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137 results about "Speech clarity" patented technology

Speech clarity. Speech clarity concerns the quality of speech transfer to the listeners. In a reverberant room with disturbing background noise, it can be difficult to pick up speech.

Enhancing speech intelligibility using variable-rate time-scale modification

The method and preprocessor enhances the intelligibility of narrowband speech without essentially lengthening the overall time duration of the signal. Both spectral enhancements and variable-rate time-scaling procedures are implemented to improve the salience of initial consonants, particularly the perceptually important formant transitions. Emphasis is transferred from the dominating vowel to the preceding consonant through adaptation of the phoneme timing structure. In a further embodiment, the technique is applied as a preprocessor to a speech coder.
Owner:NUANCE COMM INC

Systems and methods for reducing speech intelligibility while preserving environmental sounds

ActiveUS20090306988A1Reduced speech intelligibilitySecret communicationSpeech synthesisSyllableRelative energy
An audio privacy system reduces the intelligibility of speech in an audio signal while preserving prosodic information, such as pitch, relative energy and intonation so that a listener has the ability to recognize environmental sounds but not the speech itself. An audio signal is processed to separate non-vocalic information, such as pitch and relative energy of speech, from vocalic regions, after which syllables are identified within the vocalic regions. Representations of the vocalic regions are computed to produce a vocal tract transfer function and an excitation. The vocal tract transfer function for each syllable is then replaced with the vocal tract transfer function from another prerecorded vocalic sound. In one aspect, the identity of the replacement vocalic sound is independent of the identity of the syllable being replaced. A modified audio signal is then synthesized with the original prosodic information and the modified vocal tract transfer function to produce unintelligible speech that preserves the pitch and energy of the speech as well as environmental sounds.
Owner:FUJIFILM BUSINESS INNOVATION CORP

Robust and reliable acoustic echo and noise cancellation system for cabin communication

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes an adaptive speech enhancement filter for receiving an audio signal that includes a first component indicative of the spoken voice, a second component indicative of a feedback echo of the spoken voice and a third component indicative of the ambient noise, the speech enhancement filter filtering the audio signal by removing the third component to provide a filtered audio signal, the speech enhancement filter adapting to the audio signal at a first adaptation rate, and an adaptive acoustic echo cancellation system for receiving the filtered audio signal and removing the second component in the filtered audio signal to provide an echo-cancelled audio signal, the echo cancellation signal adapting to the filtered audio signal at a second adaption rate, wherein the first adaptation rate and the second adaptation rate are different from each other so that the speech enhancement filter does not adapt in response to operation of the echo-cancellation system and the echo-cancellation system does not adapt in response to operation of the speech enhancement filter.
Owner:LEAR CORP

Voice intelligibility enhancement system

Intelligibility of a human voice projected by a loudspeaker in an environment of high ambient noise is enhanced by processing a voice signal in accordance with the frequency response characteristics of the human hearing system. Intelligibility of the human voice is derived largely from the pattern of frequency distribution of voice sounds, such as formants, as perceived by the human hearing system. Intelligibility of speech in a voice signal is enhanced by filtering and expanding the voice signal with a transfer function that approximates an inverse of equal loudness contours for tones in a frontal sound field for humans of average hearing acuity.
Owner:DTS

Sound perception using frequency transposition by moving the envelope

The application relates to a method of improving a user's perception of an input sound. The application further relates to an audio processing device and to its use. The object of the present application is to increase the sound quality of a sound signal as perceived by a user, e.g. a hearing impaired user. The method comprises a) Defining a critical frequency fcrit between a low frequency range and a high frequency range; b) Analyzing an input sound in a number of frequency bands below and above said critical frequency; c) Defining a cut-off frequency fcut below said critical frequency fcrit; d) Identifying a source frequency band above said cut-off frequency fcut; e) Extracting the envelope of said source band; f) Identifying a corresponding target band below said critical frequency fcrit; g) Extracting the phase of said target band; h) Combining the envelope of said source band with the phase of said target band. This has the advantage of increasing the sound quality, and the potential to further improve speech intelligibility in frequency transposition, e.g. frequency lowering systems. The invention may e.g. be used in communication devices, such as telephones, or listening devices, e.g. hearing instruments, headsets, head phones, active ear protection devices or combinations thereof.
Owner:OTICON

Spectral enhancement using digital frequency warping

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.
Owner:GN HEARING AS

Hearing Device Sound Simulation System and Method of Using the System

The present invention relates to hearing aid training systems (100). More particularly, the present invention relates to the simulation of a hearing aid environment (108) prior to a user's (105) purchase of a hearing aid. To create the simulated environment, the user's hearing profile (111) is collected from all prior hearing tests. Prior hearing tests include information on all aspects of the user's hearing, such as frequency and speech intelligibility. The software program (126) of this invention, and the audiologist using the software program (126), analyzes the user's hearing profile and creates a simulation that demonstrates to the user how he or she would hear with a hearing aid. Furthermore, this invention provides a way to make additional adjustments to the hearing aid's DSP data based upon user preferences prior to ordering the individual customized hearing aid.
Owner:JOHNSON & JOHNSON CONSUMER COPANIES

Method and apparatus for filtering and compressing sound signals

InactiveUS6873709B2Preserve spectrum contrastPreserving intelligibilityGain controlHearing aids signal processingFrequency spectrumSpeech sound
Improved approaches are disclosed to filter and compress sound signals so as to achieve not only speech audibility and intelligibility at low levels but also preserves spectrum contrast at high levels. According to one aspect of the invention, gain amounts for different frequency bands are individually constrained based on signal levels for the frequency bands. Hence, the gain amounts for each of the frequency bands may or may not be constrained depending on the corresponding signal levels. As a result, the most critical information for speech intelligibility, speech clarity, and speech quality can be made available to hearing impaired people over wide range of signal level. The invention is particularly useful for hearing aids or other sound systems for the hearing impaired.
Owner:OTOTRONIX

Speech enhancement to improve speech intelligibility and automatic speech recognition

The present invention provides a system and method to enhance speech intelligibility and improve the detection rate of automatic speech recognizer in noisy environments. The present invention reduces an acoustically coupled loudspeaker signal from a plurality of microphone signals to enhance a near end user speech signal. A decision unit checks a system configuration parameter to determine if the cleaned speech is intended for human communication and / or Automatic Speech Recognition (ASR). A formant emphasis filer and a spectrum band reconstruction unit are used to further enhance the speech quality and improve the ASR recognition rate. The present invention can also apply to devices which has a foreground microphone(s) and a background microphone(s).
Owner:LOU XIA

Time-domain receive-side dynamic control

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.
Owner:BLACKBERRY LTD

Speech Intelligibility

The perceived quality of a speech signal output from a user apparatus is improved by storing ambient noise profiles each indicating a model power distribution of a respective ambient noise type as a function of frequency; the ambient noise profile at the user apparatus is measured, the measured ambient noise profile is correlated with each of the stored ambient noise profiles, the stored ambient noise profile is selected with which the measured ambient noise profile is most highly correlated, and the speech signal is manipulated in dependence on which of the stored ambient noise profiles is selected, so as to form an improved speech signal.
Owner:QUALCOMM TECH INT

System for processing an audio signal to enhance speech intelligibility

An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.
Owner:DTS

System and method of detecting speech intelligibility of audio announcement systems in noisy and reverberant spaces

A system and method to detect and remediate unacceptable levels of speech intelligibility evaluates received test audio transmitted across and received in a space or region of interest. Intelligibility is improved by altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.
Owner:HONEYWELL INT INC

Method and apparatus for voice clarity and speech intelligibility detection and correction

Systems, methods and apparatus are described herein for continuously measuring voice clarity and speech intelligibility by evaluating a plurality of telecommunications channels in real time. Voice clarity and speech intelligibility measurements may be formed from chained, configurable DSPs that can be added, subtracted, reordered, or configured to target specific audio features. Voice clarity and speech intelligibility may be enhanced by altering the media in one or more of the plurality of telecommunications channels. Analytics describing the measurements and enhancements may be displayed in reports, or in real time via a dashboard.
Owner:VAIL SYST

Hearing aid and a method for enhancing speech intelligibility

A hearing aid (22) having a microphone (1), a processor (53) and an output transducer (12), is adapted for obtaining an estimate of a sound environment, determining an estimate of the speech intelligibility according to the sound environment estimate, and for adapting the transfer function of the hearing aid processor in order to enhance the speech intelligibility estimate. The method according to the invention achieves an adaptation of the processor transfer function suitable for optimizing the speech intelligibility in a particular sound environment. Means for obtaining the sound environment estimate and for determining the speech intelligibility estimate may be incorporated in the hearing aid processor, or they may be wholly or partially implemented in an external processing means (56), adapted for communicating data to the hearing aid processor via an appropriate link.
Owner:WIDEX AS

Methods and apparatus for cochlear implant signal processing

A cochlear implant processing strategy increases speech clarity and provides higher temporal performance. The strategy determines the power spectral component within each channel, and dynamically selects or de-selects the channels through which a stimulation pulse is provided as a function of whether the spectral power of the channel is high or low. “High” and “low” are estimated relative to a selected spectral power, for example. The selected spectral power can be estimated by signal average or mean, or by other criteria. Once a selection of the channels to stimulate has been made, the system can decide that only those channels are stimulated, and stimulation is removed from the other channels. The selected channels are the ones on which the spectral power is above the mean of all the available channels. Fewer channels are stimulated at any time and the contrast of the stimulation is enhanced. Also, the temporal resolution increases as the number of channels that must be stimulated on a given frame decreases. This way, the channels which are presented to the patient are fewer in number and contain more temporal information.
Owner:ADVANCED BIONICS AG

Methods, devices and systems using signal processing algorithms to improve speech intelligibility and listening comfort

ActiveUS20090226015A1Improve naturalnessWell maintained structureElectrotherapyEar treatmentAuditory neuropathyHearing perception
Methods, devices and systems for improving hearing and for treating hearing disorders, such as auditory neuropathies. A hearing enhancement system of this invention generally comprises; an amplitude modulation processor, a frequency high-pass processor, a frequency upward-shifting processor and a formant upward-shifting processor.
Owner:RGT UNIV OF CALIFORNIA

Digital deaf-aid frequency response compensation method based on mask curve

InactiveCN1870135AResolve claritySolve the problem of decreased intelligibilitySpeech analysisDeaf-aid setsMasking thresholdFrequency response
A method for compensating frequency response of digital deaf-aid based on masked curve includes techniques of time frequency domain switch-over, critical band division, masking threshold calculation and frequency response compensation to improve hearing threshold rising phenomenon caused by hearing-masking effect.
Owner:北达万坤(北京)科技发展有限公司

Method and System for Speech Intelligibility Measurement of an Audio Transmission System

InactiveUS20100211395A1Improved intelligibility score measurementImprove intelligibilitySpeech analysisMeasurement deviceVolumetric Mass Density
Method and processing system for measuring the intelligibility of a degraded output signal (Y(t)) from an audio transmission system (10) in response to a reference input signal (X(t)). A measurement device (11) is arranged for outputting a measure (I) for the speech intelligibility of the output signal (Y(t)). The measurement device (11) executes processing of the input signal (X(t)) and output signal (Y(t)) to obtain a disturbance density function (D(f)n). The disturbance density function (D(f)n) is corrected by multiplying it with a correction function for each frame derived from a correlation calculation of the compensated pitch power densities (PPX′(f)n) associated with the input signal (X(t)) of a present frame (n) and an independent previous frame (n−2). The corrected disturbance density function (D′(f)n) is aggregated over frequency and time to obtain a measure (I) for the speech intelligibility of the output signal (Y(t)).
Owner:NEDERLANDSE ORG VOOR TOEGEPAST-NATUURWETENSCHAPPELIJK ONDERZOEK (TNO) +1

Method and apparatus of increasing speech intelligibility in noisy environments

A method (400, 600, 700) and apparatus (220) for enhancing the intelligibility of speech emitted into a noisy environment. After filtering (408) ambient noise with a filter (304) that simulates the physical blocking of noise by a at least a part of a voice communication device (102) a frequency dependent SNR of received voice audio relative to ambient noise is computed (424) on a perceptual (e.g. Bark) frequency scale. Formants are identified (426, 600, 700) and the SNR in bands including certain formants are modified (508, 510) with formant enhancement gain factors in order to improve intelligibility. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains factors yielding combined gains which are clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532) and used to reconstruct (532, 534) an audio signal.
Owner:GOOGLE TECH HLDG LLC

Circuit for improving the intelligibility of audio signals containing speech

The speech intelligibility of an audio signal of unchanged volume is improved by raising the total audio signal by a constant factor and lowering the amplitude of this raised signal by a high-pass filter. The corner frequency fc of the high-pass filter is adjusted such that the output amplitude of the audio signal at the end of the processing segment is equal or proportional to the input amplitude of the audio signal.
Owner:ENTROPIC COMM INC

Speech Intelligibility Measurement and Open Space Noise Masking

Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes outputting a test signal from a speaker, the test signal operable to measure a speech transmission quality of a transmission channel. The method includes receiving the test signal at a microphone, outputting a detected test signal, and processing the detected test signal to determine the speech transmission quality of the transmission channel. The method further includes adjusting an output level of a noise masking signal from the speaker responsive to the speech transmission quality.
Owner:PLANTRONICS

Apparatus and method for enhancing speech intelligibility in a mobile terminal

An apparatus and a method for enhancing speech intelligibility in a mobile terminal. A complex spectrum calculator calculates complex spectra of one input frame of an input speech signal by Fourier transform, a speech level calculator calculates its instant levels, an average speech level calculator calculates an average speech level of the speech frame using the instant levels, if the input frame is a speech frame, a scaling factor calculator calculates scaling factors by comparing the average speech level with the instant levels, an HPF characteristic calculator calculates amplitude characteristics using the scaling factors, a HPF high-pass-filters the complex spectra using the amplitude characteristics, a synthesizer converts high-pass-filtered signals to time signals by inverse Fourier transform and synthesizes the time signals, and a combiner outputs an enhanced intelligibility speech signal by combining the synthesized time signal with the input frame.
Owner:SAMSUNG ELECTRONICS CO LTD
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