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: 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 apparatus according to an embodiment of the present invention for mixing a first frame of a first input data stream and a second frame of a second input data stream has a processing unit adapted to generate an output frame, wherein the output frame has output spectral data describing a lower part of an output spectrum up to an output cross-over frequency, and wherein the output frame further has output SBR-data describing a higher part of the output spectrum above the output cross-over frequency by way of energy-related values in an output time/frequency grid resolution.

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: A method (400, 600, 700) and apparatus (220) for enhancing the intelligibility of speech emitted into a noisy environment. After filtering (408) ambient noise with a filter (304) that simulates the physical blocking of noise by a at least a part of a voice communication device (102) a frequency dependent SNR of received voice audio relative to ambient noise is computed (424) on a perceptual (e.g. Bark) frequency scale. Formants are identified (426, 600, 700) and the SNR in bands including certain formants are modified (508, 510) with formant enhancement gain factors in order to improve intelligibility. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains factors yielding combined gains which are clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532) and used to reconstruct (532, 534) an audio signal.

Abstract: A sound signal generating method includes: generating, using a computer, a plurality of unit waveform signals by dividing the original sound signal having a periodic length of repeating similar waveforms by the length of the waveform; generating, using a computer, a repetitive waveform signal for each of the generated unit waveform signals by repeating the waveform of the unit waveform signal a given number of times; and generating, using a computer, an outputsound signal by shifting each of the repetitive waveform signals in each length with a sequence in which the unit waveform signals form the original sound signal and then superimposing on one another.

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 adaptively suppressing noise in an input signal frequency spectrum derived from overlapping input frames is provided. The system includes a psychoacoustic power computation module configured to compute a noisy signal power in psychoacoustic bands, a voice activity scoring module configured to compute a probabilistic score for a presence of a speech, and a noise estimation module configured to estimate a noise power in the psychoacoustic bands based on information of past frames, the probabilistic score, and the computed noisy signal power. The system also includes a gain computation module configured to compute a gain for each frequency, based on a probabilistic heuristic, the probabilistic score and the information on the past frames, and a gain post-processing module configured to perform a gain time smoothing, a gain frequency smoothing, and a gain regulation for the computed gain.

Abstract: An apparatus for decoding an encoded audio signal having first and second portions encoded in accordance with first and second encoding algorithms, respectively, BWE parameters for the first and second portions and a coding mode information indicating a first or a second decoding algorithm, includes first and second decoders, a BWE module and a controller. The decoders decode portions in accordance with decoding algorithms for time portions of the encoded signal to obtain decoded signals. The BWE module has a controllable crossover frequency and is configured for performing a bandwidth extension algorithm using the first decoded signal and the BWE parameters for the first portion, and for performing a bandwidth extension algorithm using the second decoded signal and the bandwidth extension parameter for the second portion. The controller controls the crossover frequency for the BWE module in accordance with the coding mode information.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: A method for modification of a cepstro-temporally smoothed gain function of a gain function resulting in a bias compensated spectral gain function is provided. The cepstro-temporal smoothing increases the quality of an enhanced output signal, as it affects only spectral outliers caused by estimation errors, while the speech characteristics are well preserved. However, due to the cepstral transform, the temporal smoothing is done in the logarithmic domain rather than the linear domain, and hence results in a certain bias. Thus, the method for a general bias compensation for a cepstro-temporal smoothing of spectral filter gain functions that is only dependent on the lower limit of the spectral filter-gain function.

Abstract: A scalable encoding device capable of reducing an encoding rate to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. An extension layer is coarsely divided into a system for processing a first channel and a system for processing a second channel. A sound source predictor for processing the first channel predicts a drive sound source signal of the first channel from a drive sound source signal of a monaural signal, and outputs the predicted drive sound source signal through a multiplier to a first CELP encoder. A sound source predictor for processing the second channel predicts the drive sound source signal of the second channel from the drive sound source signal of the monaural signal and the output from the first CELP encoder, and outputs the predicted drive sound source signal through a multiplier to a second CELP encoder.

Abstract: A method for decoding an audio signal comprises receiving a combined downmix, a combined object information, and a mix information, the combined downmix being generating using at least two downmix signals, the combined object information being made by combination of at least two sets of object information, generating a downmix processing information using the combined object information and the mix information, and processing the combined downmix using the downmix processing information. The method and an apparatus for decoding an audio signal comprising the combined downmix and the combined object information can control object gain and output in a remote conference and so on. The method and the apparatus for decoding audio signal that contains multi-object signals are fast and efficiently by reducing process time, computer resource, thereby relieving the resource requirement like the wide bandwidth by using the combined object information.

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: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: Disclosed is an adaptive sound source vector quantization device capable of reducing deviation of the quantization accuracy of the adaptive sound source vector quantization of each sub-frame when performing an adaptive sound source vector quantization in a sub-frame unit by using a greater information amount in a first sub-frame than in a second sub-frame.

Abstract: Embodiments of the invention provide a method and encoder for encoding a frame in of a communication system. The method includes calculating a first set of parameters associated with the frame, wherein said first set of parameters comprises filter bank parameters. The method further includes selecting, in a first stage, one of a plurality of encoding methods based on the first set of parameters one of modes for encoding, calculating a second set of parameters associated with the frame, selecting, in a second stage, one of the plurality of encoding methods based on the result of the first stage selection and the second set of parameters one of modes for encoding, and encoding the frame using the selected encoding excitation method from the second stage.

Abstract: A combination of techniques for modifying sound provided to headphones to simulate a surround-sound speaker environment with listener adjustments. In one embodiment, Head Related Transfer Functions (HRTFs) are grouped into multiple groups, with four types of HRTF filters or other perceptual models being used and selectable by a user. Alternately, a custom filter or perceptual model can be generated from measurements of the user's body, such as optical or acoustic measurements of the user's head, shoulders and pinna. Also, the user can select a speaker type, as well as other adjustments, such as head size and amount of wall reflections.

Abstract: Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3.

Abstract: The invention concerns a method for spectral enhancement and a device therefor. The inventive method is a method for enhancing spectral content of a signal having an incomplete spectrum including a first spectral band, the method including the following steps: at least transposing the spectral content of the first band into a second spectral band not included in the spectrum to generate a transposed spectrum signal, with spectrum limited to the second spectral band; transforming the spectrum of the transposed spectrum signal to obtain an enhancing signal; combining the incomplete spectrum signal and the enhancing signal to produce a spectrum enhanced signal. The invention is characterized in that the spectral content is subjected to a whitening step.

Abstract: A method is provided for coding a source audio signal. The method includes the following steps: coding a quantization profile of coefficients representative of at least one transform of the source audio signal, according to at least to distinct coding techniques, delivering at least two sets of data representative of a quantization profile; selecting one of the sets of data representative of a quantization profile, as a function of a predetermined selection criterion; transmitting and/or storing the set of data representative of a selected quantization profile and an indicator representative of the corresponding coding technique.

Abstract: According to one embodiment, an improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

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: Provided are, among other things, systems, methods and techniques for quantizing a joint-channel-encoded audio signal, e.g., by: identifying a target quantization unit for reduction of quantization step size based on quantization errors; determining whether the target quantization unit has been jointly sum/difference encoded with another quantization unit; if the target quantization unit has been jointly sum/difference encoded with another quantization unit, then (i) designating the sum or difference channel quantization unit as a target S/D quantization unit in based on which has a greater quantization error and (ii) re-quantizing the target S/D channel quantization using a decreased quantization step size; recalculating the quantization error for the target quantization unit; and repeating the process until a specified criterion is satisfied.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: An audio/speech sender and an audio/speech receiver and methods thereof. The audio/speech sender comprising a core encoder adapted to encode a core frequency band of an input audio/speech signal having a first sampling frequency, wherein the core frequency band comprises frequencies up to a cut-off frequency. The audio/speech sender further comprises a segmentation device adapted to perform a segmentation of the input audio/speech signal into a plurality of segments, a cut-off frequency estimator adapted to estimate a cut-off frequency for each segment and adapted to transmit information about the estimated cut-off frequency to a decoder, a low-pass filter adapted to filter each segment at said estimated cut-off frequency, and a re-sampler adapted to resample the filtered segments with a second sampling frequency that is related to said cut-off frequency in order to generate an audio/speech frame to be encoded by said core encoder.

Abstract: An iterative rate-distortion optimization algorithm for MPEG I/II Layer-3 (MP3) encoding based on the method of Lagrangian multipliers. Generally, an iterative method is performed such that a global quantization step size is determined while scale factors are fixed, and thereafter the scale factors are determined while the global quantization step size is fixed. This is repeated until a calculated rate-distortion cost is within a predetermined threshold. The methods are demonstrated to be computationally efficient and the resulting bit stream is fully standard compatible.

Abstract: Samples of a component having a frequency less than a predetermined frequency in an input audio signal that is a digital signal having a predetermined sampling frequency are written in a memory. A harmonic-overtone signal having a frequency N times a frequency of the input audio signal is generated by repeating an operation N times, where N is an integer more than one, the operation including reading one sample and thinning out (N?1) samples for every N samples from the memory within each cycle period from a first one-direction zero-crossing point to a second one-direction zero-crossing point subsequent to the first one-direction zero-crossing point, each one-direction zero-crossing point being a point at which a level of the input audio signal changes from negative to positive or a point at which the level of the input audio signal changes from positive to negative.

Abstract: A memory optimization method for a MP3 decoder. In a pipeline structure for speeding matrix calculation in Mp3 decoding, an output sequence of IMDCT calculation is altered so that matrix calculation is activated before completing the IMDCT calculation. A decoding control method allows pipeline processing in MP3 decoding, with decoding procedures for subsequent granules activated while the current granule is still being processing in the matrix calculation.

Abstract: Disclosed is an adaptive excitation vector quantization device capable of improving quantization accuracy of adaptive excitation vector quantization while suppressing increase of the calculation amount in CELP encoding which performs encoding in sub-frame units. An adaptive excitation vector generator cuts out an adaptive excitation vector of a frame length (n) from an adaptive excitation codebook. An impulse response matrix former forms a n×n impulse response matrix using impulse response matrixes of sub-frames inputted from a synthesis filter. A target vector generator adds a linear prediction residual vector of each sub-frame to form a target vector of frame length (n). An evaluation measure calculator calculates an evaluation measure of the adaptive excitation vector quantization by using the adaptive excitation vector, the impulse response matrix, and the target vector.

Abstract: A non-intrusive signal quality assessment apparatus includes a feature vector calculator that determines parameters representing frames of a signal and extracts a collection of per-frame feature vectors (?;(n)) representing structural information of the signal from the parameters. A frame selector preferably selects only frames (?\with a feature vector (?;(n)) lying within a predetermined multi-dimensional window (?). Means determine a global feature set (?) over the collection of feature vectors (?;(n)) from statistical moments of selected feature vector components ((1^,02, . . . O11). A quality predictor predicts a signal quality measure (Qj from the global feature set (?)).

Abstract: In one alternative, an audio signal is analyzed using multiple psychoacoustic criteria to identify a region of the signal in which time scaling and/or pitch shifting processing would be inaudible or minimally audible, and the signal is time scaled and/or pitch shifted within that region. In another alternative, the signal is divided into auditory events, and the signal is time scaled and/or pitch shifted within an auditory event. In a further alternative, the signal is divided into auditory events, and the auditory events are analyzed using a psychoacoustic criterion to identify those auditory events in which the time scaling and/or pitch shifting processing of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: A voice communication apparatus includes a communication portion that receives a plurality of frames including at least a first frame having first voice data and a second frame having second voice data subsequent to the first frame, the first voice data and the second voice data being encoded by a predetermined encoding system, a decoding portion that decodes the first voice data and the second voice data received by the communication portion, a buffer that retains the first voice data and the second voice data decoded by the decoding portion, a calculation portion that calculates an amplitude envelope based on the first voice data decoded by the decoding portion, and a controlling portion that judges whether or not the second voice data decoded by the decoding portion exceeds the amplitude envelope and corrects the second voice data that exceeds the amplitude envelope.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Provided are an encoding method and apparatus for efficiently encoding a sinusoidal signal whose magnitude is less than a masking value according to a psychoacoustic model, a decoding method and apparatus for decoding an encoded sinusoidal signal, and a computer-readable recording medium having recorded thereon a program for executing the encoding method/the decoding method. By using a particular code indicating that the magnitude of a first sinusoidal signal is less than a masking value according to a psychoacoustic model to encode the first sinusoidal signal, difference coding for a third sinusoidal signal of a next frame, which is connected to the first sinusoidal signal, is performed using a sinusoidal signal or sinusoidal signals selected according to a method to use the particular code, and a decoding apparatus obtains a sum with a transmitted difference using the selected sinusoidal signal(s).

Abstract: Audiovisual presentation methods, systems and apparatus for improving and enhancing the listening experience of attendees of audiovisual presentations. An exemplary audiovisual presentation system includes an audio processing and distribution unit (APDU) configured to generate and broadcast a wireless audio service containing audio of an audiovisual presentation (e.g., soundtrack and dialogue audio of a movie, in the case of a movie presentation) throughout an audiovisual presentation room or space (e.g., a movie theater, in the case of a movie presentation). The wireless audio service is received by mobile receiving devices (MRDs) having or comprising headsets, headphones or earbuds, through which MRD users listen to the audio of the audiovisual presentation provided by the wireless audio service while viewing images of the audiovisual presentation.

Abstract: A system [100] is provided that includes a first set of sensors [140] to sense a set of conditions of at least one participant in a conversation and generate raw data corresponding to the sensed set of conditions. A first aggregation engine [160] aggregates the raw data and outputs a file corresponding to the raw data. A heuristic engine [175] receives the file and compares the raw data with predetermined state data and outputs a state based on a comparison of the raw data and the predetermined state data. A feedback device [180] determines a corrective action to enhance an outcome of the conversation based on the state.

Abstract: Provided are a method and apparatus for encoding an audio signal and a method and apparatus for decoding an audio signal. The method includes performing sinusoidal analysis on an audio signal in order to extract a sinusoidal signal of a current frame, determining continuation sinusoidal signal information indicating a number of continuation sinusoidal signals of next frames, which continue from the sinusoidal signal of the current frame, by performing sinusoidal tracking on the extracted sinusoidal signal of the current frame, and encoding the determined continuation sinusoidal signal information by using different Huffman tables according to index information of the current frame, thereby allowing efficient encoding with a low bitrate.

Abstract: In one embodiment, the present invention comprises a vocoder having at least one input and at least one output, an encoder comprising a filter having at least one input operably connected to the input of the vocoder and at least one output, a decoder comprising a synthesizer having at least one input operably connected to the at least one output of the encoder, and at least one output operably connected to the at least one output of the vocoder, wherein the encoder comprises a memory and the encoder is adapted to execute instructions stored in the memory comprising classifying speech segments and encoding speech segments, and the decoder comprises a memory and the decoder is adapted to execute instructions stored in the memory comprising time-warping a residual speech signal to an expanded or compressed version of the residual speech signal.

Abstract: A method for decoding a multi-audio-object signal having audio signals of first and second types encoded therein, the multi-audio-object signal having a downmix signal and side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, the method including computing a prediction coefficient matrix C based on the level information; and up-mixing the downmix signal based on the prediction coefficients to obtain a first and/or a second up-mix audio signal approximating the audio signals of the first and second types, respectively, wherein up-mixing yields the first and/or second up-mix signals S1 and S2 from the downmix signal d according to a computation representable by ( S 1 S 2 ) = D - 1 ? { ( 1 C ) ? d + H } , with “1” denoting—depending on the number of channels of d—a scalar, or an identity matrix, and D?1 being a matrix uniquely determined by a downmix prescription according

Abstract: An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.

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: A wideband speech encoder according to one embodiment includes a lowband encoder and a highband encoder. The lowband encoder is configured to encode a lowband portion of a wideband speech signal as a set of filter parameters and an encoded excitation signal. The highband encoder is configured to calculate values for coding parameters that specify a spectral envelope and a temporal envelope of a highband portion of the wideband speech signal. The temporal envelope is based on a highband excitation signal that is derived from the encoded excitation signal. In one such example, the temporal envelope is based on a difference in levels between the highband portion and a synthesized highband signal, wherein the synthesized highband signal is generated according to the highband excitation signal and a set of highband filter parameters.

Abstract: Techniques for introducing information into a data stream first obtains the spectral values of the short-term spectrum of the audio signal. Separately, information to be introduced are combined with a spread sequence obtaining a spread information signal, whereupon a spectral representation of the spread information is generated, then weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal are then summed and afterwards processed again to obtain a processed data stream including audio information and information to be introduced.

Abstract: An apparatus and method for processing an encoded signal are discussed. The method includes: if a coding type of an audio signal is a mixed signal coding type, extracting spectral data and a linear predictive coefficient from the audio signal; generating a residual signal for the linear prediction by performing an inverse frequency conversion on the spectral data; reconstructing the audio signal by performing a linear prediction coding on the linear predictive coefficient and the residual signal; and reconstructing a high frequency region signal using an extension base signal corresponding to a partial region of the reconstructed audio signal and band extension information.

Abstract: A device relating to information processing technologies and including an encoding and decoding method configured to solve the poor decoding quality problem. The method includes: encoding each sample of an input signal to generate an encoded signal of a core layer; comparing residuals of all or a part of the samples of the input signal with encoding thresholds, where the residuals are generated by core layer encoding, and performing encoding according to comparison results to generate an encoded signal of an enhancement layer; and writing the encoded signal of the core layer and the encoded signal of the enhancement layer into a bitstream to generate an encoded signal of the input signal.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.