Abstract: A decoder arrangement comprising a receiver input for parameters of frame-based coded signals and a decoder arranged to provide frames of decoded audio signals based on the parameters. The receiver input and/or the decoder is arranged to establish a time difference between the occasion when parameters of a first frame is available at the receiver input and the occasion when a decoded audio signal of the first frame is available at an output of the decoder, which time difference corresponds to at least one frame. A postfilter is connected to the output of the decoder and to the receiver input. The postfilter is arranged to provide a filtering of the frames of decoded audio signals into an output signal in response to parameters of a respective subsequent frame.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: Disclosed is an acoustic communication method that includes filtering an audio signal to attenuate a high frequency section of the audio signal; generating a residual signal which corresponds to a difference between the audio signal and the filtered signal; generating a psychoacoustic mask for the audio signal based on a predetermined psychoacoustic model; generating a psychoacoustic spectrum mask by combining the residual signal with the psychoacoustic mask; generating an acoustic communication signal by modulating digital data according to the acoustic signal spectrum mask; and combining the acoustic communication signal with the filtered signal.

Abstract: An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mask ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate.

Abstract: The invention relates to transform coding/decoding of a digital audio signal represented by a succession of frames, using windows of different lengths. For the coding within the meaning of the invention, it is sought to detect (51) a particular event, such as an attack, in a current frame (Ti); and, at least if said particular event is detected at the start of the current frame (53), a short window (54) is directly applied in order to code (56) the current frame (Ti) without applying a transition window. Thus, the coding has a reduced delay in relation to the prior art. In addition, an ad hoc processing is applied during decoding in order to compensate for the direct passage from a long window to a short window during coding.

Abstract: An SBR encoder includes a filter bank that receives an input signal, a time/frequency grid generator that controls a number of bits of various parameters, a parameter calculator that calculates various parameters, a parameter coding unit that encodes the parameters, an upper-limit number-of-bit storage unit that stores an upper limit of the number of bit of encoded data of high-frequency component finally generated in a high-pass encoding process, and a number-of-bit controller. The number-of-bit controller controls the high-pass encoding process by preferentially encoding a parameter having a large influence to sound quality and not encoding a parameter having a small influence to the sound quality relative to a plurality of parameters, so that the number of bits of the encoded data of high-frequency component finally generated in the high-pass encoding process becomes equal to or less than the upper limit to be stored in the upper-limit number-of-bit storage unit.

Abstract: A voice recognition device that recognizes a voice of an input voice signal, comprises a voice model storage unit that stores in advance a predetermined voice model having a plurality of detail levels, the plurality of detail levels being information indicating a feature property of a voice for the voice model; a detail level selection unit that selects a detail level, closest to a feature property of an input voice signal, from the detail levels of the voice model stored in the voice model storage unit; and a parameter setting unit that sets parameters for recognizing the voice of an input voice according to the detail level selected by the detail level selection unit.

Abstract: The invention relates to a method for quantifying components, wherein certain components are each determined based on a plurality of audio signals and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function to be applied to the components by testing a condition relative to an audio signal and depending on a comparison made between a psycho-acoustic masking threshold relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Abstract: The present technology provides noise reduction of an acoustic signal using a configurable classification threshold which provides a sophisticated level of control to balance the tradeoff between positional robustness and noise reduction robustness. The configurable classification threshold corresponds to a configurable spatial region, such that signals arising from sources within the configurable spatial region are preserved, and signals arising from sources outside it are rejected. In embodiments, the configurable classification threshold can be automatically and dynamically adjusted in real-time based on evaluated environmental conditions surrounding an audio device implementing the noise reduction techniques described herein.

Abstract: Methods of encoding a signal using a perceptual model are described in which a signal to mask ratio parameter within the perceptual model is tuned. The signal to mask ratio parameter is tuned based on a function of the bitrate of the part of the signal which has already been encoded and the target bitrate for the encoding process. The tuned signal to mask ratio parameter is used to compute a masking threshold for the signal which is then used to quantize the signal.

Abstract: Method and arrangement for processing of a speech quality estimate, which involve adaption of a speech quality estimate based on information related to the bandwidth of a reference signal used when determining said speech quality estimate, such that the adapted speech quality estimate is independent of the bandwidth of the reference signal. The method and arrangement enable objective speech quality measurements or assessments to be performed on a unified bandwidth scale, independent of the bandwidth of a reference signal, which allows e.g. a more relevant comparison of communication systems and/or equipment, such as e.g. codecs.

Abstract: Disclosed is a recording and reproducing apparatus comprising: an apparatus main body; and a remote controller to perform remote control of the apparatus main body, wherein the remote controller comprises: a key operating section to receive a key operation by a user; a sound information inputting section to input sound information; and a transmitting section to transmit sound data based on the sound information to the apparatus main body, and the apparatus main body comprises: a recording section to record input content data on a recording medium; a reproducing section to reproduce the content data; a receiving section to receive the sound data; a sound information recording section to record the sound data so as to be associated with a piece of the content data; and a sound information outputting section to reproduce the sound data to output the reproduced sound data.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: An apparatus for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information, which are included in a bitstream representation of an audio content, in independence on a user-specified rendering matrix, the apparatus has a distortion limiter configured to obtain a modified rendering matrix using a linear combination of a user-specified rendering matrix in a target rendering matrix in dependence on a linear combination parameter. The apparatus also has a signal processor configured to obtain the upmix signal representation on the basis of the downmix signal representation and the object-related parametric information using the modified rendering matrix. The apparatus is also configured to evaluate a bitstream element representing the linear combination parameter in order to obtain the linear combination parameter.

Abstract: A method for determining a speech quality measure of an output speech signal with respect to an input speech signal, wherein the input signal passes through a signal path of a data transmission system resulting in the output signal, includes the steps of pre-processing the output signal; determining at least one of an interruption rate of the pre-processed output signal and a measure for an intensity of musical tones present in the pre-processed output signal; and determining the speech quality measure from at least one of the interruption rate and the measure for the intensity of the musical tones.

Abstract: Disclosed is a method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and decoding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information. The mode information is information for indicating that a prescribed mode corresponds to one of at least three modes. The method includes detecting when a restricted mode change occurs and changing at least one mode when detecting a restricted mode change.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: Provided are an apparatus and method for visualizing multichannel audio signals. The apparatus includes a spatial audio decoding unit for receiving a downmix signal of a time domain, converting the downmix signal into a signal of a frequency domain to output a frequency domain downmix signal, and synthesizing a multichannel audio signal based on the spatial parameter and the downmix signal; and a multichannel visualizing unit for creating visualization information of the multichannel audio signal based on the frequency domain downmix signal and the spatial parameter.

Abstract: A voice encoding device accurately encodes a spectrum shape of a signal having a strong tonality such as a vowel. The device includes: a sub-band divider which divides a first layer error conversion coefficient to be encoded into M sub-bands so as to generate M sub-band conversion coefficients; a shape vector encoder which performs encoding on each of the M sub-band conversion coefficients so as to obtain M shape encoded information and calculates a target gain of each of the M sub-band conversion coefficients; a gain vector former which forms one gain vector by using M target gains; a gain vector encoder which encodes the gain vector so as to obtain gain encoded information; and a multiplexer which multiplexes the shape encoded information with the gain encoded information.

Abstract: The present document relates to the design of anti-aliasing and/or anti-imaging filters for resamplers using rational resampling factors. In particular, the present document relates to a method for designing such filters having a reduced number of filter coefficients or an increased perceptual performance, as well as to the filters designed using such method. A method for designing a filter (102) configured to reduce imaging and/or aliasing of an output audio signal (113) at an output sampling rate (fsout) is described. The output audio signal (113) is a resampled version of an input audio signal (110) at an input sampling rate (fsin). The ratio of the output sampling rate (fsout) and the input sampling rate (fsin) is a rational number N/M. The filter (102) operates at an upsampled sampling rate which equals N times the input sampling rate (fsin).

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

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: A multi-channel signal enhancement system reinforces signal content and improves the signal-to-noise ratio of a multi-channel signal. The system detects, tracks, and reinforces non-stationary periodic signal components of a multi-channel signal. The periodic signal components of the signal may represent vowel sounds or other voiced sounds. The system may detect, track, or attenuate quasi-stationary signal components in the multi-channel signal.

Abstract: Disclosed herein is a pitch estimation apparatus and associated methods for estimating a fundamental frequency of an audio signal from a fundamental frequency probability density function by modeling the audio signal as a weighted mixture of a plurality of tone models corresponding respectively to harmonic structures of individual fundamental frequencies, so that the fundamental frequency probability density function of the audio signal is given as a distribution of respective weights of the plurality of the tone models.

Abstract: An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, and a residual signal specifying residual level values in a second predetermined time/frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type.

Abstract: An apparatus for analyzing a sound signal is based on an ear model for deriving, for a number of inner hair cells, an estimate for a time-varying concentration of transmitter substance inside a cleft between an inner hair cell and an associated auditory nerve from the sound signal so that an estimated inner hair cell cleft contents map over time is obtained. This map is analyzed by means of a pitch analyzer to obtain a pitch line over time, the pitch line indicating a pitch of the sound signal for respective time instants. A rhythm analyzer is operative for analyzing envelopes of estimates for selected inner hair cells, the inner hair cells being selected in accordance with the pitch line, so that segmentation instants are obtained, wherein a segmentation instant indicates an end of the preceding note or a start of a succeeding note. Thus, a human-related and reliable sound signal analysis can be obtained.

Abstract: A method of providing a quality measure for an output voice signal generated to reproduce an input voice signal, the method comprising: partitioning the input and output signals into frames; for each frame of the input signal, determining a disturbance relative to each of a plurality of frames of the output signal; determining a subset of the determined disturbances comprising one disturbance for each input frame such that a sum of the disturbances in the subset set is a minimum; and using the set of disturbances to provide the measure of quality.

Abstract: A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.

Abstract: A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data.

Abstract: An acoustic communication method and device are provided that filter an audio signal to attenuate a high frequency section of the audio signal. A residual signal is generated that corresponds to a difference between the audio signal and the filtered signal. A psychoacoustic mask is generated for the audio signal based on a predetermined psychoacoustic model. A psychoacoustic spectrum mask is generated by combining the residual signal with the psychoacoustic mask, an acoustic communication signal is generating by modulating digital data according to the acoustic signal spectrum mask, the acoustic communication signal is combined with the filtered signal, and radiating, by a speaker, the combined acoustic communication signal and the filtered signal in a form of sound waves.

Abstract: Presented herein are system(s), method(s), and apparatus for reducing on-chip memory requirements for audio decoding. In one embodiment, there is presented a method for decoding encoded audio signals. The method comprises fetching a first one or more tables from an off-chip memory; loading the first one or more tables to an on-chip memory; applying a first function to the encoded audio signals using the first one or more tables; fetching a second one or more tables from an off-chip memory after applying the first function; loading the second one or more tables to an on-chip memory; and applying a second function to the encoded audio signals, using the second one or more tables.

Abstract: Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.

Abstract: A modulation device including: a modulation unit for modulating a carrier in an audible sound range by an encoded transmission signal to generate a modulated signal; a masker sound generation unit for generating a masker signal outputted as a masker sound for making the modulated signal harder to hear when transmitted with the modulated signal; and an acoustic signal generation unit for inserting the masker signal in the modulated signal to generate an acoustic 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: The invention relates to audio signal processing and speech enhancement. In accordance with one aspect, the invention combines a high-quality audio program that is a mix of speech and non-speech audio with a lower-quality copy of the speech components contained in the audio program for the purpose of generating a high-quality audio program with an increased ratio of speech to non-speech audio such as may benefit the elderly, hearing impaired or other listeners. Aspects of the invention are particularly useful for television and home theater sound, although they may be applicable to other audio and sound applications. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: A method for generating background noise and a noise processing apparatus are provided in order to improve user experience. The method includes: if an obtained signal frame is a noise frame, a high band noise encoding parameter is obtained from the noise frame; weighting and/or smoothing is performed on the high band noise encoding parameter to obtain a second high band noise encoding parameter; and a high band background noise signal is generated according to the second high band noise encoding parameter. A noise processing apparatus is also provided.

Abstract: A method is provided in one example and includes receiving data propagating in a network environment and separating the data into one or more fields. At least some of the fields are evaluated in order to identify nouns and noun phrases within the fields. The method also includes identifying selected words within the nouns and noun phrases based on a whitelist and a blacklist. The whitelist includes a plurality of designated words to be tagged and the blacklist includes a plurality of rejected words that are not to be tagged. A resultant composite is generated for the selected nouns and noun phrases that are tagged. The resultant composite is incorporated into the whitelist if the resultant composite is approved.

Abstract: A stereo encoding method and apparatus are provided, so as to reduce distortion caused by delay adjustment. The stereo encoding method includes: extracting a current interchannel delay of a stereo signal and a previous delay adjacent to the current interchannel delay; performing adjustment frame judgment according to characteristics of the current stereo signal when the current delay and the previous delay are different; and performing delay adjustment on the stereo signal by using the current interchannel delay if it is judged that a frame where the current delay occurs is an adjustment frame.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract: Provided are parametric audio encoding and decoding apparatuses and methods thereof. In the parametric audio encoding method, an audio signal is segmented into a plurality of segments. At least one sine wave is extracted from each of the segments, and the extracted sine waves are connected. It is determined whether an extracted sine wave is a birth sine wave. If the extracted sine wave is a birth sine wave, a bit stream is generated by encoding the phase of the birth sine wave on the basis of the frequency of the birth sine wave, wherein the number of bits allocated to encode the phase of the birth sine wave is adjusted according to the frequency of the birth sine wave.

Abstract: A method for embedding digital information into an audio signal, is provided. The method includes dividing the digital information into low-priority data and high-priority data; dividing the audio signal into first and second signal parts; embedding at least one echo signal into the first signal part; embedding a communication signal modulated with low-priority data, which has a spectrum according to psychoacoustic analysis of the second signal part, into the second signal part; and combining the embedded first and second signal parts.

Abstract: There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units (122, 128) for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit (124) and a gain codebook (125) select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit (126) expresses the gain of the sub-band not selected, by mutually interpolating the selected gains.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving downmix information, object information and mix information, generating and transferring multi-channel information using at least one of the downmix information, the object information and the mix information, and selectively generating and transferring either first gain information or extra multi-channel information including second gain information in accordance with a decoding mode using at least one of the object information and the mix information.

Abstract: In a method of smoothing background noise in a telecommunication speech session; receiving and decoding S1O a signal representative of a speech session, the signal comprising both a speech component and a background noise component. Subsequently, determining LPC parameters S20 and an excitation signal S30 for the received signal. Thereafter, synthesizing and outputting (S40) an output signal based on the determined LPC parameters and excitation signal. In addition, modifying S35 the determined excitation signal by reducing power and spectral fluctuations of the excitation signal to provide a smoothed output signal.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

Abstract: Components of a method and system that allow identification of music from the song or sound using only the sound of the audio being played. A system built using the method and device components disclosed processes inputs sent from a mobile phone over a telephone or data connection, though inputs might be sent through any variety of computers, communications equipment, or consumer audio devices over any of their associated audio or data networks.

Abstract: There is provided a method or a device for extending a bandwidth of a first band speech signal to generate a second band speech signal wider than the first band speech signal and including the first band speech signal. The method comprises receiving a segment of the first band speech signal having a low cut off frequency and a high cut off frequency; determining the high cut off frequency of the segment; determining whether the segment is voiced or unvoiced; if the segment is voiced, applying a first bandwidth extension function to the segment to generate a first bandwidth extension in high frequencies; if the segment is unvoiced, applying a second bandwidth extension function to the segment to generate a second bandwidth extension in the high frequencies; using the first bandwidth extension and the second bandwidth extension to extend the first band speech signal beyond the high cut off frequency.

Abstract: An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.

Abstract: A method for encoding a digital signal into a scalable bitstream comprising quantizing the digital signal, and encoding the quantized signal to form a core-layer bitstream, performing an error mapping based on the digital signal and the core-layer bitstream to remove information that has been encoded into the core-layer bitstream, resulting in an error signal, bit-plane coding the error signal based on perceptual information of the digital signal, resulting in an enhancement-layer bitstream, wherein the perceptual information of the digital signal is determined using a perceptual model, and multiplexing the core-layer bitstream and the enhancement-layer bitstream, thereby generating the scalable bitstream.

Abstract: Embodiments of the invention provides a method and device for assigning bitrates to a plurality of channels in a scalable audio encoding/truncation process. Different bitrates are assigned to different channels in the scalable audio encoding/truncation process.