italic">πik</mi><mo>/</mo><msub><mi>K</mi><mn>2</mn></msub></mfenced></mrow></mstyle></mfenced></math><img file="EP4336500A2_D0020.tif" /></maths>

0050That is <maths id="math0021" num="(7)"><math display="block"><mi>R</mi><mfenced><mn>0</mn></mfenced><mo>=</mo><mfrac><mn>1</mn><msub><mi>K</mi><mn>2</mn></msub></mfrac><mfenced><mi>P</mi><mfenced><mn>0</mn></mfenced><mo>+</mo><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn></mfenced><mo>+</mo><mn>2</mn><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>k</mi><mo>=</mo><mn>1</mn></mrow><mrow><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mn>1</mn></mrow></munderover><mrow><mi>P</mi><mfenced><mi>k</mi></mfenced></mrow></mstyle></mfenced></math><img file="EP4336500A2_D0021.tif" /></maths><maths id="math0022"><math display="block"><mi>R</mi><mfenced><mi>i</mi></mfenced><mo>=</mo><mfrac><mn>1</mn><msub><mi>K</mi><mn>2</mn></msub></mfrac><mfenced><mi>P</mi><mfenced><mn>0</mn></mfenced><mo>−</mo><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn></mfenced><mo>−</mo><mn>2</mn><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>k</mi><mo>=</mo><mn>1</mn></mrow><mrow><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mn>1</mn></mrow></munderover><mrow><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mi>k</mi></mfenced><mi>cos</mi><mfenced><mn>2</mn><mi mathvariant="italic">πik</mi><mo>/</mo><msub><mi>K</mi><mn>2</mn></msub></mfenced></mrow></mstyle></mfenced><mspace width="1ex" /><mi>for</mi><mspace width="1ex" /><mi>i</mi><mo>=</mo><mn>1,3</mn><mo>,</mo><mo>…</mo><mo>,</mo><mi>M</mi><mo>−</mo><mn>1</mn></math><img file="EP4336500A2_D0022.tif" /></maths><maths id="math0023"><math display="block"><mi>R</mi><mfenced><mi>i</mi></mfenced><mo>=</mo><mfrac><mn>1</mn><msub><mi>K</mi><mn>2</mn></msub></mfrac><mfenced><mi>P</mi><mfenced><mn>0</mn></mfenced><mo>+</mo><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn></mfenced><mo>+</mo><mn>2</mn><mstyle displaystyle="true"><munderover><mo>∑</mo><mrow><mi>k</mi><mo>=</mo><mn>1</mn></mrow><mrow><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mn>1</mn></mrow></munderover><mrow><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mi>k</mi></mfenced><mi>cos</mi><mfenced><mn>2</mn><mi mathvariant="italic">πik</mi><mo>/</mo><msub><mi>K</mi><mn>2</mn></msub></mfenced></mrow></mstyle></mfenced><mspace width="1ex" /><mi>for</mi><mspace width="1ex" /><mi>i</mi><mo>=</mo><mn>2,4</mn><mo>,</mo><mo>…</mo><mo>,</mo><mi>M</mi></math><img file="EP4336500A2_D0023.tif" /></maths>

0051After the autocorrelations are computed at sampling rate S2, Levinson-Durbin algorithm (see Reference [1]) can be used to compute the parameters of the LP filter at sampling rate S2. Then, the LP filter parameters are transformed to the LSF domain for interpolation with the LSFs of frame F2 in order to obtain LP parameters at each subframe.

0052In the illustrative example where the coder encodes a wideband signal and is switching from a frame with an internal sampling rate S1=16 kHz to a frame with internal sampling rate S2=12.8 kHz, assuming that <i>K</i> = 100, the length of the truncated power spectrum is <i>K</i>₂ = 100(12800/16000) = 80 samples. The power spectrum is computed for 41 samples using Equation (4), and then the autocorrelations are computed using Equation (7) with <i>K</i>₂ = 80 .

0053In a second case, when the test (operation 330) determines that S1 is smaller than S2, the length of the extended power spectrum is <i>K</i>₂ = <i>K</i>(<i>S2</i> / <i>S</i>1) samples (operation 350). After computing the power spectrum from <i>k</i> = 0,..., <i>K</i> / 2 , the power spectrum is extended to <i>K</i>₂ / 2. Since there is no original spectral content between <i>K</i> / 2 and <i>K</i>₂ / 2 , extending the power spectrum can be done by inserting a number of samples up to <i>K</i>₂ / 2 using very low sample values. A simple approach is to repeat the sample at <i>K</i> / 2 up to <i>K</i>₂ 12. Since the power spectrum is symmetric around <i>K</i>₂ / 2 then it is assumed that <maths id="math0024"><math display="block"><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>+</mo><mi>k</mi></mfenced><mo>=</mo><mi>P</mi><mfenced><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mi>k</mi></mfenced><mo>,</mo><mspace width="1ex" /><mi>from</mi><mspace width="1ex" /><mi>k</mi><mo>=</mo><mn>1</mn><mo>,</mo><mo>…</mo><mo>,</mo><msub><mi>K</mi><mn>2</mn></msub><mo>/</mo><mn>2</mn><mo>−</mo><mn>1</mn></math><img file="EP4336500A2_D0024.tif" /></maths>

0054In either cases, the inverse DFT is then computed as in Equation (6) to obtain the autocorrelations at sampling rate S2 (operation 360) and the Levinson-Durbin algorithm (see Reference [1]) is used to compute the LP filter parameters at sampling rate S2 (operation 370). Then filter parameters are transformed to the LSF domain for interpolation with the LSFs of frame F2 in order to obtain LP parameters at each subframe.

0055Again, let's take the illustrative example where the coder is switching from a frame with an internal sampling rate S1=12.8 kHz to a frame with internal sampling rate S2=16 kHz, and let's assume that <i>K</i> = 80. The length of the extended power spectrum is <i>K</i>₂ = 80(16000/12800) = 100 samples. The power spectrum is computed for 51 samples using Equation (4), and then the autocorrelations are computed using Equation (7) with <i>K</i>₂ =100.

0056Note that other methods can be used to compute the power spectrum of the LP synthesis filter or the inverse DFT of the power spectrum without departing from the spirit of the present disclosure.

0057Note that in this illustrative embodiment converting the LP filter parameters between different internal sampling rates is applied to the quantized LP parameters, in order to determine the interpolated synthesis filter parameters in each subframe, and this is repeated at the decoder. It is noted that the weighting filter uses unquantized LP filter parameters, but it was found sufficient to interpolate between the unquantized filter parameters in new frame F2 and sampling-converted quantized LP parameters from past frame F1 in order to determine the parameters of the weighting filter in each subframe. This avoids the need to apply LP filter sampling conversion on the unquantized LP filter parameters as well.

<i>Other considerations when switching at frame boundaries with different sampling rates</i>

0058Another issue to be considered when switching between frames with different internal sampling rates is the content of the adaptive codebook, which usually contains the past excitation signal. If the new frame has an internal sampling rate S2 and the previous frame has an internal sampling rate S1, then the content of the adaptive codebook is re-sampled from rate S1 to rate S2, and this is performed at both the encoder and the decoder.

0059In order to reduce the complexity, in this disclosure, the new frame F2 is forced to use a transient encoding mode which is independent of the past excitation history and thus does not use the history of the adaptive codebook. An example of transient mode encoding can be found in <patcit id="pcit0001" dnum="WO2008049221A1"><text>PCT patent application WO 2008/049221 A1</text></patcit> "Method and device for coding transition frames in speech signals", the disclosure of which is incorporated by reference herein.

0060Another consideration when switching at frame boundaries with different sampling rates is the memory of the predictive quantizers. As an example, LP-parameter quantizers usually use predictive quantization, which may not work properly when the parameters are at different sampling rates. In order to reduce switching artefacts, the LP-parameter quantizer may be forced into a non-predictive coding mode when switching between different sampling rates.

0061A further consideration is the memory of the synthesis filter, which may be resampled when switching between frames with different sampling rates.

0062Finally, the additional complexity that arises from converting LP filter parameters when switching between frames with different internal sampling rates may be compensated by modifying parts of the encoding or decoding processing. For example, in order not to increase the encoder complexity, the fixed codebook search may be modified by lowering the number of iterations in the first subframe of the frame (see Reference [1] for an example of fixed codebook search).

0063Additionally, in order not to increase the decoder complexity, certain post-processing can be skipped. For example, in this illustrative embodiment, a post-processing technique as described in <patcit id="pcit0002" dnum="US7529660B"><text>US patent 7,529,660</text></patcit> "Method and device for frequency-selective pitch enhancement of synthesized speech", the disclosure of which is incorporated by reference herein, may be used. This post-filtering is skipped in the first frame after switching to a different internal sampling rate (skipping this post-filtering also overcomes the need of past synthesis utilized in the post-filter).

0064Further, other parameters that depend on the sampling rate may be scaled accordingly. For example, the past pitch delay used for decoder classifier and frame erasure concealment may be scaled by the factor S2/S1.

0065<figref idref="f0005">Figure 5</figref> is a simplified block diagram of an example configuration of hardware components forming the encoder and/or decoder of <figref idref="f0001">Figures 1</figref> and <figref idref="f0002">2</figref>. A device 400 may be implemented as a part of a mobile terminal, as a part of a portable media player, a base station, Internet equipment or in any similar device, and may incorporate the encoder 106, the decoder 110, or both the encoder 106 and the decoder 110. The device 400 includes a processor 406 and a memory 408. The processor 406 may comprise one or more distinct processors for executing code instructions to perform the operations of <figref idref="f0004">Figure 4</figref>. The processor 406 may embody various elements of the encoder 106 and of the decoder 110 of <figref idref="f0001">Figures 1</figref> and <figref idref="f0002">2</figref>. The processor 406 may further execute tasks of a mobile terminal, of a portable media player, base station, Internet equipement and the like. The memory 408 is operatively connected to the processor 406. The memory 408, which may be a non-transitory memory, stores the code instructions executable by the processor 406.

0066An audio input 402 is present in the device 400 when used as an encoder 106. The audio input 402 may include for example a microphone or an interface connectable to a microphone. The audio input 402 may include the microphone 102 and the A/D converter 104 and produce the original analog sound signal 103 and/or the original digital sound signal 105. Alternatively, the audio input 402 may receive the original digital sound signal 105. Likewise, an encoded output 404 is present when the device 400 is used as an encoder 106 and is configured to forward the encoding parameters 107 or the digital bit stream 111 containing the parameters 107, including the LP filter parameters, to a remote decoder via a communication link, for example via the communication channel 101, or toward a further memory (not shown) for storage. Non-limiting implementation examples of the encoded output 404 comprise a radio interface of a mobile terminal, a physical interface such as for example a universal serial bus (USB) port of a portable media player, and the like.

0067An encoded input 403 and an audio output 405 are both present in the device 400 when used as a decoder 110. The encoded input 403 may be constructed to receive the encoding parameters 107 or the digital bit stream 111 containing the parameters 107, including the LP filter parameters from an encoded output 404 of an encoder 106. When the device 400 includes both the encoder 106 and the decoder 110, the encoded output 404 and the encoded input 403 may form a common communication module. The audio output 405 may comprise the D/A converter 115 and the loudspeaker unit 116. Alternatively, the audio output 405 may comprise an interface connectable to an audio player, to a loudspeaker, to a recording device, and the like.

0068The audio input 402 or the encoded input 403 may also receive signals from a storage device (not shown). In the same manner, the encoded output 404 and the audio output 405 may supply the output signal to a storage device (not shown) for recording.

0069The audio input 402, the encoded input 403, the encoded output 404 and the audio output 405 are all operatively connected to the processor 406.

0070Those of ordinary skill in the art will realize that the description of the methods, encoder and decoder for linear predictive encoding and decoding of sound signals are illustrative only and are not intended to be in any way limiting. Other embodiments will readily suggest themselves to such persons with ordinary skill in the art having the benefit of the present disclosure. Furthermore, the disclosed methods, encoder and decoder may be customized to offer valuable solutions to existing needs and problems of switching linear prediction based codecs between two bit rates with different sampling rates.

0071In the interest of clarity, not all of the routine features of the implementations of methods, encoder and decoder are shown and described. It will, of course, be appreciated that in the development of any such actual implementation of the methods, encoder and decoder, numerous implementation-specific decisions may need to be made in order to achieve the developer's specific goals, such as compliance with application-, system-, network- and business-related constraints, and that these specific goals will vary from one implementation to another and from one developer to another. Moreover, it will be appreciated that a development effort might be complex and time-consuming, but would nevertheless be a routine undertaking of engineering for those of ordinary skill in the field of sound coding having the benefit of the present disclosure.

0072In accordance with the present disclosure, the components, process operations, and/or data structures described herein may be implemented using various types of operating systems, computing platforms, network devices, computer programs, and/or general purpose machines. In addition, those of ordinary skill in the art will recognize that devices of a less general purpose nature, such as hardwired devices, field programmable gate arrays (FPGAs), application specific integrated circuits (ASICs), or the like, may also be used. Where a method comprising a series of operations is implemented by a computer or a machine and those operations may be stored as a series of instructions readable by the machine, they may be stored on a tangible medium.

0073Systems and modules described herein may comprise software, firmware, hardware, or any combination(s) of software, firmware, or hardware suitable for the purposes described herein.

0074Although the present disclosure has been described hereinabove by way of non-restrictive, illustrative embodiments thereof, these embodiments may be modified at will within the scope of the appended claims without departing from the spirit and nature of the present disclosure.

0075The following embodiments are part of this description relating to the invention.

0076Embodiment 1. A method implemented in a sound signal encoder for converting linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2, the method comprising: <ul id="ul0004" list-style="none"><li>computing, at the sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters;</li><li>modifying the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2;</li><li>inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the sampling rate S2; and</li><li>using the autocorrelations to compute the LP filter parameters at the sampling rate S2.</li></ul>

0077Embodiment 2. A method as recited in embodiment 1, wherein modifying the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2 comprises: <ul id="ul0005" list-style="none"><li>if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on a ratio between S1 and S2;</li><li>if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.</li></ul>

0078Embodiment 3. A method as recited in any one of embodiments 1 and 2, wherein the conversion of the LP filter parameters occurs when an encoder switches from a frame with the sampling rate S1 to a frame with the sampling rate S2.

0079Embodiment 4. A method as recited in embodiment 3, comprising computing LP filter parameters in each subframe of a current frame by interpolating LP filter parameters of the current frame at the sampling rate S2 with LP filter parameters of a past frame converted from the sampling rate S1 to the sampling rate S2.

0080Embodiment 5. A method as recited in embodiment 4, comprising forcing the current frame to an encoding mode that does not use a history of an adaptive codebook.

0081Embodiment 6. A method as recited in any one of embodiments 4 or 5, comprising forcing a LP- parameter quantizer to use a non-predictive quantization method in the current frame.

0082Embodiment 7. The method as recited in any one of embodiments 1 to 6, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

0083Embodiment 8. A method as recited in any one of embodiments 1 to 7, comprising: <ul id="ul0006" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>extending the power spectrum of the LP synthesis filter to K(S2/S1) samples when the sampling rate S1 is less than the sampling rate S2; and</li><li>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.</li></ul>

0084Embodiment 9. A method as recited in any one of embodiments 1 to 8, comprising computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

0085Embodiment 10. A method as recited in any one of embodiments 1 to 9, comprising inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0086Embodiment 11. A method as recited in any one of embodiments 1 to 10, comprising searching a fixed codebook using a reduced number of iterations.

0087Embodiment 12. A method implemented in a sound signal decoder for converting received linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2, the method comprising: <ul id="ul0007" list-style="none"><li>computing, at the sampling rate S1, a power spectrum of a LP synthesis filter using the received LP filter parameters;</li><li>modifying the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2;</li><li>inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the sampling rate S2; and</li><li>using the autocorrelations to compute the LP filter parameters at the sampling rate S2.</li></ul>

0088Embodiment 13. A method as recited in embodiment 12, wherein modifying the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2 comprises: <ul id="ul0008" list-style="none"><li>if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on a ratio between S1 and S2;</li><li>if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.</li></ul>

0089Embodiment 14. A method as recited in any one of embodiments 12 and 13, wherein the conversion of the received LP filter parameters occurs when a decoder switches from a frame with the sampling rate S1 to a frame with the sampling rate S2.

0090Embodiment 15. A method as recited in embodiment 14, comprising computing LP filter parameters in each subframe of a new frame by interpolating LP filter parameters of a current frame at the sampling rate S2 with LP filter parameters of a past frame converted from the sampling rate S1 to the sampling rate S2.

0091Embodiment 16. The method as recited in any one of embodiments 12 to 15, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

0092Embodiment 17. A method as recited in any one of embodiments 12 to 16, comprising: <ul id="ul0009" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>extending the power spectrum of the LP synthesis filter to K(S2/S1) samples when the sampling rate S1 is less than the sampling rate S2; and</li><li>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.</li></ul>

0093Embodiment 18. A method as recited in any one of embodiments 12 to 17, comprising computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

0094Embodiment 19. A method as recited in any one of embodiments 12 to 18, comprising inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0095Embodiment 20. A method as recited in any one of embodiments 12 to 19, wherein a post filtering is skipped to reduce decoding complexity.

0096Embodiment 21. A device for use in a sound signal encoder for converting linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2, device comprising: a processor configured to: <ul id="ul0010" list-style="none"><li>compute, at the sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2,</li><li>inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the sampling rate S2, and</li><li>use the autocorrelations to compute the LP filter parameters at the sampling rate S2.</li></ul>

0097Embodiment 22. A device as recited in embodiment 21, wherein the processor is configured to: <ul id="ul0011" list-style="none"><li>extend the power spectrum of the LP synthesis filter based on a ratio between S1 and S2 if S1 is less than S2; and</li><li>truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.</li></ul>

0098Embodiment 23. A device as recited in any one of embodiments 21 and 22, 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 sampling rate S2 with LP filter parameters of a past frame converted from the sampling rate S1 to the sampling rate S2.

0099Embodiment 24. A device as recited in any one of embodiments 21 to 23, wherein the processor is configured to: <ul id="ul0012" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>extend the power spectrum of the LP synthesis filter to K(S2/S1) samples when the sampling rate S1 is less than the sampling rate S2; and</li><li>truncate 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.</li></ul>

0100Embodiment 25. A device as recited in any one of embodiments 21 to 24, 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.

0101Embodiment 26. A device as recited in any one of embodiments 21 to 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.

0102Embodiment 27. An encoder as recited in any one of embodiments 21 to 26, further comprising a non-transitory memory storing code instructions executable by the processor.

0103Embodiment 28. A computer-readable non-transitory memory storing code instructions for performing, when running on the processor of any one of embodiments 21 to 27, a method as recited in any one of embodiments 1 to 11.

0104Embodiment 29. A device for use in a sound signal decoder for converting received linear predictive (LP) filter parameters from a sound signal sampling rate S1 to a sound signal sampling rate S2, the device comprising: a processor configured to: <ul id="ul0013" list-style="none"><li>compute, at the sampling rate S1, a power spectrum of a LP synthesis filter using the received LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the sampling rate S1 to the sampling rate S2,</li><li>inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the sampling rate S2, and</li><li>use the autocorrelations to compute the LP filter parameters at the sampling rate S2.</li></ul>

0105Embodiment 30. A device as recited in embodiment 29, wherein the processor is further configured to: <ul id="ul0014" list-style="none"><li>extend the power spectrum of the LP synthesis filter based on a ratio between S1 and S2 if S1 is less than S2; and</li><li>truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.</li></ul>

0106Embodiment 31. A device as recited in any one of embodiments 29 and 30, 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 sampling rate S2 with LP filter parameters of a past frame converted from the sampling rate S1 to the sampling rate S2.

0107Embodiment 32. A device as recited in any one of embodiments 29 to 31, wherein the processor is configured to: <ul id="ul0015" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>extend the power spectrum of the LP synthesis filter to K(S2/S1) samples when the sampling rate S1 is less than the sampling rate S2; and</li><li>truncate 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.</li></ul>

0108Embodiment 33. A device as recited in any one of embodiments 29 to 32, 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.

0109Embodiments 34. A device as recited in any one of embodiments 29 to 33, wherein the processor is configured to inverse transform the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0110Embodiment 35. A decoder as recited in any one of embodiments 29 to 34, further comprising a non-transitory memory storing code instructions executable by the processor.

0111Embodiment 36. A computer-readable non-transitory memory storing code instructions for performing, when running on the processor of any one of embodiments 29 to 35, a method as recited in any one of embodiments 12 to 20.

0112Embodiment 37. A method for encoding a sound signal, comprising: <ul id="ul0016" list-style="none"><li>sampling the sound signal during successive sound signal processing frames;</li><li>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, 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: <ul id="ul0017" list-style="none"><li>computing, at the internal sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters;</li><li>modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2 based on a ratio between the internal sampling rates S1 and S2;</li><li>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</li><li>using the autocorrelations to compute the LP filter parameters at the internal sampling rate S2; and</li><li>encoding the sound signal encoding parameters into a bitstream.</li></ul></li></ul>

0113Embodiment 38. The method as recited in embodiment 37, 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 further comprises: <ul id="ul0018" list-style="none"><li>if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on the ratio between S1 and S2;</li><li>if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.</li></ul>

0114Embodiment 39. The method as recited in embodiments 37 or 38, 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.

0115Embodiment 40. The method as recited in any one of embodiments 37 to 39, comprising: forcing the current frame to an encoding mode using no history of an adaptive codebook.

0116Embodiment 41. The method as recited in any one of embodiments 37 to 40, comprising: forcing a LP-parameter quantizer to use a non-predictive quantization method in the current frame.

0117Embodiment 42. The method as recited in any one of embodiments 37 to 41, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

0118Embodiment 43. The method as recited in any one of embodiments 37 to 42, comprising: <ul id="ul0019" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>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</li><li>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.</li></ul>

0119Embodiment 44. The method as recited in any one of embodiments 37 to 43, comprising: computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

0120Embodiment 45. The method as recited in any one of embodiments 37 to 44, comprising: inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0121Embodiment 46. The method as recited in any one of embodiments 37 to 45, comprising: searching a fixed codebook using a reduced number of iterations.

0122Embodiment 47. A method for decoding a sound signal, comprising: <ul id="ul0020" list-style="none" compact="compact"><li>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;</li><li>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 the 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: <ul id="ul0021" list-style="none"><li>computing, at the internal sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters;</li><li>modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2 based on a ratio between the internal sampling rates S1 and S2;</li><li>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</li><li>using the autocorrelations to compute the LP filter parameters at the internal sampling rate S2; and</li><li>synthesizing the sound signal using LP synthesis filtering in response to the decoded LP filter parameters and the LP synthesis filter excitation signal.</li></ul></li></ul>

0123Embodiment 48. The method as recited in embodiment 47, 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: <ul id="ul0022" list-style="none"><li>if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on the ratio between S1 and S2;</li><li>if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.</li></ul>

0124Embodiment 49. The method as recited in embodiment 47 or 48, 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.

0125Embodiment 50. The method as recited in any one of embodiments 47 to 49, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

0126Embodiment 51. The method as recited in any one of embodiments 47 to 50, comprising: <ul id="ul0023" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>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</li><li>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.</li></ul>

0127Embodiment 52. The method as recited in any one of embodiments 47 to 51, comprising: computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

0128Embodiment 53. The method as recited in any one of embodiments 47 to 52, comprising inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0129Embodiment 54. The method as recited in any one of embodiments 47 to 53, comprising skipping postfiltering to reduce decoding complexity.

0130Embodiment 55. A device for encoding a sound signal, comprising: <ul id="ul0024" list-style="none"><li>at least one processor; and</li><li>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: <ul id="ul0025" list-style="none"><li>compute, at the internal sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2 based on a ratio between the internal sampling rates S1 and S2,</li><li>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</li><li>use the autocorrelations to compute the LP filter parameters at the internal sampling rate S2, and</li><li>encode the sound signal encoding parameters into a bitstream.</li></ul></li></ul>

0131Embodiment 56. The device as recited in embodiment 55, wherein the processor is configured to: <ul id="ul0026" list-style="none"><li>extend the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is less than S2; and</li><li>truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.</li></ul>

0132Embodiment 57. The device as recited in embodiment 55 or 56, 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.

0133Embodiment 58. The device as recited in any one of embodiments 55 to 57, wherein the processor is configured to: <ul id="ul0027" list-style="none"><li>compute the power spectrum of the LP synthesis filter at K samples;</li><li>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</li><li>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.</li></ul>

0134Embodiment 59. The device as recited in any one of embodiments 55 to 58, 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.

0135Embodiment 60. The device as recited in any one of embodiments 55 to 59, wherein the processor is configured to inverse transform the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0136Embodiment 61. A device for decoding a sound signal, comprising: <ul id="ul0028" list-style="none"><li>at least one processor; and</li><li>a memory coupled to the processor and comprising non-transitory instructions that when executed cause the processor to: <ul id="ul0029" list-style="none"><li>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;</li><li>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: <ul id="ul0030" list-style="none"><li>compute, at the internal sampling rate S1, a power spectrum of a LP synthesis filter using the received LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2 based on a ratio between the internal sampling rates S1 and S2,</li><li>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</li><li>use the autocorrelations to compute the LP filter parameters at the internal sampling rate S2, and</li><li>synthesize the sound signal using LP synthesis filtering in response to the decoded LP filter parameters and the LP synthesis filter excitation signal.</li></ul></li></ul></li></ul>

0137Embodiment 62. The device as recited in embodiment 61, wherein the processor is configured to: <ul id="ul0031" list-style="none"><li>extend the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is less than S2; and</li><li>truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.</li></ul>

0138Embodiment 63. The device as recited in embodiment 61 or 62, 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.

0139Embodiment 64. The device as recited in any one of embodiments 61 to 63, wherein the processor is configured to: <ul id="ul0032" list-style="none"><li>compute the power spectrum of the LP synthesis filter at K samples;</li><li>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</li><li>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.</li></ul>

0140Embodiment 65. The device as recited in any one of embodiments 61 to 64, 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.

0141Embodiment 66. The device as recited in any one of embodiments 61 to 65, wherein the processor is configured to inverse transform the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0142Embodiment 67. A method implemented in a CELP-based sound signal encoder or a CELP-based sound signal decoder for converting, when the encoder or the decoder switches from a first frame with an internal sampling rate S1 to a second frame, divided into subframes, with an internal sampling rate S2, linear predictive, LP, filter parameters of the first frame from the internal sampling rate S1 to the internal sampling rate S2, the method being characterized by : <ul id="ul0033" list-style="none"><li>computing, at the internal sampling rate S1, a power spectrum of an LP synthesis filter using the LP filter parameters;</li><li>modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2;</li><li>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</li><li>using the autocorrelations to compute the LP filter parameters at the internal sampling rate S2;</li><li>wherein the method further comprises: determining for at least one subframe of the second frame interpolated LP filter parameters by interpolation between LP filter parameters of the second frame determined at the internal sampling rate S2 and the LP filter parameters of the first frame converted from the internal sampling rate S1 to the internal sampling rate S2.</li></ul>

0143Embodiment 68. A method as recited in embodiment 67, wherein the step of determining interpolated LP filter parameters in the at least one subframe of the second frame further comprises: transforming the LP filter parameters of the first frame and second frame to line spectrum frequencies, LSF, representation or to line spectrum pairs, LSP, representation.

0144Embodiment 69. A method as recited in embodiment 67 or 68, 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: <ul id="ul0034" list-style="none"><li>if S1 is less than S2, extending the power spectrum of the LP synthesis filter based on a ratio between S1 and S2;</li><li>if S1 is larger than S2, truncating the power spectrum of the LP synthesis filter based on the ratio between S1 and S2.</li></ul>

0145Embodiment 70. A method as recited in embodiment 69, comprising, when implemented in a CELP-based sound signal encoder, forcing the current frame to an encoding mode that does not use a history of an adaptive codebook.

0146Embodiment 71. A method as recited in any one of embodiments 69 and 70, comprising, when implemented in a CELP-based sound signal encoder, forcing a LP-parameter quantizer to use a non-predictive quantization method in the current frame.

0147Embodiment 72. A method as recited in any one of embodiments 67 to 71, wherein the power spectrum of the LP synthesis filter is a discrete power spectrum.

0148Embodiment 73. A method as recited in any one of embodiments 67 to 72, comprising: <ul id="ul0035" list-style="none"><li>computing the power spectrum of the LP synthesis filter at K samples;</li><li>extending the power spectrum of the LP synthesis filter to K2 = K*S2/S1 samples when the internal sampling rate S1 is less than the internal sampling rate S2; and</li><li>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.</li></ul>

0149Embodiment 74. A method as recited in embodiment 73, wherein the step of extending the power spectrum comprises repeating the sample at K/2 up to K2/2.

0150Embodiment 75. A method as recited in any one of embodiments 67 to 74, comprising computing the power spectrum of the LP synthesis filter as an energy of a frequency response of the LP synthesis filter.

0151Embodiment 76. A method as recited in any one of embodiments 67 to 75, comprising inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0152Embodiment 77. A method as recited in any one of embodiments 67 to 76, comprising searching a fixed codebook using a reduced number of iterations.

0153Embodiment 78. A method as recited in any one of embodiments 67 to 77, wherein, when the method is implemented in a CELP-based sound signal decoder, a post filtering is skipped to reduce decoding complexity.

0154Embodiment 79. A device for use in a CELP-based sound signal encoder or a CELP-based sound signal decoder for converting, when the encoder or the decoder switches from a first frame with an internal sampling rate S1 to a second frame, divided into subframes, with an internal sampling rate S2, linear predictive, LP, filter parameters of the first frame from the internal sampling rate S1 to the internal sampling rate S2, the device being characterized in that it comprises: a processor configured to: <ul id="ul0036" list-style="none"><li>compute, at the internal sampling rate S1, a power spectrum of a LP synthesis filter using the LP filter parameters,</li><li>modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S1 to the internal sampling rate S2,</li><li>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</li><li>use the autocorrelations to compute the LP filter parameters at the internal sampling rate S2;</li><li>wherein the processor is further configured to: determine for at least one subframe of the second frame interpolated LP filter parameters by interpolation between LP filter parameters of the second frame determined at the internal sampling rate S2 and the LP filter parameters of the first frame converted from the internal sampling rate S1 to the internal sampling rate S2.</li></ul>

0155Embodiment 80. A device as recited in embodiment 79, wherein the processor is configured to: transform the LP filter parameters of the first frame and second frame to line spectrum frequencies, LSF, representation or to line spectrum pairs, LSP, representation.

0156Embodiment 81. A device as recited in embodiment 79 or 80, wherein the processor is configured to: <ul id="ul0037" list-style="none"><li>extend the power spectrum of the LP synthesis filter based on a ratio between S1 and S2 if S1 is less than S2; and</li><li>truncate the power spectrum of the LP synthesis filter based on the ratio between S1 and S2 if S1 is larger than S2.</li></ul>

0157Embodiment 82. A device as recited in any one of embodiments 79 to 81, wherein the processor is configured to: <ul id="ul0038" list-style="none"><li>compute the power spectrum of the LP synthesis filter at K samples;</li><li>extend the power spectrum of the LP synthesis filter to K2 = K*S2/S1 samples when the internal sampling rate S1 is less than the internal sampling rate S2; and</li><li>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.</li></ul>

0158Embodiment 83. A device as recited in embodiment 82, wherein the processor is configured to extend the power spectrum by repeating the sample at K/2 up to K2/2.

0159Embodiment 84. A device as recited in any one of embodiments 79 to 83, 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.

0160Embodiment 85. A device as recited in any one of embodiments 79 to 84, wherein the processor is configured to inverse transform the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

0161Embodiment 86. A device as recited in any one of embodiments 79 to 85, further comprising a non-transitory memory storing code instructions executable by the processor to perform the computing, modifying, inverse transforming and using operations.

0162Embodiment 87. A computer-readable non-transitory memory storing code instructions which, when running on a processor, cause the processor to perform a method as recited in any one of embodiments 67 to 78.

REFERENCES

0163The following references are incorporated by reference herein. <ol id="ol0001" compact="compact"><li>[1] <nplcit id="ncit0001" npl-type="s"><text>3GPP Technical Specification 26.190, "Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Transcoding functions," July 2005; http://www.3gpp.org</text></nplcit><u>.</u></li><li>[2] <nplcit id="ncit0002" npl-type="s"><text>ITU-T Recommendation G.729 "Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP)", 01/2007</text></nplcit>.</li></ol>