Abstract: An artificial intelligence, AI, planning controller control the timing of when a plan (16) to accomplish a task (14) is synthesized. The AI planning controller in this regard determines a quiescent phase (20) during which values of at least some predicates describing a state of the system (12) will remain stable. The AI planning controller then controls artificial intelligence planning to synthesize the plan (16) during at least some of the quiescent phase (20).

Abstract: A method, executed by a processor for compressing an audio signal in multiple layers, may comprise: (a) restoring, in a highest layer, an input audio signal as a first signal; (b) restoring, in at least one intermediate layer, a signal obtained by subtracting an upsampled signal, which is obtained by upsampling the audio signal restored in the highest layer or an immediately previous intermediate layer, from the input audio signal as a second signal; and (c) restoring, in a lowest layer, a signal obtained by subtracting an upsampled signal, which is obtained by upsampling the audio signal restored in an intermediate layer immediately before the lowest layer, from the input audio signal as a third signal, wherein the first signal, the second signal, and the third signal are combined to output a final restoration audio signal.

Abstract: An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

Abstract: An encoding parameter adjustment method is performed at a computer device. The method includes: obtaining a first audio signal, and determining a psychoacoustic masking threshold within a service frequency band in the first audio signal; obtaining a second audio signal, and determining a background environmental noise estimation value of the frequency within the service frequency band in the second audio signal; determining a masking tag corresponding to the service frequency band according to the psychoacoustic masking threshold of the first audio signal and the background environmental noise estimation value of the second audio signal; determining a masking rate of the service frequency band according to the masking tag corresponding to the frequency within the service frequency band; determining a first reference bit rate according to the masking rate of the service frequency band; and configuring an encoding bit rate of an audio encoder based on the first reference bit rate.

Abstract: Disclosed is a method of processing a residual signal for audio coding and an audio coding apparatus. The method learns a feature map of a reference signal through a residual signal learning engine including a convolutional layer and a neural network and performs learning based on a result obtained by mapping a node of an output layer of the neural network and a quantization level of index of the residual signal.

Abstract: A decoding apparatus includes: a bandwidth extending part 25 obtaining a decoded extended frequency spectrum sequence by arranging samples based on K samples included in a frequency-domain sample sequence obtained by decoding, on a higher side than the frequency-domain sample sequence; and a fricative sound adjustment releasing part 23 obtaining, if inputted information indicating whether a hissing sound or not indicates being a hissing sound, what is obtained by exchanging all or a part of a low-side frequency sample sequence existing on a lower side than a predetermined frequency in the decoded extended frequency spectrum sequence for all or a part of a high-side frequency sample sequence existing on a higher side than the predetermined frequency in the decoded extended frequency spectrum sequence as an adjusted frequency spectrum sequence, the number of all or the part of the high-side frequency spectrum sequence being the same as the number of all or the part of the low-side frequency spectrum sequence.

Abstract: An audio signal decoder for providing an upmix signal representation in dependence on a downmix signal representation and an object-related parametric information includes an object separator configured to decompose the downmix signal representation, to provide a first audio information describing a first set of one or more audio objects of a first audio object type and a second audio information describing a second set of one or more audio objects of a second audio object type, in dependence on the downmix signal representation and using at least a part of the object-related parametric information.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Multi-channel audio signals are coded into a monaural audio signal and information allowing to recover the multi-channel audio signal from the monaural audio signal and the information. The information is generated by determining a first portion of the information for a first frequency region of the multi-channel audio signal, and by determining a second portion of the information for a second frequency region of the multi-channel audio signal. The second frequency region is a portion of the first frequency region and thus is a sub-range of the first frequency region. The information is multi-layered enabling a scaling of the decoding quality versus bit rate.

Abstract: The present invention relates to a machine implemented method for spectral analysis that determines a measure of cross coherence between application of two spectral estimation filters to data; and identifies a spectral feature of the measure of cross coherence. One example embodiment of the present invention provides a complete statistical summary of the joint dependence of the Bartlett and Capon power spectral statistics, showing that the coupling is expressible via a 2×2 complex Wishart matrix, where the degree coupling is determined by a single measure of cross coherence defined herein. This measure of coherence leads to a new two-dimensional algorithm capable of yielding significantly better resolution than the Capon algorithm, often commensurate with but at times exceeding finite sample based MUSIC.

Abstract: An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal.

Abstract: An encoding method of an encoder is provided. The encoder generates first MDCT coefficients by transforming an input signal, and generates MDCT indices by quantizing the first MDCT coefficients. The encoder generates second MDCT coefficients by dequantizing the MDCT indices, and calculates MDCT residual coefficients using differences between the first MDCT coefficients and the second MDCT coefficients. The encoder generates a residual index by encoding the MDCT residual coefficients, and generates gain indices corresponding to gains from the first MDCT coefficients and the second MDCT coefficients.

Abstract: An audio communication device comprises an input connectable to a narrowband audio signal source. The input can receive a narrowband audio signal having a first bandwidth. An extraction unit is connected to the input and arranged to extract a plurality of narrowband parameters from the narrowband audio signal. An extrapolation unit is connected to receive the plurality of narrowband parameters and arranged to generate a plurality of wideband parameters from the plurality of narrowband parameters. The extrapolation unit comprises one or more adaptive neuro-fuzzy inference system modules. The device further comprises a synthesis unit connected to receive the plurality of wideband parameters and arranged to generate, using the wideband parameters, a synthesized wideband audio signal having a second bandwidth wider than the first bandwidth.

Abstract: Embodiments of the present invention provide an audio signal coding and decoding method and device. The coding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

Abstract: The present invention relates to a speech processing apparatus, a speech processing method and a program which, when multichannel audio signals are downmixed and coded, prevent delay and an increase in the computation amount upon decoding of the audio signals. An inverse multiplexing unit (101) acquires coded data on which a BC parameter is multiplexed. An uncorrelated frequency-time transform unit (102) performs IMDCT transform and IMDST transform of frequency spectrum coefficients of a monaural signal (XM) obtained from this coded data to generate the monaural signal XM) which is a time domain signal and a signal (XD?) which is substantially uncorrelated with this monaural signal (XM). The stereo synthesis unit (103) generates a stereo signal by synthesizing the monaural signal (XM) and the signal (XD?) using the BC parameter. The present invention is applicable to, for example, a speech processing apparatus which decodes a downmixed and coded stereo signal.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Provided are a vector quantization device, a voice coding device, a vector quantization method, and a voice coding method which enable a reduction in the calculation amount of voice codec without deterioration of voice quality. In the vector quantization device, a first reference vector calculation unit (201) calculates a first reference vector by multiplying a target vector (x) by an auditory weighting LPC synthesis filter (H), and a second reference vector calculation unit (202) calculates a second reference vector by multiplying an element of the first reference vector by a filter having a high pass characteristic. A polarity preliminary selection unit (205) generates a polar vector by disposing a unit pulse having a positive or negative polarity, which is selected on the basis of the polarity of an element of the second reference vector, in the position of said element.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: Method, system, and computer program product for voice transformation are provided. The method includes transforming a source speech using transformation parameters, and encoding information on the transformation parameters in an output speech using steganography, wherein the source speech can be reconstructed using the output speech and the information on the transformation parameters. A method for reconstructing voice transformation is also provided including: receiving an output speech of a voice transformation system wherein the output speech is transformed speech which has encoded information on the transformation parameters using steganography; extracting the information on the transformation parameters; and carrying out an inverse transformation of the output speech to obtain an approximation of an original source speech.

Abstract: A method, medium, and apparatus encoding and/or decoding an audio signal to surround data. While encoding spatial information, which can up-mix an audio signal to a surround signal, to extension data, a length of a payload corresponding to the spatial information is encoded and a payload of the spatial information is decoded using the length of the payload. Accordingly, compatibility of the spatial information can be provided, and the spatial information can be transmitted by effectively embedding the spatial information.

Abstract: A transform coding apparatus includes an input scale factor calculating section that calculates an input scale factor having a predetermined number of scale factors associated with an input spectrum as an element, and a codebook that stores a plurality of scale factor candidates having a predetermined number of elements and outputs one scale factor candidate. The transform coding apparatus also includes an error calculating section that calculates an error on a per element basis, a weighted error calculating section that determines a weight on a per element basis and calculates a sum of products of the error and the weight to calculate a weighted error, and a searching section that searches for a scale factor candidate that minimizes the weighted error in the codebook.

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.

Abstract: An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: In accordance with an embodiment, A method of encoding an audio bitstream at an encoder includes encoding an original low band signal at the encoder by using a closed loop analysis-by-synthesis approach to obtain a coded low band signal, encoding an original high band signal at the encoder by using an open loop energy matching approach to obtain coded high band energy envelopes, comparing an energy of the coded low band signal with an energy of a corresponding original low band signal for a subframe, and generating an indication flag that indicates whether an energy envelope perceptual correction is needed for the subframe based on comparing the energy.

Abstract: An apparatus for generating a representation of a bandwidth-extended signal on the basis of an input signal representation includes a phase vocoder configured to obtain values of a spectral domain representation of a first patch of the bandwidth-extended signal on the basis of the input signal representation. The apparatus also includes a value copier configured to copy a set of values of the spectral domain representation of the first patch, which values are provided by the phase vocoder, to obtain a set of values of a spectral domain representation of a second patch, wherein the second patch is associated with higher frequencies than the first patch. The apparatus is configured to obtain the representation of the bandwidth-extended signal using the values of the spectral domain representation of the first patch and the values of the spectral domain representation of the second patch.

Abstract: A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.

Abstract: Method and decoder for processing of audio signals. The method and decoder relate to deriving a processed vector {circumflex over (d)} by applying a post-filter directly on a vector d comprising quantized MDCT domain coefficients of a time segment of an audio signal. The post-filter is configured to have a transfer function H which is a compressed version of the envelope of the vector d. A signal waveform is reconstructed by performing an inverse MDCT transform on the processed vector {circumflex over (d)}.

Abstract: A method for decoding audio frames includes producing a first frame of coded audio samples, producing at least a portion of a second frame of coded audio samples, generating audio gap filler samples based on parameters representative of a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples, and forming a sequence including the audio gap filler samples and the portion of the second frame of coded audio samples.

Abstract: Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

Abstract: A sound pickup apparatus includes: a microphone array including at least three microphones, wherein a first pair of microphones in which two of the at least three microphones are aligned on a first axis, and a second pair of microphones in which two of the at least three microphones are aligned on a second axis; a first null signal generator which outputs a first null signal based on a differential output of the first pair of microphones; a second null signal generator which outputs a second null signal based on a differential output of the second pair of microphones; and a combiner which generates a target signal based on the first null signal and the second null signal, the target signal having a directional characteristic in which the lowest sensitivity is formed in a direction to a line along which the first null surface meets the second null surface.

Abstract: An apparatus for generating bandwidth extension output data for an audio signal has a noise floor measurer, a signal energy characterizer and a processor. The audio signal has components in a first frequency band and components in a second frequency band, the bandwidth extension output data are adapted to control a synthesis of the components in the second frequency band. The noise floor measurer measures noise floor data of the second frequency band for a time portion of the audio signal. The signal energy characterizer derives energy distribution data, the energy distribution data characterizing an energy distribution in a spectrum of the time portion of the audio signal. The processor combines the noise floor data and the energy distribution data to obtain the bandwidth extension output data.

Abstract: Methods, and corresponding codec-containing devices are provided that have source coding schemes for encoding a component of an excitation. In some cases, the source coding scheme is an enumerative source coding scheme, while in other cases the source coding scheme is an arithmetic source coding scheme. In some cases, the source coding schemes are applied to encode a fixed codebook component of the excitation for a codec employing codebook excited linear prediction, for example an AMR-WB (Adaptive Multi-Rate-Wideband) speech codec.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

Abstract: A parameter calculator calculates lower resolution parametric information and interpolation information. On a decoder-side, an upmixer is used for generating the output channels. The upmixer uses high resolution parametric information generated by a parameter interpolator using the low resolution parametric information and decoder-side derived interpolation information or encoder-generated interpolation information for selecting one of a plurality of different interpolation characteristics.

Abstract: Disclosed is a spectral smoothing device with a structure whereby smoothing is performed after a nonlinear conversion has been performed for a spectrum calculated from an audio signal, and with which the amount of processing calculation is significantly reduced while maintaining excellent audio quality. With this spectral smoothing device, a sub band division unit (102) divides an input spectrum into multiple sub bands; a representative value calculation unit (103) calculates a representative value for each sub band using an arithmetic mean and a geometric mean; with respect to each representative value, a nonlinear conversion unit (104) performs a nonlinear conversion the characteristic of which is further emphasized as the value increases; and a smoothing unit (105) that smoothes the representative value which has undergone the nonlinear conversion for each sub band, at the frequency domain.

Abstract: Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.

Abstract: Provided is a method and apparatus for encoding and decoding an enhancement layer to reduce quantization error in a G.711 codec. Exponent indices of additional mantissa information of each sample are calculated based upon exponent information of each sample in a frame. A process of allocating 1 bit to each sample with a current exponent index is repeated, the exponent index starting from the maximum value while decreasing by 1 at every repetition until the total number of bits allocated to the samples is equal to the total number of available bits in the frame. And the most significant bits, as many as the number of bits allocated to each sample, are extracted from the additional mantissa information of each sample in the frame.

Abstract: In a method of perceptual transform coding of audio signals in a telecommunication system, performing the steps of determining transform coefficients representative of a time to frequency transformation of a time segmented input audio signal; determining a spectrum of perceptual sub-bands for said input audio signal based on said determined transform coefficients; determining masking thresholds for each said sub-band based on said determined spectrum; computing scale factors for each said sub-band based on said determined masking thresholds, and finally adapting said computed scale factors for each said sub-band to prevent energy loss for perceptually relevant sub-bands.

Abstract: The invention relates to a method for supporting an encoding of an audio signal, wherein at least one section of the audio signal is to be encoded with a coding model that allows the use of different coding frame lengths. In order to enable a simple selection of the respectively best suited coding frame length, it is proposed that at least one control parameter is determined based on signal characteristics of the audio signal. The control parameter is then used for limiting the options of possible coding frame lengths for the at least one section. The invention relates equally to a module 10,11 in which this method is implemented, to a device 1 and a system comprising such a module 10,11, and to a software program product including a software code for realizing the proposed method.

Abstract: Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract: There is provided an encoder capable of improving inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units (113, 114, 115) use, as reference signal candidates, a frequency coefficient (sL?(f)) in the low-band portion of a side residual signal, a frequency coefficient (mM,i(f)) in each sub-band portion of a monaural residual signal, and a frequency coefficient (mL(f)) in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the respective these candidates and a frequency coefficient (sM,i(f)) in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: The embodiments of a transcoding method, a transcoding device, and a communication apparatus are provided. The embodiment of a method includes: receiving a bit stream input from a sending end; determining an attribute of discontinuous transmission (DTX) used by a receiving end and a frame type of the input bit stream; and transcoding the input bit stream in a corresponding processing manner according to a determination result. Thereby, a corresponding transcoding operation is performed on the input bit stream according to the attribute of DTX used by the receiving end and the frame type of the input bit stream. In such a manner, input bit streams of various types can be processed, and the input bit streams can be correspondingly transcoded according to the requirements of the receiving end. Therefore, the average computational complexity and peak computational complexity can be effectively decreased without decreasing the quality of the synthesized speech.

Abstract: An apparatus and method for adaptive sub-band allocation of spectral coefficients are disclosed. The sizes of sub-bands are determined according to the distribution of spectral coefficients transformed from an input speech/audio signal to perform more elaborate quantization in units of sub-bands. Thus, quantization noise of the spectral coefficients is reduced, and sound quality in a frequency region is enhanced, thereby improving the quality of the signal.

Abstract: A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k)(0?k<Na). A band extension section extends the effective frequency band of first spectrum S1(k) to 0?k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). An extended spectrum assignment section assigns extended spectrum S1?(k)(Na?k<Nb) input to the extended frequency band from the outside. A spectral information specification section outputs information necessary to specify extended spectrum S1?(k) out of the spectrum given from the extended spectrum assignment section as a code.

Abstract: The present invention applies spectral flatness characteristic values to simplify psychoacoustic analysis of a sound signal. If the sound signal comprises a plurality of frames, the present invention calculates the energy of the sound signal in a frequency domain, calculates a plurality of spectral flatness, and decides to use a short-block or a long-block Modified Discrete Cosine Transform accordingly. If the sound signal comprises left and right channel signals, the present invention performs psychoacoustic analysis on the sound signal to count energy of the left and right channel signals in a frequency domain, counts spectral flatness of the left and right channel signals, and decides to use middle/side transform or left and right channel encoding to transform the left and right channel signals accordingly.

Abstract: The invention relates to the coding/decoding of a signal into several sub-bands, in which at least a first and a second sub-bands which are adjacent are transform coded (601, 602). In particular, in order to apply a perceptual weighting, in the transformed domain, to at least the second sub-band, the method comprises:—determining at least one frequency masking threshold (606) to be applied on the second sub-band; and normalizing said masking threshold in order to provide a spectral continuity between the above-mentioned first and second sub-bands. An advantageous application of the invention involves a perceptual weighting of the high-frequency band in the TDAC transform coding of a hierarchical encoder according to standard G.729.1.

Abstract: Provided is an adaptive sound source vector quantization device which can always perform a pitch cycle search with a resolution appropriate for any section of the pitch cycle search range of a second sub-frame when a pitch cycle search range of the second sub-frame changes in accordance with a pitch cycle of a first sub-frame. The device includes a first pitch cycle instruction unit (111), a search range calculation unit (112), and a second pitch cycle instruction unit (113). The first pitch cycle instruction unit (111) successively instructs pitch cycle search candidates in a predetermined search range having a search resolution which transits over a predetermined pitch cycle candidate for the first sub-frame. The search range calculation unit (112) calculates a predetermined range before and after the pitch cycle of the first sub-frame as the pitch cycle search range for the second sub-frame, if the predetermined range includes the predetermined pitch cycle search candidate.

Abstract: Character extraction section extracts character amounts, pertaining to a prosody of voice, from a voice signal sequentially in a time-serial manner. Difference value calculation calculates a difference value between each of the extracted character amounts and a reference value. Processing values, corresponding to the individual character amounts, are generated in accordance with the respective difference values, and a voice processing section controls the individual character amounts of the voice signal in accordance with the processing values corresponding to the character amounts and thereby generates an output signal having a prosody changed from the prosody of the voice signal.

Abstract: Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed.