Abstract: A logarithmic frequency spectrum within a predetermined time range is calculated from a speech signal. The logarithmic frequency spectrum has a frequency element at equal intervals along a logarithmic frequency axis. A logarithmic frequency spectrogram is calculated by connecting a plurality of logarithmic frequency spectrums. A value of the frequency element along a straight line on the logarithmic frequency spectrogram is voted onto a Hough plane. The Hough plane has a voted value in correspondence with a gradient of the straight line. The voted value above a threshold and the gradient corresponding to the voted value are extracted from the Hough plane. A fundamental frequency change is calculated using the voted value and the gradient extracted.

Abstract: The invention relates to the field of audio encoding. In particular, it relates to the transcoding of audio metadata between different audio coding schemes. It describes a method and a system for transcoding audio gain metadata related to dynamic range control from first gain metadata of a first audio coding scheme to second gain metadata of a second audio coding scheme, wherein the first and second audio coding schemes use coding blocks and wherein each coding block has at least one associated gain value. The method and the system select a gain value of the second gain metadata based on the gain values of the first gain metadata such that within a time interval around the time instance associated with the gain value of the second gain metadata, the minimum gain value of the first gain metadata is selected.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, an Internet Protocol Television (IPTV) system having a controller to retrieve a user profile, cause a set-top box (STB) to present an avatar having characteristics that correlate to the user profile, receive from the STB one or more responses of the user, detect from the one or more responses a change in an emotional state of the user, adapt a search for media content according to the user profile and the detected change in the emotional state of the user, adapt a portion of the characteristics of the avatar relating to emotional feedback according to the user profile and the detected change in the emotional state of the user, and cause the STB to present the adapted avatar presenting content from a media content source identified from the adapted search for media content. Other embodiments are disclosed.

Abstract: A voice activated vehicle diagnostic system provides for hands free operation of the system. Using such a system it is possible for the diagnostic system to be used safely in a test drive of the vehicle with the operator of the system not required to physically touch the system to implement one or more tests as required.

Abstract: A System and Method for Improving the Performance of Voice biometrics is provided wherein a digitized audio signal originating from at least one input client device is compressed (standards-based or proprietary) or uncompressed, the signal optionally being passed to a network which then passes the uncompressed signal to at least a voice biometrics engine and the compressed signal to a voice recorder. The signal is compressed using a compressor utilizing CELP-based technology such as MASC® technology and then sends the compressed signal optionally to a voice recorder where the signal is stored. The compressed signal is then sent to a decompressor which decompresses the signal and forwards the decompressed signal to a voice biometrics engine before being processed with or without a signal processing filter.

Abstract: A process for distributing digital audio sequences according to a nominal flux format including a succession of fields, each of which includes at least one digital block clusterizing a selected number of coefficients corresponding to single audio elements that are digitally coded inside the flux and utilized by audio decoders that are able to play it to be able to decode it correctly, including a preparatory step including modifying at least one of the coefficients, and a transmission step including a primary flux in compliance with a nominal format including blocks that were modified during the preparatory step and by a route separated from the primary flux by an additional piece of digital information which allows reconstruction of the original flux starting with a calculation, on recipient equipment, as a function of the primary flux and of the additional information.

Abstract: A voice recognition search apparatus includes: a dictionary create unit creating a first voice recognition dictionary from a search subject data; a voice acquisition unit acquiring first and second voices; a voice recognition unit creating first and second text data by recognizing the first and second voices using the first and second voice recognition dictionaries; a first search unit searching the search subject data by the first text data; and a second search unit searching a search result of the first search unit by the second text data.

Abstract: A home appliance system includes a home appliance outputting product information as a sound and a mobile terminal confirming the product information based on the sound. The mobile terminal can receive the sound, convert the sound into the product information and output the product information to an external user and a repairman.

Abstract: A voice emphasizing device emphasizes in a speech a “strained rough voice” at a position where a speaker or user of the speech intends to generate emphasis or musical expression. Thereby, the voice emphasizing device can provide the position with emphasis of anger, excitement, tension, or an animated way of speaking, or musical expression of Enka (Japanese ballad), blues, rock, or the like. As a result, rich vocal expression can be achieved. The voice emphasizing device includes: an emphasis utterance section detection unit (12) detecting, from an input speech waveform, an emphasis section that is a time duration having a waveform intended by the speaker or user to be converted; and a voice emphasizing unit (13) increasing fluctuation of an amplitude envelope of the waveform in the detected emphasis section.

Abstract: The present invention includes receiving a plurality of frame data including first frame data and second frame data encoded by at least one coding schemes, obtaining first flag information indicating whether the first frame data and the second frame data are encoded by frequency domain transform coding scheme, respectively, decoding the first frame data by frequency domain transform coding scheme based on the first flag information when the first frame data is encoded by frequency domain transform coding scheme, obtaining second flag information indicating whether subframe data is encoded by time domain transform coding scheme or time-frequency domain coding scheme when the second frame data is not encoded by frequency domain transform coding scheme, the at least two subframe data being included in the second frame data, decoding the subframe data by time domain transform coding scheme or time-frequency domain transform coding scheme based on the second flag information, and compensating for discontinuity exi

Abstract: Technologies are described herein for providing a hands-free and non-visually occluding interaction with object information. In one method, a visual capture of a portion of an object is received through a hands-free and non-visually occluding visual capture device. An audio capture is also received from a user through a hands-free and non-visually occluding audio capture device. The audio capture may include a request for information about a portion of the object in the visual capture. The information is retrieved and is transmitted to the user for playback through a hands-free and non-visually occluding audio output device.

Abstract: Low bit rate audio coding such as BWE algorithm often encounters conflict goal of achieving high time resolution and high frequency resolution at the same time. In order to achieve best possible quality, input signal can be first classified into fast signal and slow signal. This invention focuses on classifying signal into fast signal and slow signal, based on at least one of the following parameters or a combination of the following parameters: spectral sharpness, temporal sharpness, pitch correlation (pitch gain), and/or spectral envelope variation. This classification information can help to choose different BWE algorithms, different coding algorithms, and different postprocessing algorithms respectively for fast signal and slow signal.

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

Abstract: Techniques have been developed for transmitting and receiving information conveyed through the air from one portable device to another as a generally unperceivable coding within an otherwise recognizable acoustic signal. For example, in some embodiments in accordance with the present invention(s), information is acoustically communicated from a first handheld device toward a second by encoding the information in a signal that, when converted into acoustic energy at an acoustic transducer of the first handheld device, is characterized in that the acoustic energy is discernable to a human ear yet the encoding of the information therein is generally not perceivable by the human. The acoustic energy is transmitted from the acoustic transducer of the first handheld device toward the second handheld device across an air gap that constitutes a substantially entirety of the distance between the devices.

Abstract: A method for coding data, includes: grouping data into frames; classifying the frames into classes; for each class, transforming the frames belonging to the class into filter parameter vectors, which are extracted from the frames by applying a first mathematical transformation; for each class, computing a filter codebook based on the filter parameter vectors belonging to the class; segmenting each frame into subframes; for each class, transforming the subframes belonging to the class into source parameter vectors, which are extracted from the subframes by applying a second mathematical transformation based on the filter codebook computed for the corresponding class; for each class, computing a source codebook based on the source parameter vectors belonging to the class; and coding the data based on the computed filter and source codebooks.

Abstract: Described herein is a novelty item and method for creating a novelty item that delivers a personalized message at the same time that at least a first sensory action occurs. The personalized message can be audio or visual and the sensory action can be audible, physical, visual or olfactory. The invention further provides a message can be rerecorded numerous times until a satisfactory message has been recorded. The invention further provides means for disabling the recording mechanism after satisfactory vocals are recorded.

Abstract: A system for voice-enabled location and execution for playback of media content selections stored on a media content playback device has a voice input circuitry for inputting voice-based commands into the playback device; codec circuitry for converting voice input from analog content to digital content for speech recognition and for converting voice-located media content to analog content for playback; and a media content synchronization device for maintaining at least one grammar list of names representing media content selections in a current state according to what is currently stored and available for playback on the playback device.

Abstract: A voice conversion apparatus stores, in a parameter memory, target speech spectral parameters of target speech, stores, in a voice conversion rule memory, a voice conversion rule for converting voice quality of source speech into voice quality of the target speech, extracts, from an input source speech, a source speech spectral parameter of the input source speech, converts extracted source speech spectral parameter into a first conversion spectral parameter by using the voice conversion rule, selects target speech spectral parameter similar to the first conversion spectral parameter from the parameter memory, generates an aperiodic component spectral parameter representing from selected target speech spectral parameter, mixes a periodic component spectral parameter included in the first conversion spectral parameter with the aperiodic component spectral parameter, to obtain a second conversion spectral parameter, and generates a speech waveform from the second conversion spectral parameter.

Abstract: Disclosed herein are systems, methods, and computer readable-media for performing an audible human verification. The method includes determining that a human verification is needed, presenting an audible challenge to a user which exploits a known issue with automatic speech recognition processes, receiving a response to the audible challenge, and verifying that a human provided the response. The known issue with automatic speech recognition processes can be recognition of a non-word, in which case the user can be asked to spell the recognized non-word. The known issue with automatic speech recognition processes can be differentiation of simultaneous input for multiple audio streams. Multiple audio streams contained in the audible challenge can be provided monaurally. Verifying that a human provided the response can include confirming the contents of one of the multiple audio streams. Audible human verification can be performed in combination with visual human verification.

Abstract: It possible not only to reduce a delay, but also to enhance the coding efficiency and reduce audio artifact upon coding.

Abstract: Disclosed herein are systems, methods, and computer readable-media for improving name dialer performance. The method includes receiving a speech query for a name in a directory of names, retrieving matches to the query, if the matches are uniquely spelled homophones or near-homophones, identifying information that is unique to all retrieved matches, and presenting a spoken disambiguation statement to a user that incorporates the identified unique information. Identifying information can include multiple pieces of unique information if necessary to completely disambiguate the matches. A hierarchy can establish priority of multiple pieces of unique information for use in the spoken disambiguation statement.

Abstract: A multiplexing apparatus for multiplexing audio data into transport stream (TS) packets includes a first encoding section encoding the audio data by a first encoding method; a second encoding section encoding the audio data by a second encoding method, which is a variable-length encoding method and which differs from the first encoding method, for attaching a timing value indicating a timing used when audio data is decoded in units of predetermined audio data; a packetization section packetizing the audio data encoded by the first encoding section and the audio data encoded by the second encoding section into TS packets and for attaching the same ID to a plurality of packetized TS packets; a determination section determining a TS packet to be multiplexed from among the plurality of TS packets packetized by the packetization section; and a multiplexing section multiplexing the TS packet determined by the determination section.

Abstract: A system and method are disclosed for extending the bandwidth of a narrowband signal such as a speech signal. The method applies a parametric approach to bandwidth extension but does not require training. The parametric representation relates to a discrete acoustic tube model (DATM). The method comprises computing narrowband linear predictive coefficients (LPCs) from a received narrowband speech signal, computing narrowband partial correlation coefficients (parcors) using recursion, computing Mnb area coefficients from the partial correlation coefficient, and extracting Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors.

Abstract: There is disclosed a speech switching device capable of improving quality of a decoded signal. In the device, a weighted addition unit (114) outputs a mixed signal of a narrow-band speech signal and a wide-band speech signal when switching the speech signal band. A mixing unit formed by an extended layer decoded speech amplifier (122) and an adder (124) mixes the narrow-band speech signal with the wide-band speech signal while changing the mixing ratio of the narrow-band speech signal and the wide-band speech signal as the time elapses, thereby obtaining a mixed signal. An extended layer decoded speech gain controller (120) variably sets the degree of change of the mixing ratio by the time.

Abstract: Methods and apparatuses to assist a user in the performance of a plurality of tasks are provided. The method may comprise storing at least one care plan in a voice assistant, the care plan defining a plurality of tasks to be performed, capturing speech input from the user, determining, from the speech input, a selected interaction with a care plan, and in response to the selected interaction, providing a speech dialog with the user reflective of the care plan. Alternatively, the method may comprise capturing speech input from a user, determining from the speech input, a first weight associated with a resident, associating the first weight with a care plan in turn associated with the resident, comparing the first weight to a second weight associated with the resident and the care plan, and providing a speech dialog regarding reweighting the resident based on the comparison.

Abstract: A sound determination device (100) includes: an FFT unit (2402) which receives a mixed sound including a to-be-extracted sound and a noise, and obtains a frequency signal of the mixed sound for each of a plurality of times included in a predetermined duration; and a to-be-extracted sound determination unit (101 (j)) which determines, when the number of the frequency signals at the plurality of times included in the predetermined duration is equal to or larger than a first threshold value and a phase distance between the frequency signals out of the frequency signals at the plurality of times is equal to or smaller than a second threshold value, each of the frequency signals with the phase distance as a frequency signal of the to-be-extracted sound. The phase distance is a distance between phases of the frequency signals when a phase of a frequency signal at a time t is ?(t) (radian) and the phase is represented by ??(t)=mod 2?(?(t)?2?ft) (where f is an analysis-target frequency).

Abstract: A first matrix (W(k)) indicating frequency characteristics of a separation filter is calculated from input signals of a plurality of channels. A second matrix (Ws(k)) is calculated by using the restriction coefficients (Ci(k)) for restricting the separation filter and the first matrix, and separation filter coefficients (wsij(s)) are calculated by using the second matrix. With use of the separation filter coefficients, separation signals (ysi(t)) are then calculated from the input signals. A third matrix (Ws?1(k)) is then calculated by transforming the second matrix into an inverse matrix at each frequency, and reproduction filter coefficients (a?I1(s), a?I2(s)) are calculated by using the third matrix. With use of the reproduction filter coefficients, the synthesized signal of each channel is calculated by using the separation signals.

Abstract: In a gaming system, a user controls actions of characters in the game environment using speech commands. In a learning mode, available speech commands are displayed in a command menu on a display device. In a non-learning mode, the available speech commands are not displayed. A speaker-independent context-sensitive speech recognition module contains a vocabulary of available speech commands. Use of speech commands is combined with input from a controller device to control actions of a character or characters in the game environment.

Abstract: A difference signal calculating unit of a noise detecting device calculates a difference between the amplitudes of a residual signal at each sample timing and a residual signal at the preceding sample timing. A difference signal comparing unit determines whether or not an impulsive noise is present on the basis of the difference signal at the current sample timing, and the difference signal at each sample timing within a predetermined duration from the current sample timing.

Abstract: An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

Abstract: Character extraction section extracts character amounts, pertaining to a prosody of voice, from a voice signal sequentially in a time-serial manner. Difference value calculation calculates a difference value between each of the extracted character amounts and a reference value. Processing values, corresponding to the individual character amounts, are generated in accordance with the respective difference values, and a voice processing section controls the individual character amounts of the voice signal in accordance with the processing values corresponding to the character amounts and thereby generates an output signal having a prosody changed from the prosody of the voice signal.

Abstract: An audio encoding apparatus includes: a sub-band division part dividing a quantized value into sub-bands; an integrated codeword length table including a plurality of codeword length tables storing codeword lengths of individual code books and a plurality of codeword length tables; a code book selection part selecting, from the plurality of code books, the given combination of code books; a codeword length table selection part selecting a codeword length table; an index value calculation part sequentially calculating an index value to be used by the code book; a codeword length calculation part collectively obtaining codeword lengths for each code book; and a coding part determining a code book of a codeword length of a minimum accumulated result and encoding the quantized values in the sub-band based on the determined code book.

Abstract: Methods and apparatus are disclosed for a technician to access a systems interface to back-end legacy systems by voice input commands to a speech recognition module. Generally, a user logs a computer into a systems interface which permits access to back-end legacy systems. Preferably, the systems interface includes a first server with middleware for managing the protocol interface. Preferably, the systems interface includes a second server for receiving requests and generating legacy transactions. After the computer is logged-on, a request for voice input is made. A speech recognition module is launched or otherwise activated. The user inputs voice commands that are processed to convert them to commands and text that can be recognized by the client software. The client software formats the requests and forwards them to the systems interface in order to retrieve the requested information.

Abstract: A quality high frequency signal is generated through simple processing, and practical high frequency signal interpolation is carried out. A digital audio signal reproduced by an apparatus, which carries out compression, is provided to an input terminal 1 as an original signal. This original signal is sent to a peak value detection and holding circuit 2, which detects and holds a peak value and then generates a square wave signal; wherein a higher harmonic component is included in the square wave signal. This higher harmonic component is extracted by a high pass filter (HPF) 3. On the other hand, the original signal from the input terminal 1 is sent to a delay circuit 4, which delays it for a time equivalent to the processing time of the peak value detection and holding circuit 2 described above. The resulting delayed, aligned signal is sent to a low pass filter (LPF) 5, which then generates a high frequency component-removed signal.

Abstract: Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. The device includes: a first position identification unit (201) which uses a first layer error conversion coefficient indicating an error of decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band; a second position identification unit (202) which searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identification unit (201) and generates second position information indicating the identified target frequency band; and an encoding unit (203) which encodes a first layer decoding error conversion coefficient contained in the target frequency band.

Abstract: Disclosed is a method for constructing a cross-linked system whose topology of components creates a network, especially a method for creating predetermined functional units, such as cell types and tissues as well as biological and/or physical components that are based thereupon, by developing the cross-linking of the system in a self-organizing manner. The inventive method is characterized by the following steps: a) the network is represented by graph; b) edges of the said graph are provided with markings which are formed such that the graph can be unambiguously assigned to a minimal automaton; c) the automaton is described by a formal grammar representing a system of equations whose solution are defined in text form. The approach to obtain the solutions of the system of equations describes a way to construct the system, while transducers insert the components into the network in order to entirely construct the system.

Abstract: For postprocessing spectral values which are based on a first transformation algorithm for converting the audio signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided.

Abstract: Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality.

Abstract: Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit (103) performs LPC analysis on an L channel signal and an R channel signal so as to obtain an L channel LPC coefficient and an R channel LPC coefficient. An LPC coefficient adaptive filter (105) obtains an LPC coefficient adaptive filter parameter to minimize the mean square error between the L channel LPC coefficient and the R channel LPC coefficient. An LPC coefficient reconfiguration unit (106) reconfigures the R channel LPC coefficient by using the L channel LPC coefficient and the LPC coefficient adaptive filter parameter. A route calculation unit (107) calculates a polynomial route indicating the safety of the R channel reconfigured LPC coefficient. A selection unit (108) selects and outputs the LPC coefficient adaptive filter parameter or the R channel LPC coefficient according to the safety of the R channel reconfigured LPC coefficient.

Abstract: To provide a noise suppressing method and apparatus capable of achieving high-quality noise suppression using a lower amount of operations. Noise contained in an input signal is suppressed by transforming the input signal into frequency-domain signals; integrating bands of the frequency-domain signals to determine integrated frequency-domain signals; determining estimated noise based on the integrated frequency-domain signals; determining spectral gains based on the estimated noise and said integrated frequency-domain signals; and weighting said frequency-domain signals by the spectral gains.

Abstract: An audio output apparatus includes an audio codec receiving a digital audio signal from a system controller and outputting an analog audio signal corresponding to the digital audio signal to an amplifier through a capacitor, and a switch controller triggered by a trigger signal to output a control signal to the amplifier. The amplifier is disabled to cease outputting the analog audio signal in response to the control signal. The trigger signal is generated by one of the system controller, the audio codec, and a power circuit supplying electric power to the system controller, the audio codec, the switch controller and the amplifier upon occurrence of a condition associated with pop noise, and is outputted to the switch controller before the pop noise is generated, such that the amplifier is disabled in response to the control signal from the switch controller and is unable to output the pop noise.

Abstract: The invention relates to a digital signal processing technique that changes the length of an audio signal and, thus, effectively its play-out speed. This is used for frame rate conversion or sound effects in music production. Time scaling may further be used for fast forward or slow-motion audio play-out. According said method the waveform similarity overlap add approach is modified such that a maximized similarity is determined among similarity measures of sub-sequence pairs each comprising a sub-sequence to-be-matched from a input window and a matching sub-sequence from a search window wherein said sub-sequence pairs comprise at least two sub-sequence pairs of which a first pair comprises a first sub-sequence to-be-matched and a second pair comprises a different second sub-sequence to-be-matched. The input window allows for finding sub-sequence pairs with higher similarity than with a WSOLA approach based on a single sub-sequence to-be-matched. This results in less perceivable artefacts.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: [Problems] To convert a signal of non-audible murmur obtained through an in-vivo conduction microphone into a signal of a speech that is recognizable for (hardly misrecognized by) a receiving person with maximum accuracy.

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 first and second audio signals, which are object-encoded; generating third object energy information based on first object energy information included in the first audio signal and second object energy information included in the second audio signal; and generating a third audio signal by combining the first and second object signals and the third object energy information.

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: A control system based on image or voice identification includes a display unit, an order-catching device, a control device, a database and an acting device. A control method thereof includes steps of: showing a plurality of selection codes by the display unit for a user to observe and give an order according to one of the selection codes; catching images or voices of the order through the order-catching device and delivering the caught order to the control device; interpreting the order to obtain an action code, searching the database, and identifying one of the selection codes identical to the action code by the control device; and transmitting a control signal from the control device to the acting device, with the control signal being corresponding to the selection code.

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: Disclosed is a sound decoding device capable of improving the lost frame compensation performance and improving quality of the decoded sound. In this device, a rise frame sound source compensation unit (154) generates a compensation sound source signal when the current frame is a lost frame and a rise frame. An average sound source pattern update unit (156) updates the average sound source pattern held in an average sound source pattern holding unit (157) over a plurality of frames. When a frame is lost, an LPC synthesis unit (159) performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit (158) and a decoded LPC parameter from an LPC decoding unit (152) and outputs the compensation decoded sound signal.

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.