305 results about "Bandwidth extension" patented technology

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. The method further comprises synthesizing a wideband signal using the wideband LPCs and a wideband excitation signal, highpass filtering the synthesized wideband signal to produce a highband signal, and combining the highband signal with the original narrowband signal to generate a wideband signal. In a preferred variation of the invention, the Mnb area coefficients are converted to log-area coefficients for the purpose of extracting, through shifted-interpolation, Mwb log-area coefficients. The Mwb log-area coefficients are then converted to Mwb area coefficients before generating the wideband parcors.

A first aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy the calculated maximum energy and the energy of the block, and, for each block, a gain determining from the factor is applied to the transform coefficients of the block. Another aspect of the invention is concerned with an HF coding method for coding, through a bandwidth extension scheme, an HF signal obtained from separation of a full-bandwidth sound signal into the HF signal and a LF signal, in which an estimation of the an HF gain is calculated from LPC coefficients, the energy of the HF signal is calculated, the LF signal is processed to produce a synthesized version of the HF signal, the energy of the synthesized version of the HF signal is calculated, a ratio between the energy of the HF signal and the energy of the synthesized version of the HF signal is calculated and expressing as an HF gain, and a difference between the estimation of the HF gain and the HF gain is calculated to obtain a gain correction. A third aspect of the invention is concerned with a method for producing from a decoded target signal an overlap-add target signal in a current frame coded according to a first coding mode. According to this method, the decoded target signal of the current frame is windowed and a left portion of the window is skipped. A zero-input response of a weighting filter of the previous frame coded according to a second coding mode is calculated and windowed so that the zero-input response has an amplitude monotonically decreasing to zero after a predetermined time period. Finally, the calculated zero-input response is added to the decoded target signal to reconstruct the overlap-add target signal.

A first aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy the calculated maximum energy and the energy of the block, and, for each block, a gain determining from the factor is applied to the transform coefficients of the block. Another aspect of the invention is concerned with an HF coding method for coding, through a bandwidth extension scheme, an HF signal obtained from separation of a full-bandwidth sound signal into the HF signal and a LF signal, in which an estimation of the an HF gain is calculated from LPC coefficients, the energy of the HF signal is calculated, the LF signal is processed to produce a synthesized version of the HF signal, the energy of the synthesized version of the HF signal is calculated, a ratio between the energy of the HF signal and the energy of the synthesized version of the HF signal is calculated and expressing as an HF gain, and a difference between the estimation of the HF gain and the HF gain is calculated to obtain a gain correction. A third aspect of the invention is concerned with a method for producing from a decoded target signal an overlap-add target signal in a current frame coded according to a first coding mode. According to this method, the decoded target signal of the current frame is windowed and a left portion of the window is skipped. A zero-input response of a weighting filter of the previous frame coded according to a second coding mode is calculated and windowed so that the zero-input response has an amplitude monotonically decreasing to zero after a predetermined time period. Finally, the calculated zero-input response is added to the decoded target signal to reconstruct the overlap-add target signal.

A first aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy the calculated maximum energy and the energy of the block, and, for each block, a gain determining from the factor is applied to the transform coefficients of the block. Another aspect of the invention is concerned with an HF coding method for coding, through a bandwidth extension scheme, an HF signal obtained from separation of a full-bandwidth sound signal into the HF signal and a LF signal, in which an estimation of the an HF gain is calculated from LPC coefficients, the energy of the HF signal is calculated, the LF signal is processed to produce a synthesized version of the HF signal, the energy of the synthesized version of the HF signal is calculated, a ratio between the energy of the HF signal and the energy of the synthesized version of the HF signal is calculated and expressing as an HF gain, and a difference between the estimation of the HF gain and the HF gain is calculated to obtain a gain correction. A third aspect of the invention is concerned with a method for producing from a decoded target signal an overlap-add target signal in a current frame coded according to a first coding mode. According to this method, the decoded target signal of the current frame is windowed and a left portion of the window is skipped. A zero-input response of a weighting filter of the previous frame coded according to a second coding mode is calculated and windowed so that the zero-input response has an amplitude monotonically decreasing to zero after a predetermined time period. Finally, the calculated zero-input response is added to the decoded target signal to reconstruct the overlap-add target signal.

Expansion of the bandwidth of a wideband CMOS data amplifier is accomplished using various combinations of shunt peaking, series peaking, and miller capacitance cancellation. These various combinations are employed in any of the amplifier input stage, in intermediate stages, or in the last stage.

A novel bandwidth extension technique allows information to be encoded and decoded using a fractal self similarity model or an accurate spectral replacement model, or both. Also a multi-band temporal amplitude coding technique, useful as an enhancement to any coding / decoding technique, helps with accurate reconstruction of the temporal envelope and employs a utility filterbank. A perceptual coder using a comodulation masking release model, operating typically with more conventional perceptual coders, makes the perceptual model more accurate and hence increases the efficiency of the overall perceptual coder.

Described herein are methods to enable wireless cellular operation in unlicensed and lightly licensed, (collectively referred to as license exempt spectrum. Cognitive methods are used to enable use of unlicensed bands and / or secondary use of lightly licensed bands. Wireless devices may use licensed exempt spectrum as new bands in addition to the existing bands to transmit to a wireless transmit / receive unit (WTRU) in the downlink direction, or to a base station in the uplink direction. The wireless devices may access license exempt spectrum for bandwidth aggregation or relaying using a carrier aggregation framework. In particular, a primary component carrier operating in a licensed spectrum is used for control and connection establishment and a second component carrier operating in a licensed exempt spectrum is used for bandwidth extension.

In accordance with an embodiment, a method of decoding an encoded audio bitstream at a decoder includes receiving the audio bitstream, decoding a low band bitstream of the audio bitstream to get low band coefficients in a frequency domain, and copying a plurality of the low band coefficients to a high frequency band location to generate high band coefficients. The method further includes processing the high band coefficients to form processed high band coefficients. Processing includes modifying an energy envelope of the high band coefficients by multiplying modification gains to flatten or smooth the high band coefficients, and applying a received spectral envelope decoded from the received audio bitstream to the high band coefficients. The low band coefficients and the processed high band coefficients are then inverse-transformed to the time domain to obtain a time domain output signal.

Methods of component carrier aggregation are provided. One embodiment of the method includes aggregating, at a base station, one or more first component carriers that are backwards compatible with a first carrier type and one or more second component carriers that are non-backwards compatible with the first carrier type for communication with first user equipment according to a second carrier type used by the second component carrier(s).

The present invention relates to a solution for improving the perceived sound quality of a decoded acoustic signal. The improvement is accomplished by means of extending the spectrum of a received narrow-band acoustic signal (aNB). According to the invention, a wide-band acoustic signal (aWB) is produced by extracting at least one essential attribute (zNB) from the narrow-band acoustic signal (aNB). Parameters, e.g. representing signal energies, with respect to wide-band frequency components outside the spectrum (ANB) of the narrow-band acoustic signal (aNB) are estimated based on the at least one essential attribute (zNB). This estimation involves allocating a parameter value to a wide-band frequency component, based on a corresponding confidence level. For instance, a relatively high parameter value is allowed to be allocated to a frequency component if it has a comparatively high degree certainty. In contrast, a relatively low parameter value is only allowed to be allocated to a frequency component if it is associated with a comparatively low degree certainty.

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. The method further comprises synthesizing a wideband signal using the wideband LPCs and a wideband excitation signal, highpass filtering the synthesized wideband signal to produce a highband signal, and combining the highband signal with the original narrowband signal to generate a wideband signal. In a preferred variation of the invention, the Mnb area coefficients are converted to log-area coefficients for the purpose of extracting, through shifted-interpolation, Mwb log-area coefficients. The Mwb log-area coefficients are then converted to Mwb area coefficients before generating the wideband parcors.

The invention discloses a method and system for speech or audio signal bandwidth expansion, which comprises: A. to simulate spectral envelope of the high-frequency signal components in the speech or audio signal. B. to make a synthesis of the said spectrum envelope and the low-frequency signal components corresponding to the high-frequency signal components in the frequency and spatial domain to obtain the reset high-frequency signal components. The invention also discloses the method and system to realize the said bandwidth expansion, the technical scheme offered by which has the advantage of less bit number of coding that can be adaptively adjusted based on the type features of the signals. Besides, by extracting spectrum envelope of the high-frequency signal components, the invention makes the fine structure acted on the low-frequency signal components corresponding frequency and spatial domain to guarantee the correlation between the reset high-frequency signal spectrum and the harmonization of the high-frequency signal spectrum lopped during coding.

A common narrow-band speech signal is expanded into a wide-band speech signal. The expanded signal gives the impression of a wide-band speech signal regardless of what type of vocoder is used in a receiver. The robust techniques suggested herein are based on speech acoustics and fundamentals of human hearing. That is the techniques extend the harmonic structure of the speech signal during voiced speech segments and introduce a linearly estimated amount of speech energy in the wide frequency-band. During unvoiced speech segments, a fricated noise may be introduced in the upper frequency-band.

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

A system, computer-readable medium and generated signal are disclosed for extending the bandwidth of a first signal (i.e., a narrowband signal) such as a speech signal. The system produces a second signal from a first signal by computing first area coefficients from a first signal, generating second area coefficients from the first area coefficients and generating a second signal using the second area coefficients. The first signal may be a narrowband signal and second signal may be a wideband signal. The first area coefficients may be narrowband coefficients and the second area coefficients may be wideband area coefficients.

To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

Apparatus for processing an audio signal to generate a bandwidth extended signal having a high frequency part and a low frequency part using parametric data for the high frequency part, the parametric data relating to frequency bands of the high frequency part includes a patch border calculator for calculating a patch border such that the patch border coincides with a frequency band border of the frequency bands. The apparatus further includes a patcher for generating a patched signal using the audio signal and the patch border.

A signal bandwidth extending apparatus including: a bandwidth extending section configured to extend a frequency bandwidth of a target signal, the target signal included in an input signal; a calculating section configured to calculate a degree of the target signal included in the input signal; and a controller configured to change a method of extending the frequency bandwidth by the bandwidth extending section according to a result of the calculating section.

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.

Low-frequency bandwidth extension in the form of dynamic electrical equalization may be applied to loudspeakers so long as the excursion capability of their drive units as well as velocity limits of any port(s) or excursion limits of any associated passive radiator(s), and the power limits of the drive units are not exceeded. The bandwidth extension maximizes low-frequency bandwidth dynamically such that excursion is fully utilized over a range of drive levels, without exceeding the excursion limit. Additional limiting control is available for port air velocity or passive radiator excursion, and loudspeaker drive unit electrical power. The system applies to open back, closed box, vented box, and more complex box constructions consisting of combinations of these elements for loudspeaker designs using design parameters appropriate to each system.

A system extends the bandwidth of a narrowband speech signal into a wideband spectrum. The system includes a high-band generator that generates a high frequency spectrum based on a narrowband spectrum. A background noise generator generates a high frequency background noise spectrum based on a background noise within the narrowband spectrum. A summing circuit linked to the high-band generator and the background noise generator combines the high frequency spectrum and narrowband spectrum and the high frequency background noise spectrum.

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

A method of receiving an audio signal includes measuring a periodicity of the audio signal to determine a checked periodicity. At least one best available subband is determined. At least one extended subband is composed, wherein composing includes reducing a ratio of composed harmonic components to composed noise components if the checked periodicity is lower than a threshold, and scaling a magnitude of the at least one extended subband based on a spectral envelope on the audio signal.

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. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter fulfils a predefined criterion.

A bandwidth extension module, and an associated method and computer-readable medium, suitable for use in artificially extending the bandwidth of a lowband speech signal. The bandwidth extension module comprises a band-pass filter configured to produce a band-pass signal from the lowband speech signal; at least one carrier frequency modulator, each carrier frequency modulator configured to pitch-synchronously modulate the band-pass signal about a respective carrier frequency, the at least one carrier frequency modulator collectively producing a highband speech signal component; a synthesis filter configured to determine a highband speech signal based on the highband speech signal component; and a summation module configured to combine the lowband speech signal with the highband speech signal to obtain a bandwidth-extended speech signal.

A system extends a bandwidth of bandlimited audio signals by analyzing bandlimited audio signals at a transmission cycle rate. The analyzer may obtain a bandlimited parameter at a transmission cycle rate. A mapping device or logic in the system obtains a wideband parameter based on the bandlimited parameter. An audio signal generator generates a highband and / or lowband audio signal based on the wideband parameter at the transmission cycle rate. In some systems, the bandlimited audio signal is analyzed at the transmission cycle rate. The highband and / or lowband audio signals and the combined wideband audio signal are generated at the transmission cycle rate.

Methods and apparatus to extend the bandwidth of a speech communication to yield a perceived higher quality speech communication for an enhanced user experience. In one aspect of the invention, for example, methods and apparatus can be used to extend the bandwidth of a speech communication beyond a band-limited region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is otherwise characterized absent such bandwidth extension. In another aspect of the invention, for example, methods and apparatus can be used to substitute for corrupt, missing or lost components of a given speech communication, or to otherwise enhance the perceived quality of a speech communication, by extending the speech communication to include one or more artificially created points within the region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is characterized. The result is a speech communication that is perceived to be of higher quality. The various aspects of the present invention can be applied, for example, to network devices or to end-terminal devices.