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

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

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

Abstract: A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector.

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

Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

Abstract: An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.

Abstract: A device and method for estimating a tonal stability of a sound signal include: calculating a current residual spectrum of the sound signal; detecting peaks in the current residual spectrum; calculating a correlation map between the current residual spectrum and a previous residual spectrum for each detected peak; and calculating a long-term correlation map based on the calculated correlation map, the long-term correlation map being indicative of a tonal stability in the sound signal.

Abstract: An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

Abstract: A device and a method for quantizing, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantizing device and method comprises: an absolute quantizer for first quantizing one of the LPC filters using absolute quantization; and at least one quantizer of the other LPC filters using a quantization mode selected from the group consisting of absolute quantization and differential quantization relative to at least one previously quantized filter amongst the LPC filters. For inverse quantizing, at least the first quantized LPC filter is received and an inverse quantizer inverse quantizes the first quantized LPC filter using absolute inverse quantization. If any quantized LPC filter other than the first quantized LPC filter is received, an inverse quantizer inverse quantizes this quantized LPC filter using one of absolute inverse quantization and differential inverse quantization relative to at least one previously received quantized LPC filter.

Abstract: A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions.

Abstract: An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters.

Abstract: An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.

Abstract: A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.

Abstract: There is provided a transition mode device and method for use in a predictive-type sound signal codec for producing a transition mode excitation replacing an adaptive codebook excitation in a transition frame and/or a frame following the transition in the sound signal, comprising an input for receiving a codebook index and a transition mode codebook for generating a set of codevectors independent from past excitation. The transition mode codebook is responsive to the index for generating, in the transition frame and/or frame following the transition, one of the codevectors of the set corresponding to the transition mode excitation. There is also provided an encoding device and method and a decoding device and method using the above described transition mode device and method.

Abstract: A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a minima of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients.

Abstract: A multi-reference quantization device and method for quantizing an input LPC filter, comprises a plurality of differential quantizers using respective, different references, and a selector of a reference amongst the different references of the differential quantizers using a reference selection criterion. The input LPC filter is differentially quantized by the differential quantizer using the selected reference. A device and method for inverse quantizing a multi-reference differentially quantized LPC filter extracted from a bitstream, comprises an extractor from the bitstream of information about a reference amongst a plurality of possible references used for quantizing the multi-reference differentially quantized LPC filter, and a differential inverse quantizer using the reference corresponding to the extracted reference information to inverse quantize the multi-reference differentially quantized LPC filter.

Abstract: An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

Abstract: An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters.