53 results about "Intelligibility (communication)" patented technology

The embodiment of the invention provides a voice processing method, device and system, a storage medium and a program product, and relates to the technical field of terminals. According to the voice processing method, in a face-to-face communication scene and a conversation scene, the speech barrier voice of the speech barrier user can be restored, the restored voice is obtained, the intelligibility of the restored voice is higher than that of the speech barrier voice, so that the speech barrier user can communicate by inputting the voice, and the user experience is improved. According to the invention, the technical problem of insufficient communication convenience of the speech-impaired users can be solved, so that the technical effect of improving the communication convenience of the speech-impaired users is realized.

The invention provides an ultra-low bit rate voice coding and decoding system based on text semantic information fidelity, and relates to the technical field of voice coding and decoding. The system comprises the following steps: performing voice feature extraction and text feature extraction on original voice through a multi-modal text-voice combined encoder to obtain voice features and text features, embedding the text features into the voice features to obtain text-voice features of the original voice, and performing voice feature extraction and text feature extraction on the original voice through a multi-modal text-voice combined encoder; sending the text-speech features of the original speech to a receiving end in a semantic communication system; and the receiving end inputs the received text-speech features into a text semantic fidelity decoder based on an attention mechanism to obtain reconstructed speech features and reconstructed text features, decodes the reconstructed speech features layer by layer, and performs attention calculation on the reconstructed speech features and the reconstructed text features to obtain decoded speech. By means of the voice coding and decoding technology, good voice perception quality can still be kept under the ultra-low code rate, and voice intelligibility is guaranteed.

The embodiments of the present application relate to the technical field of terminals. Provided are a speech processing method, apparatus and system, and a storage medium and a program product. In face-to-face communication scenarios and call scenarios, the speech processing method can be used to restore impaired speech of a speech-impaired user, so as to obtain restored speech, and the restored speech has higher intelligibility than the impaired speech. In this way, the speech-impaired user can also communicate by means of inputting speech. Therefore, the technical problem of the convenience of communication for speech-impaired users being insufficient can be solved, thereby achieving the technical effect of improving the convenience of communication for speech-impaired users.

The invention belongs to the technical field of audio coding, and particularly discloses an air traffic control audio coding method and system based on dynamic acoustic masking, and the method comprises the steps: collecting an air traffic control audio signal, and carrying out the complex short-time Fourier transform to generate an audio time-frequency diagram; inputting the time-frequency graph into a personalized auditory feature extraction model, extracting a physiological feature graph and a dynamic weight matrix, and fusing the physiological feature graph and the dynamic weight matrix to generate a final masking matrix; performing Hadamard product on the masking matrix and the time-frequency map to obtain a perception saliency map; extracting audio features based on the perceptual saliency map, constructing a basic codebook and a tree residual codebook to perform hierarchical vector quantization compression, and outputting compression representation; and converting the compressed representation into a coding triple consisting of a basic index, a path index and a termination flag bit, and taking the coding triple as a final coding result. The method adapts to individual hearing differences, improves instruction intelligibility and compression efficiency, and is suitable for high-noise aviation communication scenes.

The invention relates to a self-adaptive voice semantic communication method based on hierarchy time sequence importance, which belongs to the field of voice analysis, and comprises the following steps: a sending end performs priority division on a discrete feature matrix according to hierarchy and time sequence attributes of voice features, and screens the features according to importance scores in a feature matrix packaging and selecting link; the method comprises the following steps: interacting with wireless communication through a channel adaptive scheduling module, introducing a channel feedback mechanism, and dynamically adjusting transmission power and a resource allocation strategy by sensing current channel state information; and after completing signal demodulation, a receiving end inputs the acquired sparse feature flow into a voice restoration module, and globally reconstructs the received features by using voice priori knowledge of deep pre-training. According to the method, a hierarchical speech feature extraction technology based on a discrete codebook and a generative semantic repair technology are combined, so that the speech word error rate in a severe channel environment is reduced to the maximum extent, and the semantic intelligibility of a receiving end is improved.

The invention relates to the technical field of earphones, in particular to a microphone structure, an earphone and a test method. The microphone structure comprises a shell and a microphone assembly, the shell comprises a pickup hole, an exhaust hole and a containing cavity, the microphone assembly is located in the containing cavity, the microphone assembly and the outer wall of the shell are arranged in a spaced mode, the microphone assembly divides the containing cavity into a first cavity and a second cavity which are independent from each other, the pickup hole is communicated with the first cavity, and the exhaust hole is communicated with the second cavity. The earphone comprises the microphone structure. The testing method is used for testing the acoustic performance of the earphone. According to the microphone structure, the earphone and the test method, the noise reduction effect can be improved, and the definition of voice communication is ensured.

The invention belongs to the field of interphones, and provides a mobile phone auxiliary type intelligent interphone and an implementation method thereof, and the mobile phone auxiliary type intelligent interphone comprises a mobile phone end assembly which comprises a central processing unit, an audio processing chip, a first connection module, a loudspeaker and a microphone; the interphone end assembly comprises a main control chip, a wireless communication module, a second connection module and a power management module; a radio frequency transceiver chip is arranged in the wireless communication module, data transmission is carried out between the second connection module and the first connection module, and the audio processing chip adopts a CNN noise reduction model to process digital audio signals; according to the method, the digital audio signals are processed through the CNN noise reduction model, various complex background noises can be accurately identified and removed, clear voice signals can be restored, the voice definition and intelligibility are greatly improved, it is ensured that both communication parties can accurately transmit information in a noisy environment, and the user experience is improved. The problem of poor voice quality caused by noise interference in a traditional matching mode is effectively solved.

The invention relates to the technical field of intelligent plotting, and provides an intelligent plotting method based on voice input. The intelligent plotting method based on voice input comprises the following steps: S1, receiving an original audio signal containing a voice instruction input by a user in real time at a terminal equipment side; according to the intelligent plotting method based on voice input, the problem of communication interruption is solved, the problem of voice recognition under high-strength physical vibration is solved through an innovative voice reconstruction technology, full-domain and full-condition adaptation in the true sense is achieved, vibration distortion reconstruction is preferentially carried out on the voice signals, and therefore the voice recognition efficiency is improved. The method has the advantages that the jittering voice signals are corrected before the jittering voice signals are sent to the recognition engine, so that the voice definition and intelligibility are guaranteed from the information source, the accuracy of all subsequent processing links is greatly improved, and the method is particularly suitable for key instructions which are issued in high-pressure and high-risk environments and have success or failure of events and even life safety. And the reliability guarantee is provided.

The invention provides a PDMA-based ultra-long distance speech enhancement system, which comprises a hardware acquisition end and a signal processing end, and is characterized in that the hardware acquisition end comprises a parabolic reflector and a differential microphone array, the hardware acquisition end is used for acquiring an original mixed signal, the original signal passes through the parabolic reflector to obtain a pre-enhanced signal, and the pre-enhanced signal is used for processing the pre-enhanced signal; an original signal passes through the differential microphone array to obtain a differential signal, the obtained pre-enhanced signal and the differential signal are input to the signal processing end, and the signal processing end carries out fusion processing on the received pre-enhanced signal and the differential signal to realize secondary enhancement of a target and output a target voice signal. According to the invention, the voice signal can be effectively enhanced in an ultra-long distance scene of more than 20 meters, the voice quality and intelligibility can be obviously improved, the requirements of daily communication transmission are met, the complexity of the system is reduced, the declaration cost is controlled, and the performance of the system is improved.

The invention relates to a communication device for simultaneous two-way voice communication between at least two call stations (3) on a printing substrate processing machine, the call stations (3) being connected to each other via a bus system (4). The invention is characterized in that each calling station (3) has at least one loudspeaker (2) and one microphone device (1) with a plurality of microphone pockets (6) and one printed circuit board (8) with electronic components for improving speech intelligibility during speech communication between the calling stations (3).

The invention discloses a high-gain directional broadcasting device for an expressway and an installation method thereof, and relates to the technical field of communication, the high-gain directional broadcasting device comprises a high-gain directional broadcasting unit and a central management and control platform, and the high-gain directional broadcasting unit and the central management and control platform are integrated through acoustic beam forming, environmental perception and vehicle detection and high-performance digital signal processing and control. Accurate wave beam forming, real-time environment self-adaption and target vehicle directional transmission of sound waves are achieved. The method comprises the steps of field survey layout, installation height and initial angle optimization, system integration debugging, network communication and adaptive operation optimization. Through adoption of the above scheme, accurate sound wave release can be realized, the broadcast definition and intelligibility are remarkably improved, the energy consumption and the operation and maintenance cost are effectively reduced, and the system reliability is improved.

The invention relates to a hearing-impaired voice conversion and generation method based on multi-language self-adaption, and belongs to the field of voice processing and artificial intelligence. According to the invention, hearing-impaired voices in multiple languages are converted into standard voices, so that normal communication with hearing-impaired people is facilitated. According to the method, acoustic features of hearing-impaired speech are extracted and aligned with target language phonemes, the acoustic features are input into a speech adaptation model, standard speech features are generated, multi-language training is carried out based on the standard speech features and target language corpus, and the training result is used for generating target language natural speech. According to the invention, the intelligibility of hearing-impaired voice can be improved, so that the voice is closer to natural pronunciation. According to the invention, rapid adaptation of multiple languages can be realized, and the cost of cross-language training is reduced. The system can be applied to education, medical rehabilitation, cross-border communication and other scenes, and provides effective voice communication assistance for hearing-impaired people.

A hearing protection device (HPD) (110) and method include over-the-ear (OTE) earcups (114, 116) combining built-in ITE occlusion testing with compensated audio signals using previously stored hearing frequency profiles associated with a user / use or type / brand of earplugs used with the HPD. Audio signals are frequency adjusted to enhance desired content for intelligibility and user safety. Audio and / or volume compensation may be applied to wired or wirelessly received communications received by the HPD and / or to ambient sounds to adjust frequency-specific audio energy based on the earplug occlusion to enhance desired sounds such as speech and diminish undesired noise to improve situational awareness for users. The HPD may generate the hearing frequency profiles and / or remotely-generated profiles may be transferred to the HPD. Noise dosing information associated with sound level exposure over time may be used to adjust the ambient pass-thru cutoff threshold. Active noise cancellation (ANC) with feedback and / or feedforward control may also be performed.

The invention relates to the technical field of vehicle control, in particular to a vehicle-mounted communication method and device, electronic equipment and a vehicle, and the vehicle-mounted communication method comprises the steps: obtaining a multi-channel microphone signal; performing sound source classification on the multi-channel microphone signal to obtain a plurality of sound source signals; obtaining a volume adjustment instruction; selecting a target sound source signal needing volume adjustment from the plurality of sound source signals according to the volume adjustment instruction, and performing volume adjustment on the target sound source signal according to the volume adjustment instruction; and carrying out sound mixing on the target sound source signal after volume adjustment and other sound source signals except the target sound source signal to obtain a target sound signal, and sending the target sound signal to a communication opposite end. Through sound source classification and directional regulation and control, fine processing of different directions and types of sound is realized, the voice definition of a main talker is reserved, and specific environment sound, interference sound or key prompt sound can be flexibly and selectively reserved or suppressed according to an actual scene.

The embodiment of the invention provides a voice processing method, device and system, a storage medium and a program product, and relates to the technical field of terminals. According to the voice processing method, in a face-to-face communication scene and a conversation scene, the speech barrier voice of the speech barrier user can be restored, the restored voice is obtained, the intelligibility of the restored voice is higher than that of the speech barrier voice, so that the speech barrier user can communicate by inputting the voice, and the user experience is improved. According to the invention, the technical problem of insufficient communication convenience of the speech-impaired users can be solved, so that the technical effect of improving the communication convenience of the speech-impaired users is realized.

The invention relates to the technical field of voice signal processing and industrial communication, and discloses a mine scene voice enhancement method and device based on multi-modal noise feature learning, and the method comprises the following steps: synchronously collecting acoustics, vibration and acceleration signals of a mine scene, and extracting multi-dimensional features after trend term removal and standardization preprocessing; dynamically fusing the vibration and acceleration characteristics representing the physical working condition of the pure noise source with the acoustic characteristics by using a self-attention mechanism; inputting the fusion features into a CNN-LSTM deep network, and reasoning to obtain an ideal ratio masking vector; and using the masking vector to correct the amplitude spectrum, and combining original phase reconstruction to generate enhanced speech. According to the method, the non-acoustic physical mode is introduced to serve as auxiliary information, causal association between deep network mining mechanical vibration and acoustic noise is utilized, the problem that the non-stationary impact noise is difficult to restrain under the low signal-to-noise ratio is effectively solved, and the definition and intelligibility of voice communication in the complex mine environment are improved.

The embodiment of the invention provides a voice processing method and device and a medium. When the method is specifically executed, the method comprises the following steps: aiming at voice information containing background noise and reverberation voice information, firstly carrying out noise reduction processing on the voice information by adopting a multi-target deep neural network; and then de-reverberation processing is carried out on the voice information after noise reduction processing, and pure voice information is obtained. According to the embodiment of the invention, background noise and reverberation voice information are removed through a two-section mode, and compared with a mode of directly mapping the spectrum characteristics of pure voice from the spectrum characteristics of noisy voice, the quality and the intelligibility of the obtained pure voice information can be remarkably improved, so that the communication effect is improved, and the user experience is improved. And the user experience is improved.

The present application relates to the technical field of speech processing, and provides a speech repair method and device, electronic equipment and storage medium, the method comprising: determining a pathological speech to be repaired, and identifying a phoneme sequence of the pathological speech; correcting the phoneme sequence based on a phoneme correction model to obtain corrected text, the phoneme correction model being trained based on sample damaged phoneme sequences and corresponding standard texts; and performing speech synthesis on the corrected text to obtain a repaired speech of the pathological speech. The speech repair method, device, electronic equipment and storage medium provided by the present application not only improve the sound quality, but also further retain and repair the semantic information in the pathological speech, thereby comprehensively improving the intelligibility and communication efficiency of the repaired speech.

The utility model discloses a two-way talkback processing system, and belongs to the field of voice signal processing. The system combines an advanced audio processing technology, and particularly uses an audio echo processing chip and an audio power amplifier module. Through an accurate echo cancellation algorithm (including frequency domain and time domain filtering), dynamic noise suppression and real-time gain adjustment, the system can significantly improve audio signal quality, avoid echo and self-excitation squeal phenomena, and ensure voice definition. The two-way audio signal stream design and the modular structure of the system enable the system to adapt to complex noise environments and various audio application scenarios, including talkback between drivers and passengers and processing of emergency alarm signals. In addition, through the design of audio bus transmission, audio power amplifier module amplification, transformer isolation and the like, stable transmission and high-quality output of audio signals are ensured. On the whole, the stability, definition and adaptability of the audio communication system are improved, and the echo problem in a traditional system is solved.

A method for enhancement of speech intelligibility in a device arranged for a near-end side a communication with a far-end device. The method involves calculating a measure of speech intelligibility at the near-end side based on a near-end audio input and a far-end audio input. Then, based on the calculated measure of speech intelligibility optimizing parameters of a predetermined speech enhancement algorithm, where a predetermined speech intelligibility target, and an additional target are taken into account to generate an optimized speech enhancement algorithm. Next, processing the far-end audio input according to the optimized speech enhancement algorithm, and generating a near-end audio output accordingly. The algorithm can adapt to changing noise conditions and be optimized for both speech intelligibility and another target. This can be used to minimize delay, electric power consumption and audio quality while satisfying the speech intelligibility target. The optimization can be based on a closed-form solution.

The present invention relates to a single-channel speech enhancement method based on a multi-attention mechanism, and belongs to the technical field of audio signal processing. The present invention introduces a complex Conformer into a complex U-Net network to model the correlation between speech amplitude and phase, uses a three-dimensional attention mechanism to construct richer features to enhance the representation ability of the convolutional layer, and fuses speech detail features and deep features through a gated attention mechanism. The method can improve speech quality and intelligibility, and can be used for voice communication in noisy environments, command control, and the pre-processing part of speech-related tasks. Experimental results on public datasets show that the proposed method achieves evaluation results of 3.09, 4.28, 3.47, 3.72, and 95.07 on five objective evaluation indicators: PESQ, CSIG, CBAK, COVL, and STOI, respectively, and can effectively reduce noise and improve speech quality and intelligibility.

Systems and methods for vehicles to communicate with people outside the vehicle. In particular, a two-way communication system is provided including exterior microphone arrays on the vehicle. Calls performed from the exterior of a vehicle can be difficult to implement with acceptable quality in noisy environments. A pipeline having multiple beamformers is provided to improve call quality, where the pipeline operates on the incoming data for each of the microphone arrays. Each beamformer estimates direction of arrival (DOA) of the speech signal, and utilizes the estimated DOA to generate spatial filtering coefficients to filter the signal. A beamformer can be used to generate a one channel output for each microphone array, and a mixer can be used to mix the signal output from each microphone array. The beamformer enables reliable two-way communication with acceptable listening quality and good intelligibility for the remote assistant listening to the captured signal.

The application relates to an adaptive voice semantic communication method based on hierarchical time sequence importance, and belongs to the voice analysis field, which comprises the following steps: a sending end divides a discrete feature matrix according to the priority of voice features and time sequence attributes, and filters the features according to the importance score in the feature matrix encapsulation and selection link; a channel adaptive scheduling module is interacted with wireless communication, a channel feedback mechanism is introduced, the current channel state information is perceived, and the transmission power and resource allocation strategy are dynamically adjusted; after signal demodulation is completed, a receiving end inputs the obtained sparse feature stream to a voice repair module, and global reconstruction is performed on the received features by using deep pre-trained voice prior knowledge. The application combines a hierarchical voice feature extraction technology based on a discrete codebook and a generative semantic repair technology, so that the voice word error rate in a severe channel environment is maximally reduced, and the semantic intelligibility of a receiving end is improved.

The embodiments of the present application relate to the technical field of terminals. Provided are a speech processing method, apparatus and system, and a storage medium and a program product. In face-to-face communication scenarios and call scenarios, the speech processing method can be used to restore impaired speech of a speech-impaired user, so as to obtain restored speech, and the restored speech has higher intelligibility than the impaired speech. In this way, the speech-impaired user can also communicate by means of inputting speech. Therefore, the technical problem of the convenience of communication for speech-impaired users being insufficient can be solved, thereby achieving the technical effect of improving the convenience of communication for speech-impaired users.

The invention discloses a complex sound scene target voice selectivity enhancement method and system based on eye movement guidance, and the method comprises the steps: synchronously collecting a gazing direction and a multi-channel audio signal through a miniature eye movement tracking module and an annular quaternary microphone array, and constructing a dynamic sound source positioning mechanism based on eye movement guidance; based on visual confidence score adaptive switching or fusion beam forming and deep learning enhancement strategies, voice intelligibility evaluation and bone conduction tactile feedback closed loops are introduced, and children are actively guided to adjust gazing behaviors. The system comprises an eye movement tracking module, a sound source positioning module, an adaptive enhancement module, a feedback prompt module and the like, and is integrally light, low in delay and high in comfort. According to the invention, through multi-mode fusion and closed-loop interaction, the selective perception ability and communication efficiency of hearing-impaired children to the voice of the target speaker in a complex sound scene are improved.

The present application relates to the technical field of intelligent consumer equipment, image communication, audio analysis processing, intelligent voice, projection display control, energy-saving projection equipment and intelligent thermal management, and particularly relates to a sound and picture thermal cooperative control method, system and equipment for karaoke projection equipment. Specifically, microphone human voice audio signals, accompaniment audio signals, lyrics time stamps, projection picture data, thermal state data and fan speed feedback data are synchronously acquired to determine an acoustically maskable heat dissipation window. When the window is not met and a pre-adjusted heat condition is met, a thermal load transfer mark is generated to reduce the thermal load of a non-lyrics display area and maintain the readability of a lyrics display area. The fan speed increment is released at a subsequent window, and the microphone wind noise is suppressed according to a wind noise template. Based on the wind noise suppression result and the thermal state data, the next time window control quantity is closed-loop distributed. The method can balance the human voice intelligibility, lyrics readability and overall stability of heat dissipation.

Systems and methods for vehicles to communicate with people outside the vehicle. In particular, a two-way communication system is provided including exterior microphone arrays on the vehicle. Calls performed from the exterior of a vehicle can be difficult to implement with acceptable quality in noisy environments. A pipeline having multiple beamformers is provided to improve call quality, where the pipeline operates on the incoming data for each of the microphone arrays. Each beamformer estimates direction of arrival (DOA) of the speech signal, and utilizes the estimated DOA to generate spatial filtering coefficients to filter the signal. A beamformer can be used to generate a one channel output for each microphone array, and a mixer can be used to mix the signal output from each microphone array. The beamformer enables reliable two-way communication with acceptable listening quality and good intelligibility for the remote assistant listening to the captured signal.

The application provides a kind of based on text semantic information fidelity super low code rate speech coding system, it is related to speech coding technical field.The system includes: through multimodal text-speech joint encoder, speech feature extraction and text feature extraction are carried out to original speech, speech feature and text feature are obtained, and text feature is embedded into speech feature, text-speech feature of original speech is obtained, and text-speech feature of original speech is sent to receiving end in semantic communication system;The text-speech feature received by receiving end is input into text semantic fidelity decoder based on attention mechanism, to obtain reconstructed speech feature and reconstructed text feature, and the reconstructed speech feature is decoded layer by layer and attention calculation is carried out with the reconstructed text feature, to obtain decoded speech.Through the speech coding technology of the application, good speech perceptual quality can still be maintained under super low code rate, and the speech intelligibility is guaranteed.

This invention belongs to the field of audio coding technology, specifically disclosing an air traffic control audio coding method and system based on dynamic acoustic masking. The method includes: acquiring air traffic control audio signals and performing complex short-time Fourier transform to generate an audio time-frequency map; inputting the time-frequency map into a personalized auditory feature extraction model to extract physiological feature maps and a dynamic weight matrix, fusing them to generate a final masking matrix; performing a Hadamard product between the masking matrix and the time-frequency map to obtain a perceptual saliency map; extracting audio features based on the perceptual saliency map, constructing a basic codebook and a tree-structured residual codebook for hierarchical vector quantization compression, and outputting a compressed representation; converting the compressed representation into a coded triple consisting of a basic index, a path index, and a termination flag, as the final coding result. This method adapts to individual auditory differences, improves command intelligibility and compression efficiency, and is suitable for high-noise aviation communication scenarios.

The invention discloses a voice communication system intelligibility enhancement method and system based on a binaural de-masking principle, and relates to the technical field of voice signal processing. The system comprises a voice signal input module, a binaural signal generation module, a masking-removing enhancement module, a self-adaptive optimization module and a signal output module. The voice signal input module is used for receiving an original voice communication signal and generating a standard voice communication signal through channel alignment correction; the binaural signal generation module obtains corresponding left and right ear filter coefficients from the HRTF parameter library through a bilinear interpolation algorithm according to the azimuth angle and the elevation angle of the standard voice communication signal, and performs convolution operation on the binaural stereo signal to generate a binaural signal; according to the invention, by simulating a human binaural auditory mechanism and introducing dynamic phase difference and amplitude ratio regulation and control, the binaural de-masking effect is effectively enhanced, and thus the intelligibility and definition of voice signals can be significantly improved in a complex acoustic environment.