218 results about "Voice source" patented technology

The invention provides a method for enhancing microphone array voice based on combined inhibition. The method comprises the following steps of: structuring a microphone array for receiving external signals; analyzing the signals and obtaining time delays of different array signals relative to benchmark array signals in the microphone arrays opposite to a target voice source; respectively performing time delay compensation on digital signals corresponding to the two microphones, obtaining the compensated signals; respectively performing subband decomposition on the compensated array signals, and then forming fixed beams on each subband; meanwhile, respectively using blocking matrixes on each subband to obtain noise reference signals on each subband; and then respectively removing the noiseirrelative to the target voice form the fixed beam forming device on corresponding subband through an adaptive filtering processing algorithm, and then merging the subbands, thereby forming an initial gain signal; and meanwhile, making use of the previously compensated any two array signals to obtain a filter for inhibiting the noise signal related to the target voice through a recursive mutual power spectral density, thereby obtaining the final target voice signal through combining the initial gain signal.

The invention provides a multidimensional user identity identification method. The multidimensional user identity identification method comprises the steps of: detecting a human body in a photographic range by means of a camera, extracting facial features of the human body, comparing the facial features with user pictures prestored in a sample library, calculating a face matching coefficient, and preliminarily judging whether a user with permission exists in the photographic range; receiving user voice by using a microphone, converting an audio analog signal into a digital sequence, comparing the digital sequence with user voiceprints prestored in the sample library, calculating a voiceprint matching coefficient, and calculating the face matching coefficient and the voiceprint matching coefficient again to obtain a matching degree, so as to judge whether the user have the permission; and establishing a model for the user when judging that the user has the permission, carrying out human body dynamic tracking on the user, matching a voice source position with a position calculated through carrying out human body tracking on the user, judging that a command is issued by the user with permission and the command is valid when a voice position matches with an image position, and executing the command.

The invention relates to a signal processing method for enhancing target voice signal pickup in a sound environment. The method includes the steps that (1), parameters of an ESN are acquired through experiments, and a corresponding sound source model is established; (2), the model is used for two occasions, namely, when output of the model is certain expected target voice signals and input of the model is a mixture of sound environment reflected sound signals of a target voice source and the target voice signals, the model can be used for echo cancellation of field amplified sound; when the output of the model is the expected target voice signals and the input of the model is a mixture of sound environment reflected sound signals of another specific voice source and the target voice signals, the model can be used for echo cancellation of voice communication between two specific persons; (3), when the model is used by a target voice person in the actual sound environment and the pickup position changes, reflected signals of sound source signals pointed by training can be suppressed as well, and then corresponding enhanced target voice signals are output. By the adoption of the signal processing method, the influence of the change of the pickup position on quality of the voice signals is overcome.

The present invention relates to technologies of implementing voice source tracking by using a microphone array. A method and an apparatus for a mobile terminal to implement voice source tracking. The method comprises: collecting outside voice information by using a microphone array (101); analyzing the outside voice information to determine target voice source information (102); and collecting a delay time of the target voice source information according to the microphone array to calculate a position of a target voice source (103). The apparatus comprises: a voice source information collection module (20), configured to collect outside voice information by using the microphone array; and a voice source information calculation processing module (30), configured to analyze the outside voice information, to determine target voice source information; and collect a delay time of the target voice source information according to the microphone array, to calculate a position of a target voice source.

The invention provides a voice amplifying method and a mobile terminal, and solves the problem of a non-ideal voice amplification effect caused by changing a holding posture, as the holding posture is changeless, of existing voice amplification equipment. The voice amplifying method provided by the invention comprises the following steps: acquiring audio signals received by a plurality of microphones of the mobile terminal; determining the microphone, closest to a voice source, in the plurality of microphones as a primary microphone according to audio information of the audio signals received by each microphone; performing amplification process on the audio signals received by the primary microphone; and outputting amplified audio signals through a loudspeaker. According to embodiments of the invention, when the holding posture of the voice amplification equipment changes, amplification process is performed on the audio signals received by the primary microphone closest to the voice source, so that noise interference is reduced, outputted audio signals are clearer, and user experience of a user is improved.

The invention provides an electronic larynx speech reconstructing method and a system thereof. The method comprises the following steps of: firstly, extracting model parameters form collected speech as a parameter library; secondly, collecting the face image of a sounder, and transmitting the face image to an image analysis and processing module to obtain the sounding start moment, the sounding stop moment and the sounding vowel category; thirdly, synthesizing a voice source wave form through a voice source synthesizing module; and finally, outputting the voice source wave form through an electronic larynx vibration output module. Wherein the voice source synthesizing module is used for firstly setting the model parameters of a glottis voice source to synthesize the glottis voice source wave form, then simulating the transmission of the sound in the vocal tract by using a waveguide model and selecting the form parameters of the vocal tract according to the sounding vowel category so as to synthesize the electronic larynx voice source wave form. The speech reconstructed by the method and the system is closer to the sound of the sounder per se.

The embodiment of the invention discloses a method, an apparatus and a device for achieving co-location of voices and images and a medium. The method comprises the steps of: performing image recognition and voice recognition for a current playing video to obtain image features and voice features corresponding to the current playing video; based on the feature images, determining whether the current playing video has a sound production source; if yes, obtaining position information of the sound production source of the current playing video at a current video display screen; based on the voicefeatures, determining whether the current playing video has a voice source matching with the sound production source or not; if yes, generating control signals according to the position information ofthe sound production source in the current video display screen to control the voice corresponding to the position information to restore element sound production, wherein a preset image feature database is constructed in advance according to the current playing video. Through the technical scheme, the play effect of the video can be improved to bring high sense of immediacy for watchers.

The invention provides a voice signal processing method and device. The method comprises the steps that the relative positional relation between a user and an electronic device is obtained; a first distance between a voice source of voice signals to be collected and the electronic device is determined according to the relative positional relation; adjusting signal intensity compensating parameters of the voice signals in the voice signals collected by a voice collecting unit according to the first distance. According to the voice signal processing method, the quality of the voice signals output by the electronic device can be improved.

The invention discloses a robot voice addressed rotation system and a method thereof. The robot voice addressed rotation system comprises a voice activity detection unit used for detecting a received voice signal and judging whether a user sends out a voice signal or not, a voice addressed unit used for calculating the addressing angle of the voice signal, and a voice addressed rotation unit used for driving a robot to turn to the direction of the voice source of the voice signal according to the addressing angle of the voice signal. In this way, the voice recognition accuracy can be improved. Meanwhile, the voice recognition accuracy and the signal-to-noise ratio are improved. Based on the robot voice addressed rotation method, the voice signal sent from the user can be accurately and effectively obtained. Therefore, the signal-to-noise ratio and the voice recognition accuracy are improved.

The present invention provides an electronic terminal and a voice signal processing method. The electronic terminal comprises a microphone module which comprises three microphones, a positioning module which is used for determining the position of a target voice source according to the voice signals collected by the at least three microphones at the beginning of a call, a determining module for determining a sound recording range according to the position of the target voice source, and a processing module used for obtaining the voice signal in the sound recording range and transmitting according to the voice signal in the sound recording range. In the invention, the target voice source is positioned through the microphone module, the sound recording range is determined, only the voice signal in the sound recording range is obtained, thus the environment noise can be effectively filtered, and the conversation effect is improved.

The present invention to provide an earphone comprising a casing, a loudspeaker provided in the casing and adapted to amplify sound in response to receiving a voice source signal, a cap fixedly mounted onto the casing, a piezoelectric microphone provided in the cap and a sound barrier member provided between the piezoelectric microphone and the loudspeaker, wherein the sound barrier member has a diameter larger than that of the piezoelectric microphone but smaller than that of the loudspeaker such that sound amplified by the loudspeaker is adapted to transmit beyond the cap through the sound barrier member and sound vibrations transmitted to the piezoelectric microphone are substantially absorbed. Thus, sound amplified by the loudspeaker does not interfere with the external sound receiving capability of the piezoelectric microphone.

An embodiment of the invention provides a voice noise reduction method, equipment and a storage medium, wherein the method comprises the steps of acquiring a voice signal which is acquired by a microphone; according to the voice signal and a transfer function between a voice source and a microphone, determining a source signal emitted from the voice source; and if a target standard voice signal which matches the source signal exists in at least one standard voice signal, performing filtering on the voice signal which is acquired by the microphone in reference to the target standard voice signal. In the voice noise reduction method, multiple types of voice signals in the voice signal which is acquired by the microphone can be filtered, and a useful signal acquired after noise reduction is same with a standard voice signal, thereby effectively preventing interference of the noise signal to voice processing.

The invention relates to a pharyngeal cavity electronic larynx synthesis and communicated system which can be automatically regulated and a method thereof, wherein a device based on a software platform and external hardware of a computer comprises a camera, a microphone and an electronic larynx oscillator; the automatic control to the working state of the electric larynx and the voice source synthesis of the pharyngeal cavity is realized through extracting visual phonetic feature information of movement images of the face and the neck of a user, thereby the electronic larynx can be used without holding by hands, which is simpler and more convenient, and the problems that the synthesis voice source is not consistent with the electronic larynx applying part and the voice of the electronic larynx is mechanical and unnatural are solved; and meanwhile, the dynamic de-noising enhancement treatment is implemented for the voice rebuilding of the pharyngeal cavity electronic larynx, thereby the quality and the intelligibility of the rebuilt voice are improved; and the remote real-time communication of the voice of the electronic larynx is realized through the network transmission technique, the application range of the electronic larynx is further expanded, and the life quality of a patient after laryngectomy is improved.

The invention relates to a voice acquisition method and device, computer equipment and a storage medium. The voice acquisition method comprises the steps that an audio stream is received and divided into a plurality of voice segments; the voiceprint characteristics of the voice segments are extracted; and the voiceprint characteristics of the voice segments are compared in a paired mode to obtaina fraction matrix corresponding to the audio stream, and the fraction matrix is used for expressing the corresponding fractions of the similarity of paired voice fragments; the voiceprint characteristics are clustered according to the fraction matrix, and the voice fragment numbers corresponding to multiple voice sources are obtained; and the corresponding voice segment is extracted according to the voice source with the largest voice fragment number. Through the voice acquisition method, the phenomenon that later manual operation is increased due to the voice quality problem can be avoided, and the voice acquisition efficiency is improved.

The invention discloses a voice-to-character conversion method and device. The voice-to-character conversion method comprises steps of using more than two microphones to collect voice information of more than one users, performing analysis processing on voice information collected by the microphone to obtain a voice source characteristic parameter of each user, classifying the collected voice information according to the voice source characteristic parameter of each user to obtain voice information corresponding to each user, and converting voice information corresponding to each user to corresponding character information.

Disclosed is a phase difference measurement-based microphone array direction finding method. The phase difference measurement-based microphone array direction finding method comprises the steps of obtaining the sampling data of a microphone array; performing fast Fourier transform on the sampling data of the microphone array according to channel sequence numbers to obtain the frequency domain data of the microphone array; selecting available frequency units; according to the available frequency units, computing corresponding array distance difference vectors; utilizing the array distance difference vectors of all the available frequency units to compose the high-dimensional linear system of equations of voice signal direction vectors, cooperatively solving the direction vector estimated values of voice source signals; according to the direction vector estimated values of the voice source signals, computing the azimuth angle estimated values and the pitch angle estimated values of the voice source signals. The phase difference measurement-based microphone array direction finding method has the advantages of being high in direction finding speed and precision, and particularly through the phase difference information of multiple available frequency units of the voice signals, obtains high azimuth angle and pitch angle estimating performance.