Abstract: An audio similarity evaluator obtains envelope signals for a plurality of frequency ranges on the basis of an input audio signal. The audio similarity evaluator is configured to obtain a modulation information associated with the envelope signals for a plurality of modulation frequency ranges, wherein the modulation information describes the modulation of the envelope signals. The audio similarity evaluator is configured to compare the obtained modulation information with a reference modulation information associated with a reference audio signal, in order to obtain an information about a similarity between the input audio signal and the reference audio signal. An audio encoder uses such an audio similarity evaluator. Another audio similarity evaluator uses a neural net trained using the audio similarity evaluator.

Abstract: Systems and methods herein provide for adaptive subband compression of power signals in a power system. In one embodiment, a system includes an encoder is operable to partition sensor measurements into frequency subbands (e.g., including an interharmonic subband), centered at integer multiples of the power system's fundamental frequency (e.g., 50 Hz or 60 Hz). The encoder may also be operable to detect active subbands, and to compress the at least one active subband. The system also includes a data concentrator operable to transmit the at least one compressed subband to a processor for analysis. The system also includes a decoder at a processing location (a substation, a concentrator, or the control center) operable to parse the compressed waveforms into subbands, to interpolate and decompress at least one compressed subband, and to synthesize the decompressed subbands as an approximation of the original waveform (e.g., subject to reconstruction error requirements).

Abstract: Vector Quantizer and method therein for vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector with four centroids C0, C1, C0,flip and C1,flip, wherein centroid C0,flip is a flipped version of centroid C0 and centroid C1,flip is a flipped version of centroid C1, each centroid representing a respective class of codevectors. A starting point for a search related to the input target vector in the codebook is determined, based on the comparison. A search is performed in the codebook, starting at the determined starting point, and a codevector is identified to represent the input target vector. A number of input target vectors per block or time segment is variable. A search space is dynamically adjusted to the number of input target vectors. The codevectors are sorted according to a distortion measure reflecting the distance between each codevector and the centroids C0 and C1.

Abstract: Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.

Abstract: A method of concealing transmission error in a digital audio signal, wherein a signal that has been decoded after transmission is received, the samples decoded while the transmitted data is valid are stored, at least one short-term prediction operator and one long-term prediction operator are estimated as a function of stored valid samples, and any missing or erroneous samples in the decoder signal are generated using the estimated operators. The energy of the synthesized signal that is thus generated is controlled by means of a gain that is computed and adapted sample by sample.

Abstract: The present invention is a synthetic speech encoding device that produces a synthetic speech signal which closely matches an actual speech signal. The actual speech signal is digitized, and excitation pulses are selected by minimizing the error between the actual and synthetic speech signals. The preferred pattern of excitation pulses needed to produce the synthetic speech signal is obtained by using an excitation pattern containing a multiplicity of weighted pulses at timed positions. The selection of the location and amplitude of each excitation pulse is obtained by minimizing an error criterion between the synthetic speech signal and the actual speech signal. The error criterion function incorporates a perceptual weighting filter which shapes the error spectrum.

Abstract: The invention relates to the code conversion of digital signals, particularly voice signals, and in particular coding according to a second format from information obtained by carrying out a coding according to a first format. These first and second formats use LPC (linear predictive coding) short-term prediction models on digital signal sample blocks while using filters represented by respective LPC coefficients. The LPC coefficients of the second format are determined from an interpolation on the representative values of the LPC coefficients of at least the first format, between at least one given block and a preceding block. According to the invention, the interpolation (43), is dynamically effected while selecting (42), for each current block, at least one interpolation factor (?) among a preselection of factors according to a predetermined criterion such as a stationarity criterion of the digital signal (41).

Abstract: Provided is a voice encoding device which performs voice encoding by a fixed code book effectively using a bit. In the voice encoding device, a position/polarity calculation unit (205) in a search loop (204) calculates a pulse position and polarity by using values of yH and HH. Moreover, a correlation value/sound source power calculation unit (206) extracts the value of the pulse position calculated by the position/polarity calculation unit (205) using yH and HH and calculates the correlation value and the sound source power. A search loop (207) successively calculates a position, polarity, a correlation value, and a sound source power of other pulses by using the pulse position and the polarity calculated by the position/polarity calculation unit (205) and the correlation value and the sound source power calculated by the correlation value/sound source power calculation unit (206).

Abstract: An apparatus and method for estimating audio signal spectrum information. The method including the steps of performing a morphological operation on a received audio signal, extracting peaks by using various peak extraction methods and extracting a remainder signal region from the extracted peaks, selecting a high-order peaks spectrum from the extracted remainder signal region. In addition, spectral envelopes are detected by performing an interpolation operation on the high-order peaks spectrum.

Abstract: An implementation of the present invention comprises a voice encoder and decoder method and system that uses voice excitation, eliminating the voice/unvoiced pitch tracking, and the first formant up to 2400 Hertz for synchronous and up to 1600 Hertz for asynchronous, does not use pulse code modulation encoding, but uses the zero crossings only of the first formant, frequency dividing by two and sampling at the formant frequency. The resulting combination uses half or less of the bit rate for excitation and the remainder for short-term spectrum analysis. The spectrum could be updated each 20 milliseconds using 49 bits for the spectrum frame and 49 bits for excitation and one frame bit for synchronous Asynchronous operation could be update at 21.25 milliseconds using 49 bits for the spectrum information and 34 bits for excitation with one bit for frame synchronization.