Abstract: An encoding method includes dividing a to-be-encoded time-domain signal into a low band signal and a high band signal, performing encoding on the low band signal to obtain a low frequency encoding parameter, performing encoding on the high band signal to obtain a high frequency encoding parameter, obtaining a synthesized high band signal, performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal, and calculating a high frequency gain based on the high band signal and the short-time filtering signal.

Abstract: Concealing a lost audio frame of a received audio signal is provided by performing a sinusoidal analysis of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies.

Abstract: An audio processing apparatus 100 is apparatus for generating a training data in speaker recognition. The audio processing apparatus 100 includes a data acquisition unit configured to acquire an audio signal that is a source of the training data as sample data, a data generation unit configured to executes signal processing on the acquired sample data, and to generates a new audio signal as the training data whose similarity with the sample data is within the set range.

Abstract: An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i?j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Abstract: A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described.

Abstract: Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determined based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal, a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.

Abstract: Systems and methods for generating training data are described herein. Pieces of metadata captured by a plurality of networked sensor systems can be captured, where each piece of metadata is associated with a specific set of sensor data captured by one of the plurality of networked sensor systems and includes a set of characteristics for the specific set of captured sensor data. A probabilistic model can be generated based on the received metadata and simulations can be performed based upon a training corpus by generating multiple scenarios, and, for each scenario, a scenario specific version of a particular annotated sample is generated by performing a simulation using the particular annotated sample. The scenario specific versions of annotated samples from the training corpus can be stored as a training data set on the at least one network device.

Abstract: A stereo encoding method and apparatus, and a stereo decoding method and apparatus are disclosed. The stereo encoding method includes: performing downmix processing on a left channel signal of a current frame and a right channel signal of the current frame, to obtain a primary channel signal of the current frame and a secondary channel signal of the current frame; and when determining that a frame structure similarity value falls within a frame structure similarity interval, performing differential encoding on a pitch period of the secondary channel signal by using an estimated pitch period value of the primary channel signal, to obtain a pitch period index value of the secondary channel signal, where the pitch period index value of the secondary channel signal is used to generate a to-be-sent stereo encoded bitstream.

Abstract: Techniques for motion mitigation for uplink power control are disclosed. In one embodiment, a method for use in a satellite communication system comprises: generating a power margin associated with motion of an antenna of a satellite terminal; and generating a first power limit representing a maximum transmit power for the antenna based, at least in part, on the power margin.

Abstract: The present disclosure provides an authentication method, an authentication device, an electronic device and a storage medium. The authentication method includes: receiving target voice data; obtaining a first voiceprint feature parameter corresponding to the target voice data from a device voiceprint model library; performing a first encryption process on the first voiceprint feature parameter with a locally stored private key to generate to-be-verified data; transmitting the to-be-verified data to a server, so that the server uses a public key which matches the private key to decrypt the to-be-verified data to obtain the first voiceprint feature parameter, and performs authentication on the first voiceprint feature parameter to obtain an authentication result; receiving the authentication result returned by the server.

Abstract: Systems and methods are provided for applying attributes to subtitles. One example method includes accessing a subtitle file, wherein the subtitle file comprises one or more subtitles, and identifying an attribute to apply to at least a subset of the subtitles. The subtitle file is amended indicate an attribute to apply to at least a subset of the subtitles to create an amended subtitle file. At a computing device, the subtitles of the amended subtitle file are generated for display, wherein the attribute is applied to the subset of the subtitles.

Abstract: Methods, systems, and devices for a decision directed multi-modulus searching algorithm are described. A receiver may receive a signal including a set of data symbols. The receiver may iteratively determine a set of centroids for demodulating the set of data symbols (e.g., as part of a training procedure). The centroids may be used to demodulate the set of data symbols according to a modulation constellation associated with the set of data symbols. The training procedure may include, for each data symbol of a subset of data symbols, assigning a centroid of the set of centroids to each data symbol and updating the set of centroids based on assigning the centroid to each data symbol. The receiver may demodulate the set of data symbols based on the updated set of centroids.

Abstract: The present disclosure relates to systems and methods for autonomous driving. The systems may obtain driving information associated with a vehicle; determine a state of the vehicle; determine one or more candidate control signals and one or more evaluation values corresponding to the one or more candidate control signals based on the driving information and the state of the vehicle by using a trained control model; select a target control signal from the one or more candidate control signals based on the one or more evaluation values; and transmit the target control signal to a control component of the vehicle.

Abstract: Systems and methods for automatic test pattern generation (ATPG) for parametric faults are described. A model may be constructed to predict a measurement margin for an integrated circuit (IC) design based on a random sample of random variables. A set of failure events may be determined for the IC design using the model, where each failure event may correspond to a set of values of the random variables that is expected to cause a metric for the IC design to violate a threshold.

Abstract: Described herein is a method of decoding an audio or speech signal, the method including the steps of: (a) receiving, by a decoder, a coded bitstream including the audio or speech signal and conditioning information; (b) providing, by a bitstream decoder, decoded conditioning information in a format associated with a first bitrate; (c) converting, by a converter, the decoded conditioning information from the format associated with the first bitrate to a format associated with a second bitrate; and (d) providing, by a generative neural network, a reconstruction of the audio or speech signal according to a probabilistic model conditioned by the conditioning information in the format associated with the second bitrate. Described are further an apparatus for decoding an audio or speech signal, a respective encoder, a system of the encoder and the apparatus for decoding an audio or speech signal as well as a respective computer program product.

Abstract: A spectrum filler for filling non-coded residual sub-vectors of a transform coded audio signal includes a sub-vector compressor configured to compress actually coded residual sub-vectors. A sub-vector rejecter is configured to reject compressed residual sub-vectors that do not fulfill a predetermined sparseness criterion. A sub-vector collector is configured to concatenate the remaining compressed residual sub-vectors to form a first virtual codebook. A coefficient combiner is configured to combine pairs of coefficients of the first virtual codebook to form a second virtual codebook. A sub-vector filler is configured to fill non-coded residual sub-vectors below a predetermined frequency with coefficients from the first virtual codebook, and to fill non-coded residual sub-vectors above the predetermined frequency with coefficients from the second virtual codebook.

Abstract: Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determining based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal is a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.

Abstract: Systems and methods for generating training data are described herein. Pieces of metadata captured by a plurality of networked sensor systems can be captured, where each piece of metadata is associated with a specific set of sensor data captured by one of the plurality of networked sensor systems and includes a set of characteristics for the specific set of captured sensor data. A probabilistic model can be generated based on the received metadata and simulations can be performed based upon a training corpus by generating multiple scenarios, and, for each scenario, a scenario specific version of a particular annotated sample is generated by performing a simulation using the particular annotated sample. The scenario specific versions of annotated samples from the training corpus can be stored as a training data set on the at least one network device.

Abstract: Disclosed are a method and a system for speech recognition, an electronic device and a storage medium, which relates to the technical field of speech recognition. Embodiments of the application comprise performing encoded representation on an audio to be recognized to obtain an acoustic encoded state vector sequence of the audio to be recognized; performing sparse encoding on the acoustic encoded state vector sequence of the audio to be recognized to obtain an acoustic encoded sparse vector; determining a text prediction vector of each label in a preset vocabulary; recognizing the audio to be recognized and determining a text content corresponding to the audio to be recognized according to the acoustic encoded sparse vector and the text prediction vector. The acoustic encoded sparse vector of the audio to be recognized is obtained by performing sparse encoding on the acoustic encoded state vector of the audio to be recognized.

Abstract: Disclosed are a method for coding a residual signal of LPC coefficients based on collaborative quantization and a computing device for performing the method. The residual signal coding method includes: generating encoded LPC coefficients and LPC residual signals by performing LPC analysis and quantization on an input speech; Determining a predicted LPC residual signal by applying the LPC residual signal to cross module residual learning; Performing LPC synthesis using the coded LPC coefficients and the predicted LPC residual signal; It may include the step of determining an output speech that is a synthesized output according to a result of performing the LPC synthesis.

Abstract: Concealing a lost audio frame of a received audio signal is provided by performing a sinusoidal analysis (81) of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame (83) for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies.

Abstract: During text-to-speech processing, a speech model creates synthesized speech that corresponds to input data. The speech model may include an encoder for encoding the input data into a context vector and a decoder for decoding the context vector into spectrogram data. The speech model may further include a voice decoder that receives vocal characteristic data representing a desired vocal characteristic of synthesized speech. The voice decoder may process the vocal characteristic data to determine configuration data, such as weights, for use by the speech decoder.

Abstract: The present disclosure generally relates to apparatus, software and methods for predicting future network traffic. The disclosed apparatus, software and methods alleviate congestion and/or increase overall traffic flow by providing methods for reallocating future idle capacity.

Abstract: An audio signal processing device comprises a discontinuity detector configured to determine an occurrence of a discontinuity from a sudden increase of an amplitude of decoded audio obtained by decoding the first audio packet which is received correctly after an occurrence of a packet loss, and a discontinuity corrector for correcting the discontinuity of the decoded audio by changing, in a state buffer, a distance between elements of Immittance Spectral Pair/Immittance Spectral Frequency (ISF/LSF) parameters of a past frame.

Abstract: An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i?j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Abstract: A communication device is configured to correlate a first signal with a second signal at a designated interval, the second signal corresponding to the first signal and being received by the communication device where the other communication device transmits a signal including a pulse as the first signal, convert a correlation computation result that is a result of correlating the first signal with the second signal at the designated interval into a format including a matrix product of an expanded modal matrix and an expanded signal vector, the expanded modal matrix including a plurality of elements indicating the correlation computation result obtained when assuming that the signals are received at respective set times, the expanded signal vector being a vector including a plurality of elements, each of which indicates whether or not there is a signal received at each of the set times and amplitude and phase of the signal.

Abstract: A frame error concealment method based on frames including transform coefficient vectors including the following steps: It tracks sign changes between corresponding transform coefficients of predetermined sub-vectors of consecutive good stationary frames. It accumulates the number of sign changes in corresponding sub-vectors of a predetermined number of consecutive good stationary frames. It reconstructs an erroneous frame with the latest good stationary frame, but with reversed signs of transform coefficients in sub-vectors having an accumulated number of sign changes that exceeds a predetermined threshold.

Abstract: A technique for encoding a multichannel audio encoding is provided that includes quantizing a set of first LP filter coefficients for an audio signal in a first channel using a predefined first quantizer; and quantizing a set of second LP filter coefficients for an audio signal in a second channel on the basis of the quantized set of first LP filter coefficients. The quantization of the set of second LP filter coefficients includes: deriving, on basis of the quantized set of first LP filter coefficients by using a predefined predictor, a set of predicted LP filter coefficients for the audio signal in said second channel, computing prediction error as a difference between respective LP coefficients of the set of second LP filter coefficients and the set of predicted LP filter coefficients, and quantizing the prediction error.

Abstract: The speech command issued to a voice activated/controlled system is anonymized so that biometric voice data of the speaker may not be received by the voice activated/controlled system. A spoken audio command is converted to text, which is then converted to a synthesized voice signal. The synthesized voice signal is then provided to the voice-activated/controlled device. The synthesized voice signal may be provided to the voice-activated device within a sound shield or enclosure so that the original speech command issued by the speaker may not be received by the voice-activated/controlled system. In this way, the speaker's actual voice and related data may be kept private and secure.

Abstract: A computer-implemented system and method provide an audio label for a noise signal. The computer-implemented system and method include receiving an audio input and obtaining the noise signal from the audio input. The computer-implemented system and method include extracting audio features of the noise signal. The computer-implemented system and method include determining and outputting an audio label for the extracted audio features of the noise signal based on machine learning data.

Abstract: A method enables identification of a similarity level between a user-provided data item and a data item within a set of data documents. The method includes a representation generator determining, for each term in an enumeration of terms, occurrence information. The representation generator generates, for each term, a sparse distributed representation (SDR) using the occurrence information. The method includes receiving, by a filtering module, a filtering criterion. The method includes generating, by the representation generator, for the filtering criterion, at least one SDR. The method includes generating, by the representation generator, for a first of a plurality of streamed documents received from a data source, a compound SDR. The method includes determining, by a similarity engine executing on the second computing device, a distance between the filtering criterion SDR and the generated compound SDR. The method includes acting on the first streamed document, based upon the determined distance.

Abstract: Embodiments of the present disclosure provide a method and a device for generating far-field speech data, a computer device and a computer readable storage medium. The method includes obtaining environmental noise in real environment and adjusting near-field speech data in a near-field speech data set based on the environmental noise, further includes generating far-field speech data based on adjusted near-field speech data and the environmental noise.

Abstract: A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A?1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and/or (B?1) an index Q? commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1?.

Abstract: Systems, apparatuses, and methods for converting data to a tiling format when implementing convolutional neural networks are disclosed. A system includes at least a memory, a cache, a processor, and a plurality of compute units. The memory stores a first buffer and a second buffer in a linear format, where the first buffer stores convolutional filter data and the second buffer stores image data. The processor converts the first and second buffers from the linear format to third and fourth buffers, respectively, in a tiling format. The plurality of compute units load the tiling-formatted data from the third and fourth buffers in memory to the cache and then perform a convolutional filter operation on the tiling-formatted data. The system generates a classification of a first dataset based on a result of the convolutional filter operation.

Abstract: In accordance with some embodiments, an apparatus for privacy protection is provided. The apparatus includes an audio output device arranged to output sound directed to an audio input device of a second device. The apparatus further includes an audio coupling interface arranged to provide a cavity for the audio output device and the audio input device of the second device. The apparatus also includes a spectral shaper, coupled to the audio output device, operable to apply a spectral envelope to an audio signal in order to produce a shaped audio signal, wherein the shaped audio signal is selectively coupled to the audio output device.

Abstract: An audio signal encoding method and a mobile phone, where the audio signal encoding method includes obtaining a digital audio signal in time domain; transforming the digital audio signal in time domain to an audio signal in frequency domain, which comprises a current frame comprises a plurality of subbands; obtaining, reference parameters of the plurality of subbands; encoding, using a HQ algorithm, the current frame to obtain an encoded audio signal when the reference parameters meet a preset parameter condition; and transmitting the encoded audio signal via a network. The audio signal encoding method and the mobile phone help improve encoding quality or encoding efficiency in audio signal encoding.

Abstract: A computer-implemented method according to one embodiment includes creating a clean dictionary, utilizing a clean signal, creating a noisy dictionary, utilizing a first noisy signal, determining a time varying projection, utilizing the clean dictionary and the noisy dictionary, and denoising a second noisy signal, utilizing the time varying projection.

Abstract: An audio signal processing device comprises a discontinuity detector configured to determine an occurrence of a discontinuity from a sudden increase of an amplitude of decoded audio obtained by decoding the first audio packet which is received correctly after an occurrence of a packet loss, and a side information encoder configured to encode side information about the discontinuity.

Abstract: A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A-1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and/or (B-1) an index Q? commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1?.

Abstract: A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A-1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and/or (B-1) an index Q? commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1?.

Abstract: A time-varying pitch of a signal may be estimated by processing a sequence of frames of the speech signal. An estimated fractional chirp rate may be computed for each frame of the sequence of frames, and the estimated fractional chirp rates may be used to compute a pitch template for the sequence, where the pitch template indicates the time-varying pitch of the signal subject to a scale factor. A first pitch estimate for each frame of the sequence of frames may be computed by computing a scale factor and multiplying the pitch template by the scale factor. A second pitch estimate may be computed from the first pitch estimate by identifying peaks in the frequency representations using the first pitch estimates and fitting a parametric function to the peaks.

Abstract: A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A-1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and/or (B-1) an index Q? commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1?.

Abstract: Disclosed are a method and apparatus for encoding and decoding a high frequency for bandwidth extension. The method includes: estimating a weight; and generating a high frequency excitation signal by applying the weight between random noise and a decoded low frequency spectrum.

Abstract: Embodiments are disclosed for recognizing speech in a computing system. An example speech recognition method includes receiving metadata at a generation unit that includes a database of accented substrings, generating, via the generation unit, accent-corrected phonetic data for words included in the metadata, the accent-corrected phonetic data representing different pronunciations of the words included in the metadata based on the accented substrings stored in the database, receiving, at a voice recognition engine, extracted speech data derived from utterances input by a user to the speech recognition system, and receiving, at the voice recognition engine, the accent-corrected phonetic data.

Abstract: A time-varying pitch of a signal may be estimated by processing a sequence of frames of the speech signal. An estimated fractional chirp rate may be computed for each frame of the sequence of frames, and the estimated fractional chirp rates may be used to compute a pitch template for the sequence, where the pitch template indicates the time-varying pitch of the signal subject to a scale factor. A first pitch estimate for each frame of the sequence of frames may be computed by computing a scale factor and multiplying the pitch template by the scale factor. A second pitch estimate may be computed from the first pitch estimate by identifying peaks in the frequency representations using the first pitch estimates and fitting a parametric function to the peaks.

Abstract: An audio encoder for encoding an audio signal, includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal.

Abstract: A speech-enabled dialog system responds to a plurality of wake-up phrases. Based on which wake-up phrase is detected, the system's configuration is modified accordingly. Various configurable aspects of the system include selection and morphing of a text-to-speech voice; configuration of acoustic model, language model, vocabulary, and grammar; configuration of a graphic animation; configuration of virtual assistant personality parameters; invocation of a particular user profile; invocation of an authentication function; and configuration of an open sound. Configuration depends on a target market segment. Configuration also depends on the state of the dialog system, such as whether a previous utterance was an information query.

Abstract: An audio signal processing device comprises a discontinuity detector configured to determine an occurrence of a discontinuity from a sudden increase of an amplitude of decoded audio obtained by decoding the first audio packet which is received correctly after an occurrence of a packet loss, and a discontinuity corrector for correcting the discontinuity of the decoded audio.

Abstract: An audio signal, having first and second regions of frequency spectrum, is coded. Spectral peaks in the first region are encoded by a first coding method. For a segment of the audio signal, a relation between energy of bands in the first and second regions is determined. A relation between the energy of the band in the second region and energy of neighboring bands in the second region is determined. A determination is made whether available bits are sufficient for encoding at least one non-peak segment of the first region and the band in the second region. Responsive to first and second relations fulfilling a respective predetermined criterion and a sufficient number of bits, encoding the band in the second region using a second coding method different from the first coding method, and otherwise, subjecting the band in the second region to BandWidth Extension BWE or noise fill.

Abstract: A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame, the excitation of a next frame is obtained according to the reduced or limited pitch gain value of the first subframe, and the next frame is encoded according to the obtained excitation. The method is used for a voiced speech class.