Abstract: A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal, wherein the gain of the fixed excitation contribution is estimated in a sub-frame using a parameter representative of a classification of the frame. The gain of the fixed excitation contribution is then quantized in the sub-frame using the estimated gain. The device and method is used in jointly quantizing gains of adaptive and fixed contributions of an excitation in a frame of a coded sound signal. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame, the gain of the fixed excitation contribution is estimated using a parameter representative of a classification of the frame, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide a quantized gain of the fixed excitation contribution.

Abstract: A signal encoding method and device are disclosed. The method includes, when an encoding manner of a previous frame of a currently-input frame is a continuous encoding manner, predicting a comfort noise that is generated by a decoder according to the currently-input frame when the currently-input frame is encoded into an SID frame, determining an actual silence signal, determining a deviation degree between the comfort noise and the actual silence signal, determining an encoding manner of the currently-input frame according to the deviation degree, and encoding the currently-input frame according to the encoding manner of the currently-input frame. It is determined, according to the deviation degree between the comfort noise and the actual silence signal, that the encoding manner of the currently-input frame is the hangover frame encoding manner or the SID frame encoding manner, which can save communication bandwidth.

Abstract: The present invention relates to a method for performing a D2D discovery and a terminal using the same. Particularly, the present invention relates to a method for performing a D2D discovery, which compares a congestion level of another discovery slot selected randomly or according to a preset pattern with a threshold or a congestion level of a current discovery slot and determines a discovery slot to transmit a discovery message and a transmission period of the discovery message according to a result of the comparison, and a terminal using the same.

Abstract: A system, apparatus and a method for transmitting continuous audio data configured to mitigate data discontinuities in a receiving device. The method may mitigate data discontinuities by transmitting a continuous stream of audio data that has reduced changes to the audio data characteristics. The method may transmit filler audio data when no application audio data is available. The application audio data and the filler audio data are processed to reduce changes to the audio data characteristics in each stream.

Abstract: A system comprising at least one processor configured to perform: receiving a first request to access a first user profile of a first user from a first device configured to execute a first virtual assistant to interact with the first user; in response to receiving the first request, providing the first device with access to information in the first user profile so that the first virtual assistant is able to customize, based on the accessed information, its behavior when interacting with the first user; receiving a second request to access the first user profile from a second device configured to execute a second virtual assistant to interact with the first user; and in response to receiving the second request, providing the second device with access to the information so that the second virtual assistant is able to customize, based on the accessed information, its behavior when interacting with the first user.

Abstract: An apparatus for decoding an encoded audio signal including bandwidth extension control data indicating either a first harmonic bandwidth extension mode or a second non-harmonic bandwidth extension mode, includes: an input interface for receiving the encoded audio signal including the bandwidth extension control data indicating either the first harmonic bandwidth extension mode or the second non-harmonic bandwidth extension mode; a processor for decoding the audio signal using the second non-harmonic bandwidth extension mode; and a controller for controlling the processor to decode the audio signal using the second non-harmonic bandwidth extension mode, even when the bandwidth extension control data indicates the first harmonic bandwidth extension mode for the encoded signal.

Abstract: A system for carrying out voice pattern recognition and a method for achieving same. The system includes an arrangement for acquiring an input voice, a signal processing library for extracting acoustic and prosodic features of the acquired voice, a database for storing a recognition dictionary, at least one instance of a prosody detector for carrying out a prosody detection process on extracted respective prosodic features, communicating with an end user application for applying control thereto.

Abstract: A speech masking apparatus includes a microphone and a speaker. The microphone can detect a human voice. The speaker can output a masking language which can include phonemes resembling human speech. At least one component of the masking language can have a pitch, a volume, a theme, and/or a phonetic content substantially matching a pitch, a volume, a theme, and/or a phonetic content of the voice.

Abstract: A method and system for achieving emotional text to speech. The method includes: receiving text data; generating emotion tag for the text data by a rhythm piece; and achieving TTS to the text data corresponding to the emotion tag, where the emotion tags are expressed as a set of emotion vectors; where each emotion vector includes a plurality of emotion scores given based on a plurality of emotion categories. A system for the same includes: a text data receiving module; an emotion tag generating module; and a TTS module for achieving TTS, wherein the emotion tag is expressed as a set of emotion vectors; and wherein emotion vector includes a plurality of emotion scores given based on a plurality of emotion categories.

Abstract: A mechanism is disclosed to adapt in real-time the encoding bitrate of a video source in an adaptive video streaming system with the aim of avoiding playback interruptions and to guarantee the best possible video quality. The mechanism is made of a feedback loop in which a controller computes the encoding bitrate that the sender (the video server) transmits to a receiver (client) employing a packet switching network such as the Internet. The mechanism can be used in video streaming systems employing scalable encoding (f.i. H.264 SVC, VP8, VP9) or multi bit-rate encoding (stream-switching). The automatic control that computes the video encoding bitrate can be executed either at the server or at the client.

Abstract: A stereo signal encoding device is provided that enables a lower bitrate without decreasing quality when applying an intermittent transmission technique to a stereo signal. A stereo encoding unit generates first stereo encoded data by encoding the stereo signal when the stereo signal of the current frame is an audio section. A stereo DTX encoding unit is a means for encoding the stereo signal when the stereo signal of the current frame is a non-audio section. The stereo DTX encoding unit generates second stereo encoded data by encoding each of a monaural signal spectral parameter that is a spectral parameter of a monaural signal generated using the first channel signal and the second channel signal, first channel signal information relating to the first channel signal, and second channel signal information relating to the second channel signal.

Abstract: Various of the disclosed embodiments relate to systems and methods for extracting audio information, e.g. a textual description of speech, from a speech recording while retaining the anonymity of the speaker. In certain embodiments, a third party may perform various aspects of the anonymization and speech processing. Certain embodiments facilitate anonymization in compliance with various legislative requirements even when third parties are involved.

Abstract: A base station receives a message comprising at least one index identifying a subset of a second plurality of beamforming codewords. Each of the second plurality of beamforming codewords is identifiable by an index presented by a number of bits. The number of bits is greater than or equal to log2(N), N being number of the second plurality of beamforming codewords, and less than number of bits in representation of a corresponding beamforming codeword. The base station transmits signals employing a first plurality of beamforming codewords being selected employing the subset of the second plurality of beamforming codewords.

Abstract: There is provided a method of using an adaptive tilt compensation by a speech decoder. The method comprises receiving a bit stream including a plurality of parameters representative of a speech signal; identifying an adaptive code vector and a fixed code vector using the plurality of parameters; scaling the adaptive code vector and the fixed code vector to generate a scaled adaptive code vector and a scaled fixed code vector; summing the scaled adaptive code vector and the scaled fixed code vector to generate a synthesized output; calculating a first reflection coefficient based on the plurality of parameters representative of the speech signal; multiplying the first reflection coefficient by a factor to generate a tilt factor; and applying the tilt factor to the synthesized output based on an encoding bit rate.

Abstract: The present invention pertains to a speech decoding device that is capable of preventing degradation in sound quality associated with an adjustment of the slope of a spectrum of an output signal (a decoding signal), making it less likely that a loss of bandwidth sensitivity due to the attenuation of a higher band region is perceived. For each frame of the bandwidth extension layer decoding signal, a filter assessment unit (304) determines whether or not to apply a low-pass filter to the bandwidth extension layer decoding signal on the basis of a change in energy in the bandwidth extension layer decoding signal. A low-pass filtering unit (306) filters the bandwidth extension layer decoding signal of the frames to which the low-pass filter is to be applied, as determined by the filter assessment unit (304), using the low-pass filter.

Abstract: The embodiments of the present invention disclose a method and an apparatus for sub-sampling of a codebook in the LTE-A system where a precoding matrix W is a product of two matrices W1 and W2, i.e. W=W1W2, codebooks for W, W1 and W2 are denoted as C, C1 and C2 respectively, and r indicates a rank. The method includes sub-sampling the codebook C such that the sub-sampled codebook C has a size of equal to or less than 4 bits. In the sub-sampling, codewords evenly distributed in the codebook C are extracted, where some or all of the codewords are in a form of discrete Fourier transform (DFT) vector to be suitable for evenly linear arrays, and the other codewords are suitable for cross-polarized linear arrays.

Abstract: A method of updating the state of a decoder that decodes a wide-band signal including a plurality of sub-band signals, including: receiving the plurality of sub-band signals; for each sub-band signal, storing portions of that sub-band signal in a respective buffer; responsive to determining that a portion of the wide-band signal is degraded, performing a packet loss concealment algorithm to determine wide-band replacement data for the degraded portion; selecting a portion of the sub-band signal stored in each buffer in dependence on the determined wide-band replacement data; and updating the state of the decoder using the selected portions.

Abstract: Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.

Abstract: A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

Abstract: A method and system for achieving emotional text to speech. The method includes: receiving text data; generating emotion tag for the text data by a rhythm piece; and achieving TTS to the text data corresponding to the emotion tag, where the emotion tags are expressed as a set of emotion vectors; where each emotion vector includes a plurality of emotion scores given based on a plurality of emotion categories. A system for the same includes: a text data receiving module; an emotion tag generating module; and a TTS module for achieving TTS, wherein the emotion tag is expressed as a set of emotion vectors; and wherein emotion vector includes a plurality of emotion scores given based on a plurality of emotion categories.

Abstract: In a coder, a method for producing forward aliasing cancellation (FAC) parameters for cancelling time-domain aliasing caused to a coded audio signal in a first transform-coded frame by a transition between the first transform-coded frame using a first coding mode with overlapping window and a second frame using a second coding mode with non-overlapping window, comprising: calculating a FAC target representative of a difference between the audio signal of the first frame prior to coding and a sum of synthesis of the coded audio signal of the first transform-coded frame and a time reversed version of last synthesis samples of the second frame as well as a zero point response of a synthesis filter used in the second frame; and weighting the FAC target to produce the FAC parameters. In a decoder, weighted forward aliasing cancellation (FAC) parameters are received and inverse weighted to produce a FAC synthesis.

Abstract: A video processing method and system and a related device are disclosed, an encoder sends a method for compressing a pixel value of a video image to a decoder, the method includes a mapping between a pixel value of the video image before compression and a pixel value of the video image after the compression; compressing, by the encoder according to the method for compressing a pixel value, the pixel value of the video image to obtain a video image to be encoded; encoding, by the encoder, the video image to be encoded to obtain a video image encoded stream; and sending, by the encoder, the video image encoded stream to the decoder, the decoder receives and decodes the video image encoded stream to obtain a decoded video image, and decompresses a pixel value of the decoded video image to restore the pixel value of the video image before compression.

Abstract: In accordance with an embodiment, a method of decoding an audio/speech signal includes decoding an excitation signal based on an incoming audio/speech information, determining a stability of a high frequency portion of the excitation signal, smoothing an energy of the high frequency portion of the excitation signal based on the stability of the high frequency portion of the excitation signal, and producing an audio signal based on smoothing the high frequency portion of the excitation signal.

Abstract: Certain aspects of the present disclosure relate to a method for quantizing signals and reconstructing signals, and/or encoding or decoding data for storage or transmission. Points of a signal may be determined as local extrema or points where an absolute rise of the signal is greater than a threshold. The tread and value of the points may be quantized, and certain of the quantizations may be discarded before the quantizations are transmitted. After being received, the signal may be reconstructed from the quantizations using an iterative process.

Abstract: Compressibility-based reallocation of initial bit allocations for frames of an audio signal is described. Applications to redundancy-based retransmission of critical frames (e.g., for fixed-bit-rate modes of speech codec operation) are also described.

Abstract: A method of signal processing according to one embodiment includes calculating an envelope of a first signal that is based on a low-frequency portion of a speech signal, calculating an envelope of a second signal that is based on a high-frequency portion of the speech signal, and calculating a plurality of gain factor values according to a time-varying relation between the envelopes of the first and second signal. The method includes attenuating, based on a variation over time of a relation between the envelopes of the first and second signals, at least one of the plurality of gain factor values. In one example, the variation over time of a relation between the envelopes is indicated by at least one distance among the plurality of gain factor values.

Abstract: A mixed time-domain/frequency-domain coding device and method for coding an input sound signal, wherein a time-domain excitation contribution is calculated in response to the input sound signal. A cut-off frequency for the time-domain excitation contribution is also calculated in response to the input sound signal, and a frequency extent of the time-domain excitation contribution is adjusted in relation to this cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to the input sound signal, the adjusted time-domain excitation contribution and the frequency-domain excitation contribution are added to form a mixed time-domain/frequency-domain excitation constituting a coded version of the input sound signal. In the calculation of the time-domain excitation contribution, the input sound signal may be processed in successive frames of the input sound signal and a number of sub-frames to be used in a current frame may be calculated.

Abstract: An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an enc

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: Disclosed is an encoding device that improves the quality of a decoded signal in a hierarchical coding (scalable coding) method, wherein a band to be quantized is selected for every level (layer). The encoding device (101) is equipped with a second layer encoding unit (205) that selects a first band to be quantized of a first input signal from among a plurality of sub-bands, and that generates second layer encoding information containing first band information of said band; a second layer decoding unit (206) that generates a first decoded signal using the second layer encoding information; an addition unit (207) that generates a second input signal using the first input signal and the first decoded signal; and a third layer encoding unit (208) that selects a second band to be quantized of the second input signal using the first decoded signal, and that generates third layer encoding information.

Abstract: An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.

Abstract: A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

Abstract: In one embodiment of the present invention, a method of decoding an encoded audio bitstream and generating frequency bandwidth extension includes decoding the audio bitstream to produce a decoded low band audio signal and generate a low band excitation spectrum corresponding to a low frequency band. A sub-band area is selected from within the low frequency band using a parameter which indicates energy information of a spectral envelope of the decoded low band audio signal. A high band excitation spectrum is generated for a high frequency band by copying a sub-band excitation spectrum from the selected sub-band area to a high sub-band area corresponding to the high frequency band. Using the generated high band excitation spectrum, an extended high band audio signal is generated by applying a high band spectral envelope. The extended high band audio signal is added to the decoded low band audio signal to generate an audio output signal having an extended frequency bandwidth.

Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

Abstract: A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: Disclosed are an apparatus for coding a signal in a communication system including: a coding unit configured to code voice and audio signals based on a code excited linear prediction (CELP) coding method; a residual signal calculation unit configured to calculate residual signals of the voice and audio signals; a frequency transform unit configured to transform the residual signal into a signal in a frequency domain; an energy calculation unit configured to use frequency coefficients of the residual signals to calculate frequency energy of the residual signals; an energy concentration calculation unit configured to calculate energy concentrations of each vector dimension of the residual signals from the frequency energy of the residual signals; and a vector dimension determination unit configured to compare the energy concentrations of each vector dimension to determine targeted vector dimensions of the residual signals.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: An apparatus and method for encoding/decoding a speech signal which determines a variable bit rate based on reserved bits obtained from a target bit rate, is provided. The variable bit rate is determined based on a source feature of the speech signal and the reserved bits is obtained based on the target bit rate. The apparatus for encoding the speech signal may include a linear predictive (LP) analysis unit/quantization unit to determine an immittance spectral frequencies (ISF) index, a closed loop pitch search unit, a fixed codebook search unit, a gain vector quantization (VQ) unit to determine a gain vector quantization (VQ) index, and a bit rate control unit to control at least two indexes of the ISF index, the pitch index, the code index, and the gain VQ index to be encoded to be variable bit rates based on a source feature of a speech signal and the reserved bits.

Abstract: In one embodiment, a method of signal processing includes calculating an envelope of a first signal that is based on a low-frequency portion of a speech signal, calculating an envelope of a second signal that is based on a high-frequency portion of the speech signal, and calculating a first plurality of gain factor values according to a time-varying relation between the envelopes of the first and second signals. The method includes, based on the first plurality of gain factor values, calculating a plurality of smoothed gain factor values. In one example, each of the plurality of smoothed gain factor values is based on a weighted sum of at least one of the first plurality of gain factor values and at least one smoothed gain factor value. In another example, the sum is adaptively weighted based on at least one distance among the plurality of gain factor values.

Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract: In one embodiment, a method executed by at least one processor includes receiving text from submitted by a user. The method also includes determining a text score for the received text by comparing a first set of phrases included in the received text to a second set of phrases. The second set of phrases includes phrases from stored text. The stored text includes stored text known to be genuine and stored text known to be fraudulent. The method also includes determining that the received text is fraudulent based on the text score.

Abstract: An electronic device for determining a set of pitch cycle energy parameters is described. The electronic device includes a processor and executable instructions stored in memory. The electronic device obtains a frame, a set of filter coefficients and a residual signal based on the frame and the set of filter coefficients. The electronic device determines a set of peak locations based on the residual signal and segments the residual signal such that each segment includes one peak. The electronic device determines a first set of pitch cycle energy parameters based on a frame region between two consecutive peak locations and maps regions between peaks in the residual signal to regions between peaks in a synthesized excitation signal to produce a mapping. The electronic device determines a second set of pitch cycle energy parameters based on the first set of pitch cycle energy parameters and the mapping.

Abstract: An encoder whereby the bit efficiency of encoding can be improved, thereby improving the qualities of signals as decoded. In the encoder: a time-frequency converting unit (101) converts signals, which are to be encoded, to frequency domain signals; an adaptive spectrum formation encoding unit (102) determines an effective range in the frequency band of the frequency domain signals; and a pulse vector encoding unit (103) pulse vector encodes only the signal components within the effective range.

Abstract: A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: There is disclosed an encoder apparatus whereby, when a band expanding technique for encoding, based on the spectral data of a lower frequency portion, the spectral data of a higher frequency portion is applied to a lower layer in a hierarchical encoding/decoding system, an efficient encoding can be performed in an upper layer as well, thereby improving the decoded-signal quality. In an encoder apparatus (101), a second layer decoder unit (207) calculates a spectrum (differential spectrum), which is to be encoded in a third layer encoder unit (210) that is an upper layer of the second layer decoder unit (207), by applying such an ideal gain (first gain parameter a1) that minimizes the energy of the differential spectrum.

Abstract: Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal.

Abstract: In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values.

Abstract: An apparatus for modifying an input audio signal has an excitation determiner, a storage device and a signal modifier. The excitation determiner determines a value of an excitation parameter of a subband of a plurality of subbands of the input audio signal based on an energy content of the subband. Further, the storage device stores a lookup table containing a plurality of spectral weighting factors. A spectral weighting factor of the plurality of spectral weighting factors is associated to a predefined value of the excitation parameter and a subband of the plurality of subbands. The storage device provides a spectral weighting factor corresponding to the determined value of the excitation parameter and corresponding to the subband, the value of the excitation parameter is determined for. Further, the signal modifier modifies a content of the subband of the audio signal, the value of the excitation parameter is determined for, based on the provided spectral weighting factor to provide a modified subband.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.