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; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the firs

Abstract: A signal processing device according to aspects of the present disclosures comprises a measuring section configured to measure an impulse response between each of a plurality of speakers and a predetermined listening position, a Fourier transformer configured to obtain a frequency spectrum corresponding to each of the plurality of speakers by applying a Fourier transform, a phase adjustment amount calculator configured to calculate a phase adjustment amount for each frequency, a band detector configured to detect a leading phase band, a phase converter configured to convert a phase of the leading phase band to a lagging phase, and a filter coefficient generator configured to generate a filter coefficient based on the phase adjustment amount after conversion by the phase converter.

Abstract: Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments.

Abstract: Disclosed is an encoding apparatus that can efficiently encode a signal that is a broad or extra-broad band signal or the like, thereby improving the quality of a decoded signal. This encoding apparatus includes a band establishing unit (301) that generate, based on the characteristic of the input signal, band establishment information to be used for dividing the band of the input signal to establish a first band part of lower frequency side and a second band part of higher frequency side; a lower frequency encoding unit (302) for encoding, based on the band establishment information, the input signal of the first band part to generate encoded lower frequency part information; and a higher frequency encoding unit (303) for encoding, based on the band establishment information, the input signal of the second band part to generate encoded higher frequency part information.

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 synthesis of the coded audio signal of the first transform-coded 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. Upon synthesis of the coded audio signal in the first frame, the time-domain aliasing is cancelled from the audio signal synthesis using the FAC synthesis.

Abstract: A speaker speed conversion system includes: a risk site detection unit (22) for detecting sites of risk regarding sound quality from among speech that is received as input, a frame boundary detection unit (23) for searching for a plurality of points that can serve as candidates of frame boundaries from among speech that is received as input and, of these points, supplying as a frame boundary the point that is predicted to be best from the standpoint of sound quality, and an OLA unit (25) for implementing speed conversion based on the detection results in the frame boundary detection unit (23); wherein the frame boundary detection unit (23) eliminates, from candidates of frame boundaries, sites of risk regarding sound quality that were detected in the risk site detection unit (22).

Abstract: Packet loss concealment (PLC) systems and methods are described that use time-warping to merge a concealment signal generated to replace one or more bad frames of an audio signal with a received signal representing one or more subsequent good frames of the audio signal in a manner that avoids signal discontinuity and audible artifacts resulting therefrom. Prediction-based PLC systems and methods are also described that use time-warping to conceal the loss of one or more frames containing a transition region in a manner that will not result in an audible artifact.

Abstract: An apparatus calculates a number of spectral envelopes to be derived by a spectral band replication (SBR) encoder, wherein the SBR encoder is adapted to encode an audio signal using a plurality of sample values within a predetermined number of subsequent time portions in an SBR frame extending from an initial time to a final time, the predetermined number of subsequent time portions being arranged in a time sequence given by the audio signal. The apparatus has a decision value calculator for determining a decision value, the decision value measuring a deviation in spectral energy distributions of a pair of neighboring time portions. The apparatus further has a detector for detecting a violation of a threshold by the decision value and a processor for determining a first envelope border between the pair of neighboring time portions when the violation of the threshold is detected.

Abstract: An embodiment of the present invention is a method of storing a speed contour for use in playback of at least a portion of an audio or audio-visual work including: (a) generating one or more speed contours and/or average speed contours and/or democratic speed contours for the audio or audio-visual work; (b) storing the one or more speed contours and/or average speed contours and/or democratic speed contours in a database; and (c) associating retrieval information with the one or more stored contours in the database.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: A hearing aid for improving diminished hearing caused by reduced temporal resolution includes: a speech input unit (201) which receives a speech signal from outside; a speech analysis unit (202) which detects a sound segment and a segment acoustically regarded as soundless from the speech signal received by the speech input unit and detects a consonant segment and a vowel segment within the detected sound segment; and a signal processing unit (204) which temporally increments the consonant segment detected by the speech analysis unit (204) and temporally decrements at least one of the vowel segment and the segment acoustically regarded as soundless detected by the speech analysis unit (204).

Abstract: A multi-channel audio encoder (10) encodes an N-channel audio signal. A first unit (110) generates a first encoded M-channel signal, e.g. a spatial stereo down-mix, for the N-channel signal (N>M). Down-mixers (115, 116, 117) generate first enhancement data for the signal relative to the N-channel audio signal. A second M-channel signal, such as an artistic stereo mix, is generated for the N-channel signal. A processor (123) then generates second enhancement data for the second M-channel signal relative to the first M-channel signal. A second unit (120) generates an output signal comprising the second M-channel signal, the first enhancement data and the second enhancement data. The generator (123) can dynamically select between generating the second enhancement data as absolute enhancement data or as relative enhancement data relative to the second encoded M-channel signal.

Abstract: Provided are an apparatus and method for coding and decoding a multi-object audio signal. The apparatus includes a down-mixer for down-mixing the audio signals into one down-mixed audio signal and extracting supplementary information including header information and spatial cue information for each of the audio signals, a coder for coding the down-mixed audio signal, and a supplementary information coder for generating the supplementary information as a bit stream. The header information includes identification information for each of the audio signals and channel information for the audio signals.

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: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: Provided is a parameter decoding device which performs parameter compensation process so as to suppress degradation of a main observation quality in a prediction quantization. The parameter decoding device includes amplifiers (305-1 to 305-M) which multiply inputted quantization prediction residual vectors xn?1 to xn-M by a weighting coefficient ?1 to ?M. The amplifier (306) multiplies the preceding frame decoding LSF vector yn?1 by the weighting coefficients ??1. The amplifier (307) multiplies the code vector xn+1 outputted from a codebook (301) by the weighting coefficients ?0. An adder (308) calculates the total of the vectors outputted from the amplifiers (305-1 to 305-M), the amplifier (306), and the amplifier (307). A selector switch (309) selects the vector outputted from the adder (308) if the frame erasure coding Bn of the current frame indicates that ‘the n-th frame is an erased frame’ and the frame erasure coding Bn+1 of the next frame indicates that ‘the n+1-th frame is a normal frame’.

Abstract: The method and system disclosed herein reduces total bandwidth requirement for communication in a voice over Internet protocol application. Sample [101] and convert [102] the analog input audio signal into digital signals and derive sampled frames [103]. Compute spacings of order statistics [104]. Measure the entropy for each of the sampled frames [105]. Set a threshold for entropy [106]. Mark the audio frames as active speech frames or inactive speech frames [107]. Mark an audio frame as an' inactive speech frame when the entropy is greater than the threshold, and mark the audio frame as an active speech frame when the entropy is lesser than the threshold [107]. Transmit only the active speech frames [108].

Abstract: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: Presented herein are system(s) and method(s) for frequency domain audio speed up or slow down, while maintaining pitch. An encoded audio signal is received. Frames from the encoded audio signal are retrieved. The frames of the audio signal are transformed into a frequency domain, wherein each of said frames are associated with a plurality of initial phases, and a corresponding plurality of ending phases. The initial phases of at least one of the frames are replaced with the ending phases of another frame.

Abstract: There is provided an audio encoding device capable of maintaining continuity of spectrum energy and preventing degradation of audio quality even when a spectrum of a low range of an audio signal is copied at a high range a plurality of times. The audio encoding device (100) includes: an LPC quantization unit (102) for quantizing an LPC coefficient; an LPC decoding unit (103) for decoding the quantized LPC coefficient; an inverse filter unit (104) for flattening the spectrum of the input audio signal by the inverse filter configured by using the decoding LPC coefficient; a frequency region conversion unit (105) for frequency-analyzing the flattened spectrum; a first layer encoding unit (106) for encoding the low range of the flattened spectrum to generate first layer encoded data; a first layer decoding unit (107) for decoding the first layer encoded data to generate a first layer decoded spectrum, and a second layer encoding unit (108) for encoding.

Abstract: This invention locally controls the pitch of speech and audio signals. The invention is based on a seamless time scale modification (S-TSM) scheme connected to a synchronized sampling rate converter that switches between different time scale factors in a seamless manner and controls pitch during playback in a nearly continuous way.

Abstract: An encoder (109) comprises a receiver (201) which receives a time domain audio signal. A filter bank (203) generates a first subband signal from the time domain audio signal where the first subband signal corresponds to a non-critically sampled complex subband domain representation of the time domain signal. A conversion processor (205) generates a second subband signal from the first subband signal by subband processing. The second subband signal corresponds to a critically sampled complex subband domain representation of the time domain audio signals. An encode processor (207) then generates a waveform encoded data stream by encoding data values of the second subband signal. The conversion processor (205) generates the second subband signal by direct subband conversion without converting back to the time domain. The invention allows an oversampled subband signal typically generated in parametric encoding to be waveform encoded with reduced complexity. A decoder performs the inverse operation.

Abstract: A time shift calculated during a pitch-regularizing (PR) encoding of a frame of an audio signal is used to time-shift a segment of another frame during a non-PR encoding.

Abstract: A method and computer program product for providing paraphrasing in a text-to-speech (TTS) system is provided. The method includes receiving an input text, parsing the input text, and determining a paraphrase of the input text. The method also includes synthesizing the paraphrase into synthesized speech. The method further includes selecting synthesized speech to output, which includes: assigning a score to each synthesized speech associated with each paraphrase, comparing the score of each synthesized speech associated with each paraphrase, and selecting the top-scoring synthesized speech to output. Furthermore, the method includes outputting the selected synthesized speech.

Abstract: In a waveform compressing apparatus, a trial mode selecting portion selects a trial mode having the highest compression rate from a plurality of candidate modes which have not been selected before as a trial mode for generating a residue code. A waveform data compressing portion compresses a given data amount of original waveform data according to the selected trial mode so as to generate the residue code, the data amount being determined in correspondence with the selected trial mode. A waveform data restoring portion generates a restored waveform data from the compressed data using the generated residue code. A determining portion measures an evaluation value of a quantization error contained in the restored waveform data relative to the original waveform data, and determines whether the evaluation value is equal to or smaller than a predetermined allowable value.

Abstract: A auto-recording method is disclosed for auto-recording further to user request, via generating user image and voice data, extracting feature points from the image data according to pre-defined user recognition and following by considering the user as an object of following according to extracted feature points, determining whether the image and voice data satisfy a recording reference needed to perform recording. If determined that the image and voice data satisfy the recording reference, editing the image and voice data in a pre-set edit form and generating and storing at least one of recording image and recording voice data.

Abstract: A method for energy based, non-uniform time-scale compression of audio signals includes receiving a frame of data corresponding to an input audio signal and segmenting the data into a plurality of segments. The method further includes estimating a value related to energy of the frame of data, determining a peak energy estimate for the frame, determining an energy threshold based on the peak energy estimate of the frame and comparing the value related to energy of the frame of the data with the energy threshold to control time-scale compression of the audio data.

Abstract: A method for energy based, non-uniform time-scale compression of audio signals includes receiving a frame of data corresponding to an input audio signal and segmenting the data into a plurality of segments. The method further includes estimating a value related to energy of the frame of data, determining a peak energy estimate for the frame, determining an energy threshold based on the peak energy estimate of the frame and comparing the value related to energy of the frame of the data with the energy threshold to control time-scale compression of the audio data.

Abstract: A method and system may provide an interface (e.g., “API”), client side software module or other process that may accept client input defining a playback environment, such as a speech output interface, accept client input selecting preprogrammed functionality for operating the speech playback environment, accept client input tailoring the preprogrammed functionality based on the client input, create the speech playback environment, and create embedded code to embed the speech playback environment within a website for providing speech output. A method and system may provide a website including web-site code controlling the operation of the website and plug-in code providing preprogrammed functionality for operating an embedded speech playback environment, where the plug-in code is tailored by a client, where the web-site code is to query the plug-in code for speech requests and requests for preprogrammed functionality in addition to speech functionality.

Abstract: The subject mater herein relates to voice applications and, more particularly, speech recognition application grammar modeling. Various embodiments provide systems, methods, and software to present one or more user interfaces through which to receive input to define and manipulate properties of a graphical listen element, wherein the properties identify one or more data sources. Some such embodiments further include building a speech recognition program grammar as a function of one or more graphical listen elements, wherein the grammar includes a representation of data retrieved from the one or more data sources.

Abstract: An apparatus for performing stereo adaptation for audio editing includes a stereo decorrelator configured to receive a stereo audio frame in a compressed domain. The stereo audio frame includes a first channel and a second channel. The stereo decorrelator includes a bandwidth limitation element configured to receive a user input defining a desired editing operation to be performed with respect to one of the first and second channels in the compressed domain, and to limit a bandwidth of the other of the first and second channels based on the user input.

Abstract: In an audio signal expanding/compressing apparatus adapted to expand or compress, in a time domain, a plurality of channels of audio signals by using similar waveforms, a similar-waveform length detection unit calculates similarity of the audio signal between two successive intervals for each channel, and detects a similar-waveform length of the two intervals on the basis of the similarity of each channel.

Abstract: A method for the time scaling of a sampled audio signal is presented. The method includes a first step of performing a pitch and voicing analysis of each frame of the signal in order to determine if a given frame is voiced or unvoiced and to evaluate a pitch profile for voiced frames. The results of this analysis are used to determine the length and position of analysis windows along each frame. Once an analysis window is determined, it is overlap-added to previously synthesized windows of the output signal.