56 results about "Narrowband speech" 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 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 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 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.

Artificial bandwidth expansion devices, systems, methods and computer code products are disclosed for expanding a narrowband speech signal into an artificially expanded wideband speech signal. Embodiments of the invention can operate by forming an unshaped wideband signal based on the narrowband speech signal, such as through aliasing, and shaping the wideband signal into the artificially expanded wideband speech signal by amplifying / attenuating the unshaped wideband signal using a function generated by a neural network. Weights of the neural network can be set by a training / learning subsystem which generates genomes containing the neural network weights based on simulated environments in which a device employing the artificial bandwidth expansion is expected to operate.

An electronic communication system is set forth that includes the transmission of a narrowband speech signal corresponding to a narrowband version of speech utterances of a speaker as well as the transmission of speaker-dependent data. The speaker-dependent data may be used to correlate narrowband versions of the speech utterances of the speaker with corresponding wideband versions of the speech utterances of the speaker. Both the narrowband speech signal and the speaker-dependent data are received by a receiving party. A receiver at the receiving party uses the narrowband speech signal and the speaker-dependent data to generate a wideband speech signal corresponding to a wideband version of the speech utterances of the speaker.

The invention discloses a training method of a voice bandwidth expansion model and a voice bandwidth expansion method, wherein the voice bandwidth expansion method comprises the following steps: acquiring to-be-expanded narrowband voice; calculating an amplitude spectrum and a phase spectrum of the to-be-expanded narrowband voice, and extracting auxiliary characteristics of the to-be-expanded narrowband voice ; processing the amplitude spectrum and the auxiliary characteristics of the to-be-expanded narrowband voice by virtue of the voice bandwidth expansion model which is obtained from training, so that the amplitude spectrum of a reconstructed bandwidth voice high-frequency band can be obtained; conducting mirror image reversing on the phase spectrum of the to-be-expanded narrowband voice in a frequency domain, and determining the phase spectrum of the bandwidth voice high-frequency band; and on the basis of the amplitude spectrum and the phase spectrum of the narrowband voice, and in combination with the amplitude spectrum and the phase spectrum of the reconstructed bandwidth voice high-frequency band, determining a bandwidth voice signal. With the application of the voice bandwidth expansion method provided by the invention, an effect of improving the tone quality and naturalness of the narrowband voice can be achieved.

The invention discloses an artificial speech bandwidth expansion method and device based on K-SVD. The method is characterized by including 1, training original wideband speech signals at the bandwidth expansion transmitting end, and acquiring a wideband speech dictionary, a narrowband speech dictionary and narrowband speech sparse matrix; 2, at the bandwidth expansion receiving end, performing bandwidth expansion on original narrowband speech signals through the wideband speech dictionary and the narrowband speech dictionary, and acquiring the expanded final wideband speech signals. By the aid of the method, the wideband speech quality is improved, the training time and prior knowledge usage are reduced greatly, the accuracy of the original narrowband speech sparse matrix is improved effectively, practicality is high, and the method can be applied to the field of speech communication widely.

The present invention relates in general to the field of voice and data communications, and in particular, to a novel system for providing interoperable on-demand communications and services across otherwise incompatible narrowband voice and broadband systems. The system is also based on sponsored or universal communications services supported by interactive communications between sponsors and service providers and their respective databases of business rules.

A method for measuring voice quality in a wireless communication network includes measuring an MOS of a signal using a narrowband voice codec and an MOS of a signal using a wideband voice codec in a cable loopback environment, calculating a wideband voice codec correction coefficient using the measured MOS, measuring an MOS of a signal using the narrowband voice codec and an MOS of a signal using the wideband voice codec in a terminal connection environment; and outputting a value obtained by adding the wideband voice codec correction coefficient to the measured MOS in the terminal connection environment.

A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

The embodiments of the invention disclose a speech signal processing method and device, a terminal and a storage medium. The speech signal processing method includes the steps that a compressed narrowband speech signal is obtained; frequency domain characteristics of the narrowband speech signal are extracted; the frequency domain characteristics of the narrowband speech signal are input into a trained deep noise reduction self-encoder neural network model for nonlinear fitting, and frequency domain characteristics of a full frequency band speech signal are obtained; and the frequency domain characteristics of the full frequency band speech signal are converted into a power spectrum of the full frequency band speech signal, the Fourier inversion of the power spectrum of the full frequencyband speech signal is performed by using phase information of a corresponding narrowband signal, and the full frequency band speech signal is obtained. According to the speech signal processing methodand device, the terminal and the storage medium, the compressed narrowband speech signal is subjected to bandwidth recovery by using the deep noise reduction self-encoder neural network model, and the quality of the speech signal is improved and the intelligibility of the speech signal is increased.

A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

A method for compressing voice frequency of digital deaf-aid includes carrying out time frequency domain conversion by utilizing a windowing Fourier transform, using slope of signal to digital spectrum to judge energy distribution of voice signal and dynamically regulating voice spectrum, realizing low-pass filter by removing off high spectrum coefficient and compressing voice band by regulating spectrum envelope.

A system and method for speech signal enhancement upsamples a narrowband speech signal at a receiver to generate a wideband speech signal. The lower frequency range of the wideband speech signal is reproduced using the received narrowband speech signal. The received narrowband speech signal is analyzed to determine its formants and pitch information. The upper frequency range of the wideband speech signal is synthesized using information derived from the received narrowband speech signal.

The perceived quality of a narrowband speech signal truncated from a wideband speech signal is improved by generating in a third frequency band third speech components matching first speech components in a first frequency band of the narrowband signal, and generating in a fourth frequency band fourth speech components matching second speech components in a second frequency band of the narrowband signal. A first gain factor is applied to the third speech components to generate adjusted third speech components, and a second gain factor is applied to the fourth speech components to generate adjusted fourth speech components, the gain factors being selected such that the ratios of the average powers of the adjusted third and fourth speech components to the average power of the first speech components are predetermined values.

The disclosure relates to a communication terminal having a bandwidth expansion device for expanding the bandwidth of a narrowband voice signal, on a low-frequency and / or high-frequency side, by synthesizing at least one frequency band on the basis of the narrowband voice signal. A qualitatively satisfactory bandwidth expansion is thus performed using a plurality of net bit rates. The bandwidth expansion device is further connected to a memory containing a lookup table comprising at least one parameter value for the bandwidth expansion, for at least two net bit rates of the narrowband voice signal. A method for expanding a bandwidth of a narrowband voice signal having at least two net bit rates in a communication terminal is also disclosed herein.

An electronic communication system is set forth that includes the transmission of a narrowband speech signal corresponding to a narrowband version of speech utterances of a speaker as well as the transmission of speaker-dependent data. The speaker-dependent data may be used to correlate narrowband versions of the speech utterances of the speaker with corresponding wideband versions of the speech utterances of the speaker. Both the narrowband speech signal and the speaker-dependent data are received by a receiving party. A receiver at the receiving party uses the narrowband speech signal and the speaker-dependent data to generate a wideband speech signal corresponding to a wideband version of the speech utterances of the speaker.

The invention provides a broadband speech spectrum inclination characteristic parameter reconstruction method for speech intelligibility enhancement, comprising a training stage and a use stage of a spectrum inclination reconstruction network based on a cyclic neural network, wherein a training stage establishes a speech data set, and pretreats speech data in the data set; Input the preprocessed narrowband speech data, perform short-time Fourier transform to obtain narrowband speech spectrum, logarithmize the spectrum information to obtain logarithmic amplitude spectrum; Input the preprocessedwideband speech data, extract the parameters of all-pole model of wideband speech signal spectral inclination, and transform them into linear spectral pair parameters. The spectral tilt reconstruction network is trained and used to reconstruct the all-pole model parameters of the spectral tilt of wideband speech. The invention reconstructs the broadband speech signal spectrum inclination parameter according to the narrowband speech signal, and is suitable for all speech intelligibility enhancement systems based on the spectrum inclination characteristic, and can adapt to multilingual and multimodal speech signals.

At present, problems of small communication range and more blind zone are existent in an extensively used radio station intercom system, for the above-mentioned problem, the invention provides a multi-card Beidou transmitting / receiving radio station repeater. The device performs the voice transmission by using the Beidou multi-card switching, and the transmitted voice is converted into an ultra-short wave radio station signal to transmit and receive, and then the service range of the intercom and the radio station are extended. The device comprises a Beidou multi-card transmitting / receiving control module, a master control unit, and an ultra-short wave receiving / transmitting unit. The Beidou multi-card transmitting / receiving module improves the transmission frequency and bandwidth of the voice signal and improves the instantaneity of the voice intercom through the multi-user card switching transmission control; the master control unit performs low-code stream compression on the voice signal, and then performs the delay audio mixing processing to improve the differentiation degree of the narrow-band voice under the normal speed; and meanwhile, the invention further designs a radio station adaption circuit to finish the conversion between the Beidou voice signal and the radio station signal, the conversion comprises the PTT, COR and delay compensation.

The invention discloses a DECT base station communication method, device and system. The method comprises the following steps: determining whether a base station meets the broadband voice communication requirement of a handle or not according to a communication connection request sent by the handle, and returning communication response information; and if the communication response information ispositive acknowledge, enabling the base station and the handle to carry out broadband voice communication, and if the communication response information is negative acknowledge, enabling the base station and the handle to carry out narrowband voice communication. According to the invention, broadband voice communication between the DECT base station and more than 6 handles can be realized.

The invention discloses a voice direction-of-arrival estimation method. The method comprises the following steps of S1, splitting a broadband voice signal received by a microphone array into a plurality of narrowband voice signals; dividing the airspace into a plurality of initial airspace grids; S2, calculating a steering vector and a covariance matrix of each frequency point in each narrowband voice signal, separating a signal subspace from a noise subspace, and solving spatial spectrum energy in each grid direction; the estimated direction of arrival is judged through grid points corresponding to the spatial spectrum energy peak value; S3, through the direction of arrival obtained in the step S2, adding grid points in a grid interval in which a target possibly exists; and continuously repeating the step S2 on the reduced grid after the grid points are added to correct the direction of arrival until the upper limit of grid addition is reached. By the adoption of the voice direction-of-arrival estimation method, grid points are added in a self-adaptive mode in the estimation process, estimation precision and resolution of the voice signal direction-of-arrival method can be improved.

The invention discloses an encryption communication method and system for LTE bearing narrowband voice coding. The method comprises the steps: a terminal notifies a network side of the end-to-end encryption support capability information of the narrowband voice coding communication of the terminal in advance; the end-to-end encryption support capability information is used for indicating whether the terminal supports end-to-end encryption or not and a supported encryption mode, and the encryption mode comprises an encryption mode of migrating information bits and an encryption mode of not migrating information bits; when an end-to-end encryption communication service needs to be carried out, the network side determines an encryption mode adopted by the end-to-end encryption communication service according to the type of the end-to-end encryption communication service and the end-to-end encryption support capability information of the terminal related to the service; and the determined encryption mode is triggered and adopted to carry out the end-to-end encryption communication service. By applying the technical scheme disclosed by the invention, the QoS quality of the voice can be effectively improved.

A method for determining an upperband speech signal from a narrowband speech signal is disclosed. A list of narrowband line spectral frequencies (LSFs) is determined from the narrowband speech signal. A first pair of adjacent narrowband LSFs that have a lower difference between them than every other pair of adjacent narrowband LSFs in the list is determined. A first feature that is a mean of the first pair of adjacent narrowband LSFs is determined. Upperband LSFs are determined based on at least the first feature using codebook mapping.

The invention discloses an encrypted earphone, which comprises a sound receiving and transmitting module used for receiving and outputting voice; the key generation module is used for generating a key according to a key generation strategy; the voice encryption module is used for encrypting the audio signal; the voice decryption module is used for decrypting the encrypted audio signal; the communication transceiving module is used for transmitting and receiving voice data; the secret key generation module comprises a first secret key factor generation sub-module and is used for generating a binary code according to a password input by a user and taking the binary code as a first secret key factor; the second key factor generation submodule is used for generating a second key factor according to the multiple groups of massive random keys; and the key generation sub-module is used for performing exclusive-or operation on the key factor I and the key factor II to obtain a key. According to the invention, digital coding, encryption and modulation of a sound source are realized through combination of a narrowband voice coding technology and a general password technology, and the method is applied to an encryption earphone.

The invention relates to a broadband and narrow-band voice clarity improving device in a mobile communication field. A cut-off frequency of a wave filter is automatically set according to the sampling frequency of an input signal. Voice frame level detection is performed on an input signal through an input enveloping line detecting part and the detected level is supplied to a cut-off frequency prediction part. Voice frame level detection is performed on an output signal and the detected level is supplied to the cut-off frequency prediction part. The cut-off frequency prediction part acquires a difference between the level of an N-th voice frame which is received from the input enveloping line detecting part and the level of an (N-1)th voice frame that is received from the output enveloping line detecting part, and furthermore utilizes the difference for calculating the cut-off frequency of a consonant component of the input signal. An oblique wave filter performs filtering on the input signal according to the cut-off frequency which is calculated by the cut-off frequency prediction part, and performs selective amplifying on the component which is predicated as the consonant component of the input signal.

The invention relates to a wireless network communication technology, and particularly relates to a network system and method for achieving narrowband voice services with fixed wireless broadband access, which are used for economically achieving services by making full use of redundant bandwidths of bandwidth data on the premise of meeting key indexes of a variety of QoS demands of VOIP: throughput, time delay, time delay change (jitter) and packet loss, and so that the construction investments of telecom operators are reduced.

The invention discloses a method and a system for realizing two-way communication of a dual-mode terminal of a wide-narrow fusion cluster, and belongs to the technical field of wide-narrow fusion cluster communication, the system comprises the dual-mode terminal, a media server and a fusion control center, communication is established between the dual-mode terminal and the fusion control center, a sending end sends call requests simultaneously through a wide band and a narrow band, and the media server sends the call requests to the fusion control center. The receiving end compares the new call with the currently processed call based on the information of the newly received call request, and judges two call requests which belong to the same call and are respectively acquired through the broadband and the narrowband; based on two call requests of the same call, a broadband call connection and a narrowband call connection are respectively established, and meanwhile, a broadband voice channel and a narrowband voice channel are respectively established between the dual-mode terminal and the media server. According to the invention, based on a redundancy mechanism of a narrow-band mode and a wide-band mode, the communication stability and the service continuity are greatly improved.

The invention relates to the technical field of voice signal processing, and particularly discloses an actual communication scene-oriented voice bandwidth expansion method, which adopts a coarse-to-fine double-layer architecture, and comprises the following steps of: firstly, carrying out preliminary expansion on a narrow-band voice signal by utilizing a traditional voice bandwidth expansion technology based on a source filter model to obtain a roughly expanded voice; then constructing a lightweight deep neural network model, wherein the model comprises an encoder module, a decoder module and a lightweight packet LSTM (GLSTM) module; in the model training process, a multi-scale discriminator is used for adversarial training, and the model performance is optimized; and finally, performing refined spectrum optimization and feature enhancement on the roughly expanded speech through the optimized lightweight deep neural network model, and outputting high-quality broadband speech. According to the invention, the traditional signal processing and lightweight deep learning technologies are fused, the calculation complexity is significantly reduced, the high quality of voice expansion is ensured, and the voice signals can be processed in real time.