Patent classifications
G10L2019/0002
CONCEPT FOR SWITCHING OF SAMPLING RATES AT AUDIO PROCESSING DEVICES
Audio decoder device for decoding a bitstream, the audio decoder device including: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder includes a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder includes a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device including one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, for one or more of the memories by resampling a preceding memory state for synthesizing one or more audio parameters for a preceding decoded audio frame, which has a preceding sampling rate being different from the sampling rate of the decoded audio frame, for one or more of the memories and to store the memory state for synthesizing of the one or more audio parameters for the decoded audio frame for one or more of the memories into the respective memory.
METHOD AND DEVICE FOR QUANTIZATION OF LINEAR PREDICTION COEFFICIENT AND METHOD AND DEVICE FOR INVERSE QUANTIZATION
A quantization apparatus comprises: a first quantization module for performing quantization without an inter-frame prediction; and a second quantization module for performing quantization with an inter-frame prediction, and the first quantization module comprises: a first quantization part for quantizing an input signal; and a third quantization part for quantizing a first quantization error signal, and the second quantization module comprises: a second quantization part for quantizing a prediction error; and a fourth quantization part for quantizing a second quantization error signal, and the first quantization part and the second quantization part comprise a trellis structured vector quantizer.
Speech model parameter estimation and quantization
Quantizing speech model parameters includes, for each of multiple vectors of quantized excitation strength parameters, determining first and second errors between first and second elements of a vector of excitation strength parameters and, respectively, first and second elements of the vector of quantized excitation strength parameters, and determining a first energy and a second energy associated with, respectively, the first and second errors. First and second weights for, respectively, the first error and the second error, are determined and are used to produce first and second weighted errors, which are combined to produce a total error. The total errors of each of the multiple vectors of quantized excitation strength parameters are compared and the vector of quantized excitation strength parameters that produces the smallest total error is selected to represent the vector of excitation strength parameters.
Concept for switching of sampling rates at audio processing devices
Audio decoder device for decoding a bitstream, the audio decoder device including: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder includes a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder includes a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device including one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, for one or more of the memories by resampling a preceding memory state for synthesizing one or more audio parameters for a preceding decoded audio frame, which has a preceding sampling rate being different from the sampling rate of the decoded audio frame, for one or more of the memories and to store the memory state for synthesizing of the one or more audio parameters for the decoded audio frame for one or more of the memories into the respective memory.
METHOD OF TRAINING SPEECH RECOGNITION MODEL, ELECTRONIC DEVICE AND STORAGE MEDIUM
A method of training a speech recognition model is provided. The method includes that: speech data of each of a plurality of training samples is inputted into a teacher model and a to-be-trained speech recognition model separately. Additionally, an embedding outputted by the teacher model and encoded data outputted by the to-be-trained speech recognition model are obtained. Furthermore, quantized codebook data is obtained by performing a multi-codebook quantization on the embedding. A loss is calculated based on the encoded data, the quantized codebook data, and text data in the training sample. Moreover, a trained speech recognition model is obtained by stopping training the to-be-trained speech recognition model when the loss is less than or equal to a preset loss threshold and/or trained times is greater than preset trained times.
APPARATUS AND METHOD FOR IMPROVED CONCEALMENT OF THE ADAPTIVE CODEBOOK IN A CELP-LIKE CONCEALMENT EMPLOYING IMPROVED PITCH LAG ESTIMATION
An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.
Apparatus and method for generating an adaptive spectral shape of comfort noise
An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided, having: a receiving interface for receiving one or more frames, a coefficient generator, and a signal reconstructor. The coefficient generator is configured to determine one or more first audio signal coefficients, and one or more noise coefficients. Moreover, the coefficient generator is configured to generate one or more second audio signal coefficients, depending on the one or more first audio signal coefficients and depending on the one or more noise coefficients. The audio signal reconstructor is configured to reconstruct a first portion of the reconstructed audio signal depending on the one or more first audio signal coefficients and the audio signal reconstructor is configured to reconstruct a second portion of the reconstructed audio signal depending on the one or more second audio signal coefficients, if the current frame is not received by the receiving interface or if the current frame being received by the receiving interface is corrupted.
Concept for switching of sampling rates at audio processing devices
Audio decoder device for decoding a bitstream, the audio decoder device including: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder includes a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder includes a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device including one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, for one or more of the memories by resampling a preceding memory state for synthesizing one or more audio parameters for a preceding decoded audio frame, which has a preceding sampling rate being different from the sampling rate of the decoded audio frame, for one or more of the memories and to store the memory state for synthesizing of the one or more audio parameters for the decoded audio frame for one or more of the memories into the respective memory.
Method and device for quantization of linear prediction coefficient and method and device for inverse quantization
A quantization apparatus comprises: a first quantization module for performing quantization without an inter-frame prediction; and a second quantization module for performing quantization with an inter-frame prediction, and the first quantization module comprises: a first quantization part for quantizing an input signal; and a third quantization part for quantizing a first quantization error signal, and the second quantization module comprises: a second quantization part for quantizing a prediction error; and a fourth quantization part for quantizing a second quantization error signal, and the first quantization part and the second quantization part comprise a trellis structured vector quantizer.
Apparatus and method for generating an error concealment signal using an adaptive noise estimation
An apparatus for generating an error concealment signal, includes: an LPC representation generator for generating a replacement LPC representation; an LPC synthesizer for filtering a codebook information using the replacement LPC representation; and a noise estimator for estimating a noise estimate during a reception of good audio frames, wherein the noise estimate depends on the good audio frames representation generator is configured to use the noise estimate estimated by the noise estimator in generating the replacement LPC representation.