50results about How to "Improve speech intelligibility" patented technology

A composite hearing aid system comprises two hearing aids (11, 31) with respective microphones (12, 32) and electronic receivers (17, 37), a microphone (42) and a transmitter (41) adapted to transmit the signal from the microphone (42) to the electronic receivers. At least one of the hearing aids (11, 31) comprises means for inverting the phase of the signal received by the electronic receivers (17, 37). When the phase of the received signal is inverted in one of the hearing aids (11, 31), a release from masking is obtained, and the perceived signal-to-noise ratio is improved. The invention provides a composite hearing aid system, a hearing aid and a method for processing audio signals.

A system improves speech intelligibility by reconstructing speech segments. The system includes a low-frequency reconstruction controller programmed to select a predetermined portion of a time domain signal. The low-frequency reconstruction controller substantially blocks signals above and below the selected predetermined portion. A harmonic generator generates low-frequency harmonics in the time domain that lie within a frequency range controlled by a background noise modeler. A gain controller adjusts the low-frequency harmonics to substantially match the signal strength to the time domain original input signal.

A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

A system improves the speech intelligibility and the speech quality of a speech segment. The system includes a dynamic controller that detects a background noise from an input by modeling a signal. A variable gain amplifier adjusts the variable gain of the amplifier in response to an output of dynamic controller. A shaping filter adjusts a speech signal by tilting portions of the speech signal of the dynamic controller.

The invention discloses an improved spectrum subtraction method based on a human ear masking effect and Bayesian estimation. The improved spectrum subtraction method comprises the steps of: (1) adopting an improved minimum control value recursive averaging algorithm to obtain noise power spectrum estimation of an original noisy speech; (2) combining the obtained noise power spectrum estimation forperforming preliminary spectrum subtraction on a noisy speech signal; (3) performing Bayesian estimation based on weighted likelihood ratio distortion measurement on the signal after preliminary spectrum subtraction, and calculating the optimal estimated amplitude spectrum of the signal; (4) calculating a subtraction parameter of secondary spectrum subtraction by utilizing the human ear masking effect; (5) performing IMCRA noise estimation again before secondary spectrum subtraction, and carrying out secondary spectrum subtraction to obtain a final enhanced speech signal; (6) and performing inverse Fourier transform on the enhanced speech signal to obtain a final enhanced speech. The improved spectrum subtraction method better guarantees the intelligibility of the speech while improving the noise elimination capability of the algorithm, thereby improving the overall effect of speech enhancement.

The invention discloses a voice enhancement method and system based on phase compensation. The method comprises steps that a to-be-processed noisy voice signal is obtained; short-time Fourier transform of the noisy voice signal is carried out to obtain the amplitude spectrum and the phase spectrum of the noisy voice signal; a phase spectrum compensation function is obtained, and a Sigmoid type function whose factor is correspondingly changed with the change of the noisy voice signal-to-noise ratio is compensated; the phase spectrum of the noisy voice signal is compensated according to the phase spectrum compensation function to obtain the compensated phase spectrum; the amplitude of a pure voice signal is obtained according to the amplitude spectrum of the noisy voice signal; the compensated phase spectrum and the amplitude of the pure voice signal are reconstructed to obtain an enhanced voice signal. The method is advantaged in that compared with a traditional phase compensation voiceenhancement method, estimation of the noise is closer to the real noise power spectrum, the noise in an audio signal can be effectively inhibited, and intelligibility of the voice signal is improvedwhile the quality of the voice signal is enhanced.

The invention generally relates to measurement of speech intelligibility in a mobile communication network component handling two-way communication between two ends, typically a near-end and a far-end. A basic idea is to simulate (Sl) two-way speech communication based on test signals adapted for speech intelligibility measurements, detect double-talk (S2) during the simulated speech communication by using a double-talk detector, and perform (S3) speech intelligibility measurements only at periods of double-talk. In this way, it is for example possible to take the effects of echo into account in the speech intelligibility measurements, while avoiding undesirable effects from non-linear processing (and possible added comfort noise) in the signal path for which speech intelligibility is measured.; Optionally, the operation of voice enhancement devices can be adjusted (S4) in response to the estimated speech intelligibility.

The present invention provides an efficient noise reduction earphone with low power consumption. The earphone comprises an earphone housing, the earphone housing is internally provided with a play horn, and the play horn is connected with a signal processing chip; the earphone further comprises a sensor arranged at the front end of the earphone housing to allow the sensor to stretch into a wearer's ear canal to collect jaw bone vibration signals at an inner ear; and the earphone further comprises a call microphone, the sensor collects the jaw bone vibration signals at an inner ear for voice activity detection and determines whether there is a person talking or not so as to guide an uplink conversation access to perform noise estimation for the signals collected by the call microphone and perform noise inhibition for the voice signals of the call microphone. The efficient noise reduction earphone with low power consumption employs the sensor to collect the jaw bone vibration signals atthe inner ear when the wearer is speaking, and does not need to perform echo elimination compared to a traditional earphone of an inner ear microphone which cannot collect music or conversation voicesignals played by a loudspeaker so that the power consumption is lower.

The invention discloses an air pressure type base frequency-adjusted electronic artificial throat, which is characterized in that: an air hood is arranged outside a throat fistula and the air pressure inside the gas hood is detected; and the base frequency of a glottal wave is controlled according to a pneumatic signal so as to change the voice tone. The air pressure type base frequency-adjusted electronic artificial throat comprises an air pressure sensing part (101), a waveform generating and processing part (102), a power amplifier circuit (103) and a miniature electricity-force conversionpart (104), wherein the pneumatic sensing part (101) is used for detecting the air pressure outside a throat fistula of a patient and generating an air pressure detection signal; the waveform generating and processing part (102) is used for generating a glottal wave signal so as to change the base frequency according to the air pressure detection signal; the power amplifier circuit (103) amplifies the glottal wave signal; and the miniature electricity-force conversion part (104) generates vibration under the driving of the glottal wave signal. The vibration is applied to the neck of the patient to generate the glottal wave signal, and the glottal wave signal is modulated by the tongue, the nasal cavity, the oral cavity, the lips and other organs of the patient to form voice outside the lips.

This invention discloses a frequency-no-masking hearing aid for ears. The novel double-ear hearing aid uses a new method to compensate the comprehension of a hearing-impaired user in noise. The novel method comprises steps of providing at least microphone audio frequency signal (18,20) by responding to the sound; and providing at least one evaluation of evaluations of object signal (26) and noise signals (30) based on at least one microphone audio frequency signal (18,20). At least one of the evaluations of the object signals (26) and the noise signals (30) is modified by the following methods of modifying the evaluation of the object signal (26) and the evaluation of the noise signal (30) which are positioned within different frequency bands; sending an evaluation of modified the object signal to the drumhead of the user of the double-ear hearing aid (10); and sending another modified evaluation of the noise signal of the drumhead of the user.

The invention relates to a burst noise processing system and a burst noise detection and suppression method and device. The burst noise detection method comprises the following steps: calculating frequency spectrum gradient corresponding to each frequency point according to amplitude spectra of the adjacent frequency points in frame-by-frame frequency spectral information of sound signals; determining gradient threshold upper and lower limits based on the frequency spectrum gradient; determining amplitude spectral gradient marks of the frequency points according to the frequency spectral gradient and the gradient threshold upper and lower limits, wherein the amplitude spectral gradient marks comprise a first gradient mark and a second gradient mark; alternatively searching out a first frequency point corresponding to the amplitude spectral gradient mark of the first gradient mark and a second frequency point corresponding to the amplitude spectral gradient mark of the second gradient mark as a group of frequency points; judging that the first frequency point and the second frequency point in the group of the frequency points are a frequency band corresponding to the burst noise if the difference between the second frequency point and the first frequency point in at least one group of the frequency points is smaller than the preset burst noise maximum frequency band. By adopting the scheme, the specific burst noise in a background environment can be effectively detected and suppressed.

A speech signal processing method is performed at a terminal device, including: obtaining a recorded signal and a to-be-output speech signal, the recorded signal including a noise signal and an echo signal; calculating a loop transfer function according to the recorded signal and the speech signal; calculating a power spectrum of the echo signal and a power spectrum of the noise signal according to the recorded signal, the speech signal, and the loop transfer function; calculating a frequency weighted coefficient according to the two power spectra of the echo signal and the noise signal; adjusting a frequency amplitude of the speech signal based on the frequency weighted coefficient; and outputting the adjusted speech signal to a speaker electrically coupled to the terminal device. As such, the frequency amplitude of the speech signal is automatically adjusted according to the relative frequency distribution of a noise signal and the speech signal.

The method makes it possible to extend functional possibilities, and to increase sound quality and the intelligibility of speech in mobile telephone apparatuses and communication systems for hearing-impaired subscribers. The mentioned technical result is achieved in that, in the method, personalized audio signals (A) for hearing-impaired users are generated on the basis of attributes thereof received from audiograms - frequency characteristics of the hearing of the hearing-impaired user stored in a database on the server of the communications network and linked to the telephone numbers of hearing-impaired users. A is processed on the server in a broadband frequency range on the basis of attributes of the hearing of the hearing-impaired user, the power of the processed audio signals is adjusted according to the attributes of the hearing-impaired user, and the adjusted personalized audio signals are transmitted from the communication server to the telephone apparatuses of the hearing-impaired users. The communications network used is a cellular network, and the telephone apparatus used is a mobile telephone apparatus (MTA). A mode which combines the function of a mobile telephone and of a hearing device is implemented.

The invention discloses an active sound insulation earmuff with a voice enhancement function. The active sound insulation earmuff comprises an earmuff body with high passive sound insulation performance. The microphone pickup microphone circuit is used for collecting external environment noise and voice signals and collecting error noise in the earmuff cavity; the microprocessor control circuit isused for extracting a voice signal and synthesizing an anti-noise signal; the audio driving and power amplifying circuit is used for carrying out reconstruction filtering and power amplification on the voice signal and the anti-noise signal; and the silencing loudspeaker is used for sending out anti-noise and outputting a voice signal. According to the invention, most of high-frequency noise is isolated by the passive sound insulation earmuff; low-frequency noise is suppressed near ears through an active noise reduction technology; meanwhile, a microphone array is arranged outside the earmuffs, an array voice enhancement algorithm is combined to extract voice signals, the voice signals are coupled into an active noise reduction system and input into earmuff cavities, more effective hearing protection is achieved, meanwhile, the voice intelligibility between wearers is improved, and normal voice communication is guaranteed.

The invention discloses a device voice noise reduction method, an electronic device and a storage medium, and the device voice noise reduction method is characterized in that at least one noise microphone is arranged near a noise source in the device, and a voice microphone is arranged outside the device. The method comprises the following steps: performing echo cancellation on an initial voice signal based on a noise signal collected by a noise microphone and the initial voice signal collected by a voice microphone to obtain an echo cancellation voice signal; and inputting the echo cancellation voice signal into a pre-trained neural network model to suppress the equipment noise in the echo cancellation voice signal. And finally, clean user voice is obtained, and the voice intelligibility is greatly improved, so that effective real-time communication between users is established.

The invention discloses a method of suppressing communication noise of baseband voice signals. At a transmitting end, the baseband voice signal band is divided into two sub bands; low pass filter processing and automatic gain control (AGC) constant amplitude processing are carried out on original voice signals, and the obtained signals are transmitted in the low frequency sub band; an original voice signal waveform envelope is extracted during an AGC process, and envelope signals are obtained and are subjected to frequency modulation and are transmitted in the high frequency sub band; during the AGC constant amplitude processing process, voiced-segment signals are subjected to amplitude compression processing and silent-segment signals are subjected to amplitude expansion processing, the signal amplitudes tend to be consistent, and the dynamic range becomes small; at a receiving end, a voice signal envelope is extracted through bandpass filtering and frequency modulation, and according to the envelope, low pass constant amplitude voice signals are subjected to waveform recovery; and the voice waveform recovery is an AGC inverse process. As the silent-segment signals are compressed and the voiced-segment signals are expanded, the silent-segment noise is thoroughly suppressed, the voiced-segment noise is covered by the voice signals, and good voice intelligibility can be obtained.

In one embodiment, a method for customising a sound processing device for an individual listener including presentation of one or more sounds to the listener directly from the sound processing device, each sound comprising a collection of two or more harmonically related tones, spectrally positioned about a frequency of interest, and having a temporal envelope consisting of a rise time, sustain time, and decay time, obtaining information from the listener, adjusting the level of the sounds, and using the adjusted levels to set up the sound processing device for the listener.

The invention provides a voice signal processing method and device, equipment and a storage medium, and belongs to the technical field of artificial intelligence. For a to-be-processed voice signal, a first power spectrum and phase information of the voice signal at each frequency point in a frequency domain are firstly obtained, then the first power spectrum is enhanced by obtaining a frequency band gain value corresponding to each frequency point, and a second power spectrum of each frequency point is obtained; and a target voice signal meeting a voice playing condition is generated according to the second power spectrum and the phase information of each frequency point. According to the processing mode, the power spectrum of each frequency point is enhanced in a targeted manner, so that the enhancement effect of the voice signal is more stable, the voice quality is effectively improved, and the voice intelligibility is enhanced; moreover, no matter whether the to-be-processed voice signal is subjected to cascade coding processing or not, the processing mode can be adopted to enhance the voice signal, and the application range is wide.