Abstract: There is provided an audio encoding device capable of effectively encoding a stereo audio even when a correlation between channels of the stereo audio is small. In the device, a monaural signal generation unit (110) generates a monaural signal by using a first channel signal and a second channel signal contained in the stereo signal. An encoding channel selection unit (120) selects one of the first channel signal and the second channel signal. An encoding unit including a monaural signal encoding unit (112), a first channel encoding unit (122), a second channel encoding unit (124), and a switching unit (126) encodes the generated monaural signal to obtain core-layer encoded data and encodes the selected channel signal to obtain extended layer encoded data corresponding to the core-layer encoded data.

Abstract: Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

Abstract: The embodiments described herein are directed to systems and methods for transmitting audio data and control segment in a single bitstream and reducing audio disturbance associated with the control segment when the bitstream is processed by an audio digital-to-analog converter. The system, according to one aspect, comprises a first audio unit, a transmitter coupled to the first audio unit, a receiver coupled to the transmitter, a second audio unit coupled to the receiver, a first processor coupled to at least one of the first audio unit and the transmitter, a second processor coupled to the second audio unit and the receiver, and an audio digital-to-analog converter connected to the second processor.

Abstract: An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoder encodes an input signal and generates first encoded information, a first layer decoder decodes the first encoded information and generates a first decoded signal, a characteristic judger analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoder changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information.

Abstract: There is disclosed an encoding device capable of appropriately adjusting the dynamic range of spectrum inserted according to the technique for replacing a spectrum of a certain band with a spectrum of another band. The device includes a spectrum modification unit (112) which modifies a first spectrum S1(k) of the band 0?k<FL in various ways to change the dynamic range so that a way of modification for obtaining an appropriate dynamic range is checked. The information concerning the modification is encoded and given to a multiplexing unit (115). By using a second spectrum S2(k) having a valid signal band 0?k.

Abstract: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating rendering information by using spatial information indicating features between multi-sources and filter information having a surround effect; and generating a surround signal by applying the rendering information to a downmix signal generated by downmixing the multi-sources, wherein a tab number of a filter used in applying the rendering information to generate the surround signal is variable.

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

Abstract: A method to audio matrix encode/decode, which encode and decode audio signals of two or more channels into an audio signal of one or more channel while preserving the direction of a sound image includes extracting pieces of sound image information from audio signals of multi channels, encoding and allocating the extracted sound image information to an inaudible frequency domain except an audible frequency domain, and adding the sound image information allocated to the inaudible frequency domain and matrix-encoded stereo signals of the audible frequency domain.

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: This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

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

Abstract: A frame is received that has the wideband audio signal. The low band audio signal is encoded to generate an encoded low band signal. The high band signal is analyzed to determine whether the high band signal is perceptually relevant to the low band signal. If the high band signal is not perceptually relevant to the low band signal, the low band signal is encoded and provided in a frame to the decoder without including parameters corresponding to characteristics of the high band signal. If the high band signal is perceptually relevant, the high band signal is encoded to generate an encoded high band signal. The resultant frame that is sent to the decoder will include a combination of the encoded low band signal and the encoded high band signal.

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing 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: The present invention provides a frequency-domain spatial audio coding framework based on the perceived spatial audio scene rather than on the channel content. In one embodiment, time-frequency spatial direction vectors are used as cues to describe the input audio scene.

Abstract: A method and device for improving coding efficiency in audio coding. From the pitch values of a pitch contour of an audio signal, a plurality of simplified pitch contour segments are generated to approximate the pitch contour, based on one or more pre-selected criteria. The contour segments can be linear or non-linear with each contour segment represented by a first end point and a second end point. If the contour segments are linear, then only the information regarding the end points, instead of the pitch values, are provided to a decoder for reconstructing the audio signal. The contour segment can have a fixed maximum length or a variable length, but the deviation between a contour segment and the pitch values in that segment is limited by a maximum value.

Abstract: An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.

Abstract: Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.

Abstract: A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

Abstract: A method for perceptual spectral decoding comprises decoding of spectral coefficients recovered from a binary flux into decoded spectral coefficients of an initial set of spectral coefficients. The initial set of spectral coefficients are spectrum filled. The spectrum filling comprises noise filling of spectral holes by setting spectral coefficients in the initial set of spectral coefficients not being decoded from the binary flux equal to elements derived from the decoded spectral coefficients. The set of reconstructed spectral coefficients of a frequency domain formed by the spectrum filling is converted into an audio signal of a time domain. A perceptual spectral decoder comprises a noise filler, operating according to the method for perceptual spectral decoding.

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

Abstract: Methods and apparatus to audio watermarking and watermark detection and extracted are described herein. An example method includes receiving a media content signal, sampling the media content signal to generate samples, storing the samples in a buffer, determining a first sequence of samples in the buffer, determining a second sequence of samples in the buffer, wherein the second sequence of samples is of substantially equal length as the first sequence of samples, calculating an average of the first sequence of samples and the second sequence of samples to generate an average sequence of samples, extracting an identifier from the average sequence of samples, and storing the identifier in a tangible memory.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: An audio encoder adapted to encode a multi-channel audio signal. The encoder comprises an encoder combination module (ECM) for generating a dominant signal part and a residual signal part being a combined representation of first and second audio signals, the dominant and residual signal parts being obtained by applying a mathematical procedure to the first and second audio signals. The mathematical procedure involves a spatial parameter comprising a description of spatial properties of the first and second audio signals. Embodiments include a plurality of interconnected encoder combination module, so that e.g. six independent 5.1 format audio signals can be encoded to a single or two dominant signal parts and a number of parameter sets and residual signal parts.

Abstract: Disclosed is an object based audio contents generating/playing apparatus. The object based audio contents generating/playing apparatus may include an object audio signal obtaining unit to obtain a plurality of object audio signals by recording a plurality of sound source signals, a recording space information obtaining unit to obtain recording space information with respect to a recording space of the plurality of sound source signals, a sound source location information obtaining unit to obtain sound location information of the plurality of sound source signals, and an encoding unit to generate object based audio contents by encoding at least one of the plurality of object audio signals, the recording space information, and the sound source location information, thereby enabling the object based audio contents to be played using at least one of a WFS scheme and a multi-channel surround scheme regardless of a reproducing environment of the audience.

Abstract: A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi-channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down-mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102).

Abstract: A frequency domain method for phase-amplitude matrixed surround decoding of 2-channel stereo recordings and soundtracks, based on spatial analysis of 2-D or 3-D directional cues in the recording and re-synthesis of these cues for reproduction on any headphone or loudspeaker playback system.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: A binaural cue coding scheme involving one or more object-based cue codes, wherein an object-based cue code directly represents a characteristic of an auditory scene corresponding to the audio channels, where the characteristic is independent of number and positions of loudspeakers used to create the auditory scene. Examples of object-based cue codes include the angle of an auditory event, the width of the auditory event, the degree of envelopment of the auditory scene, and the directionality of the auditory scene.

Abstract: A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size.

Abstract: A motion estimation method, medium, and system with fast motion estimation. The motion estimation method includes comparing a cost indicating a difference between a current block of a current image and a block of a reference image specified by a starting point with a predetermined threshold and selectively searching for the best matching block of the current block from the starting point according to the comparison result.

Abstract: The present invention relates to a method and arrangement for improving quality of a voice transmission by extracting filter coefficient parameters with respect to a voice signal in a first speech transmission rate, and using the extracted filter coefficient parameters in a second transmission rate that is equal or lower than the first transmission rate.

Abstract: A method, a device, and a system for transcoding between two embedded codecs are disclosed. The method includes: delaying a first encoded stream in input streams for integer number of frames, where the first encoded stream includes a stream of at least one extension layer in the input streams obtained after input signals are encoded by using a first codec; and using the first codec to decode other encoded streams in the input streams to obtain the first decoded signal; and performing delay aligning and adjusting to obtain an adjusted signal so as to reduce the transcoding complexity and enhance quality of the transcoded signals.

Abstract: An apparatus and method for encoding and decoding using mutual information between a high band signal and a low band signal to increase a coding efficiency in a portable terminal are provided. The apparatus includes a bandwidth extender for extracting auxiliary information relating to a characteristic of a high band signal using the high band signal and a low band signal and an encoder for encoding residual high band signal obtained by subtracting auxiliary information acquired from the low band signal from auxiliary information acquired from the high band signal.

Abstract: Lossless audio coding performs decorrelation and encodes the transformed signal. The encoded bit stream comprises de-correlation parameters and the lossless representation data of the transformed signal. However, in the case of lossy based lossless coding, the additional amount of information exceeds the base layer amount of data. Therefore the additional data cannot be packed completely into the base layer e.g. as ancillary data. The data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal. Every higher layer depends on the lower layers and can only be reasonably decoded in combination with these lower layers. According to the invention, a special combination of one-time header information with repeated header information in a block structure is used. Assignment information data identify the different layers.

Abstract: A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal.

Abstract: An encoding data processing apparatus generates a marked version of an audio signal. The marked copy is generated by embedding data representative of a payload data word into the audio signal. A code word generator that generates a water mark code word from the payload data word and reads data representing the water mark code word into a shuffle data store. A shuffle processor that generates pseudo randomly at least one address within an address space of the shuffle data store for each predetermined period and reads data representing the water mark code word out from the data store at locations identified by the randomly generated address. A data embedding processor that receives the audio signal and embeds the data representing the water mark code word read out from the shuffle data store into the audio signal for each predetermined period.

Abstract: A method of encoding a time-domain audio signal is presented. In the method, an electronic device receives the time-domain audio signal. The time-domain audio signal is transformed into a frequency-domain signal including a coefficient for each of a plurality of frequencies, which are grouped into frequency bands. For each frequency band, the energy of the band is determined, a scale factor for the band is determined based on the energy of the band, and the coefficients of the band are quantized based on the associated scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors.

Abstract: A system and methods are provided for extending the frequency bandwidth of a harmonic signal. Harmonic content of a band-limited signal is extended to frequencies outside the signal's passband by performing a non-linear transformation on the complex spectrum of the band-limited signal in the frequency domain. The non-linear transformation may be accomplishes by a linear convolution of the complex spectrum with itself. A system for extending the frequency bandwidth of a harmonic signal includes a signal processor with a forward transform module for transforming a time domain signal into the frequency domain, a non-linear transform module for performing the non-linear transformation on the complex spectrum of the harmonic signal, and a reverse transform module for transforming the extended spectrum of the harmonic signal back into the time domain.

Abstract: A coding device is provided with features in which optimum coding in a higher layer is flexibly carried out based on a coding result of a lower layer and a quality audio signal in limited circumstances is served to users. In this coding device, a basic layer coding unit codes an input signal to generate a basic layer information source code and outputs a linear prediction coefficient (LPC) and a quantum LPC, which are parameters calculated at coding, to an expanded layer control unit. A basic layer decoding unit decodes the basic layer information source code. An adding unit reverses a polarity of a basic layer decoded signal, adds the same to the input signal, and calculates a difference signal. The expanded layer control unit generates expanded layer mode information indicative of a coding mode in an expanded layer based on the LPC and the quantum LPC. An expanded layer coding unit codes the difference signal obtained from the adding unit under control of the expanded layer control unit.

Abstract: A method of encoding a low bit-rate audio signal includes quantizing and encoding a plurality of low frequency sub-bands of an audio signal in a frequency domain, generating a codebook of codevectors using sub-bands of the audio signal spectrum, detecting an envelope of another frequency sub-band of the audio signal and quantizing and losslessly-encoding the detected envelope, selecting a codevector most similar to the higher frequency sub-band spectrum from the generated codebook's codevectors and determining its codebook codevector index, and generating a bit stream.

Abstract: A bit error concealment (BEC) system and method is described herein that detects and conceals the presence of click-like artifacts in an audio signal caused by bit errors introduced during transmission of the audio signal within an audio communications system. A particular embodiment of the present invention utilizes a low-complexity design that introduces no added delay and that is particularly well-suited for applications such as Bluetooth® wireless audio devices which have low cost and low power dissipation requirements.

Abstract: An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal, which is obtained by downmixing a plurality of object signals, and object side information, extracting metadata from the object-side information and displaying an information regarding the object signals based on the metadata.