Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

Abstract: An audio signal is processed to determine primary and ambient components by transforming the signal into frequency-domain vectors, and decomposing the left and right channel vectors into ambient and primary components by orthogonal projection.

Abstract: An audio signal decoder has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value, based on time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation, based on an encoded audio signal representation and using the rescaled version of the time warp contour.

Abstract: A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

Abstract: A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.

Abstract: An audio signal decoder for providing a decoded multi-channel audio signal representation on the basis of an encoded multi-channel audio signal representation has a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation.

Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

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: 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: 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: Systems (1600) and methods (1500) for frame synchronization. The methods involve: extracting bit sequences S0 and S1 from a Bit Stream (“BS”) of a Data Burst (“DB”); decoding S0 and S1 to obtain decoded bit sequences S?0 and S?1; using S?0 and S?1 to determine Bit Error Rate (“BER”) estimates (516, 518); combining the BER estimates to obtain a combined BER estimate; modifying S0 and S1 so that each includes at least one bit of BS which is not included in its current set of bits and so that it is absent of at least one of the bits in the current set of bits; iteratively repeating the decoding, using, combining and modifying steps to obtain more combined BER estimates; analyzing the combined BER estimates to identify a minimum combined BER estimate; and using the minimum combined BER estimate to determine a location of a vocoder voice frame within DB.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: An audio device and a method of operating the same are provided. The audio device includes a storage unit, a first memory and a second memory, a hardware decoder, a software decoder, a first direct memory access (DMA) block, a second DMA block, and a controller. The controller converts the audio device from an ultra low power mode in which the first PCM information is transmitted to an audio interface buffer through the first memory, the hardware decoder, and the first DMA block or a low power mode in which the second PCM information is transmitted to the audio interface buffer through the second memory, the software decoder, and the first DMA block to a normal mode in which the second PCM information is transmitted to the audio interface buffer through the second memory, the software decoder, and the second DMA block.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.

Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state 715 located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation 745 of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: A computing device identifies a first codeword in a first codebook to represent short-timescale information of frames in a time-based data item segmented at intervals and identifies a second codeword in a second codebook to represent long-timescale information of the frames. The computing device generates a third codebook based on the first codeword and the second codeword for the frames to add long-timescale information context to the short-timescale information of the frames.

Abstract: An encoding device includes, an estimation unit to estimate a decoded signal of a plurality of channels based on a down-mix signal obtained by down-mixing an input signal of the plurality of channels, similarity between the channels of the input signal, and an intensity difference between the channels of the input signal; an analysis unit to analyze a phase of the input signal and a phase of the decoded signal; a calculation unit to calculate phase information based on the phase of the input signal and the phase of the decoded signal; and a coding unit to encode the similarity between the channels of the input signal, the intensity difference between the channels of the input signal, and the phase information.

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: A method, a device and a system for voice encoding/decoding are disclosed in the present invention. The method includes: assembling an input pulse code modulation signal into one signal according to a designated time slot and assembly manner; and encoding the assembled signal according to a designated encoding manner to output an encoded voice signal. In the present invention, because a process of assembling or splitting the signal may be implemented through software, in the case that hardware in a current network does not need to be replaced, an effect of encoding/decoding voice with a 7 K spectrum may be achieved in the current network.

Abstract: A coding apparatus includes a generation unit configured to generate first coding information used for first coding of a first audio signal and second coding information used for second coding of a second audio signal, and generate third coding information used for the first coding of the second audio signal and fourth coding information used for the second coding of a third audio signal; a first coding unit configured to generate first data and second data; a second coding unit configured to generate third data and fourth data by performing the second coding on the third audio signal; and a multiplexing unit configured to generate a stream of the first audio signal and a stream of the second audio signal. The third data is decoded in place of the second data in a case where a loss or an error has occurred in the stream of the second audio signal.

Abstract: Encoding and decoding of residual signals are provided. In a method of encoding a residual signal of an audio signal, the residual signal is divided into a plurality of sections having different sizes, based on a change of the residual signal. Then, section division information representing information about the divided sections and section-by-section residual signal information representing characteristics of the sections of the residual signal are acquired. Thereafter, the residual signal is encoded based on the section division information and the section-by-section residual signal information.

Abstract: Use of a scalable audio codec to implement distributed mixing and/or sender bit rate regulation in a multipoint conference is disclosed. The scalable audio codec allows the audio signal from each endpoint to be split into one or more frequency bands and for the transform coefficients within such bands to be prioritized such that usable audio may be decoded from a subset of the entire signal. The subset may be created by omitting certain frequency bands and/or by omitting certain coefficients within the frequency bands. By providing various rules for each endpoint in a conference, the endpoint can determine the importance of its signal to the conference and can select an appropriate bit rate, thereby conserving bandwidth and/or processing power throughout the conference.

Abstract: Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal.

Abstract: A computation apparatus includes: a range calculation section for calculating a range of an input value that can give a predetermined discrete value obtained by discretizing a computation result of a nonlinear operation; and a discrete value output section for outputting, when the input value is input, the predetermined discrete value corresponding to the range in which the input value that has been input is contained.

Abstract: A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.

Abstract: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

Abstract: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: An apparatus for calculating bandwidth extension data of an audio signal in a bandwidth extension system, in which a first spectral band is encoded with a first number of bits and a second spectral band different from the first spectral band is encoded with a second number of bits, the second number of bits being smaller than the first number of bits, has a controllable bandwidth extension parameter calculator for calculating bandwidth extension parameters for the second frequency band in a frame-wise manner for a sequence of frames of the audio signal. Each frame has a controllable start time instant. The apparatus additionally includes a spectral tilt detector for detecting a spectral tilt in a time portion of the audio signal and for signaling the start time instant for the individual frames of the audio signal depending on spectral tilt.

Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.

Abstract: An uplink or downlink audio processor contains a multi band compressor that receives an input, uplink or downlink, audio signal. The multi-band compressor has a band splitter that splits the input audio signal into a number of different band signals. Each band signal is input to a respective compressor block, which is independently programmable so that its audio frequency response (a) differs from a linear response in at least two non-overlapping windows of its input signal, and (b) differs from the frequency response of another one of the compressor blocks. Other embodiments are also described and claimed.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information.

Abstract: The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal.

Abstract: A Joint Source-Channel Decoding (JSCD) apparatus, method, necessity judging method, and a receiver includes: a source coding rate change judging unit configured to judge whether a source coding rate of the current frame is the same as the previous frame; a source coding rate eligibility judging unit configured to judge whether the source coding rate of the current frame is less than a predetermined source coding rate threshold; a current frame SIR eligibility judging unit configured to judge whether an SIR of the current frame is lower than a predetermined SIR threshold, a necessity result determining unit configured to determine that a JSCD is necessary, when the source coding rate of the current frame is the same as the previous frame, the source coding rate of the current frame is less than the source coding rate threshold, and the SIR of the current frame is lower than the SIR threshold.

Abstract: Provided is an apparatus and method for encoding a voice and audio signal by expanding a modified discrete cosine transform (MDCT) based CODEC to a wideband and a super-wideband in a communication system. The apparatus for encoding a signal in a communication system, includes a converter configured to convert a time domain signal corresponding to a service to be provided to users to a frequency domain signal, a quantization and normalization unit configured to calculate and quantize gain of each subband in the converted frequency domain signal and normalize a frequency coefficient of the each subband, a search unit configured to search patch information of each subband in the converted frequency domain signal using the normalized frequency coefficient, and a packetizer configured to packetize the quantized gain and the searched patch information and encode gain information of each subband in the frequency domain signal.

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: In frame-based bit stream formats the data required for decoding a current frame are usually stored within the data section for that frame. One exception is the mp3 bit stream where data for a current frame is stored in previous frames. If the decoder did not receive the required previous frame, decoding of the current mp3 frame is skipped. The invention can be applied for such bit streams, in an archival mode, a streaming mode and a sample-exact cutting of an archival mode. In the streaming and cutting modes, new headers are established. The number of frames required for initializing the decoder status is signalized in the header, as well as a consistency check value in the streaming mode. These frames are used for decoder initialization but not for decoding samples or coefficients. For a sample-exact cutting, for the frame at which the cut shall occur, the number of samples or coefficients to be muted is also indicated in the header.

Abstract: A coding method, a decoding method, a coding-decoding (codec) method, a codec system and relevant apparatuses are disclosed. The coding method includes: obtaining an amplitude vector and a length vector corresponding to a vector to be coded; sorting elements of the amplitude vector and elements of the length vector; and obtaining a position index value according to the sorted amplitude vector and the sorted length vector. A decoding method, a codec system, and relevant apparatuses are also provided.

Abstract: The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag value according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag value is a first value; obtaining a second-domain predictive signal according to the LP processing result when the long-term flag value is a second value; converting the second-domain predictive signal into a first-domain predictive signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract: The signal processing is based on the concept of using a time-domain aliased frame as a basis for time segmentation and spectral analysis, performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall “segmented” time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract: A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.

Abstract: A hierarchical audio coding, decoding method and system are provided. The method includes dividing frequency domain coefficients of an audio signal after MDCT into a plurality of coding sub-bands, quantizing and coding amplitude envelope values of coding sub-bands; allocating bits to each coding sub-band of the core layer, quantizing and coding core layer frequency domain coefficients to obtain coded bits of core layer frequency domain coefficients; calculating the amplitude envelope value of each coding sub-band of the core layer residual signal; allocating bits to each coding sub-band of the extended layer, quantizing and coding the extended layer coding signal to obtain coded bits of the extended layer coding signal; multiplexing and packing amplitude value envelope coded bits of each coding sub-band composed by core layer and extended layer frequency domain coefficients, core layer frequency coefficients coded bits, and extended layer coding signal coded bits, then transmitting to the decoding end.