7 results about "Speech enhancement algorithm" patented technology

The invention discloses a distributed reconnaissance tool, relates to the technical field of voice interception, and aims to solve the problems of limited multi-node access capability, poor anti-interference performance and insufficient multi-target interception continuity of an existing reconnaissance system. The tool comprises a micro audio node network and a base station unit, a recording module, a wireless transmission module and an encryption module are arranged in the miniature audio node, local recording and remote transmission are supported, and the miniature audio node has the characteristics of water resistance and low power consumption; the base station unit adopts an orthogonal graph division multiple access technology, at most 32 micro audio nodes can be accessed, four paths of voice can be monitored in real time, the audio quality is optimized through a spectral subtraction voice enhancement algorithm, and WIFI / Ethernet connection and time synchronization and combination of multi-node recording files are also supported. According to the invention, approaching interception and multi-target centralized deployment and control of the moving target are realized, the stability, expansibility and practicability of the interception system are improved, and the method is suitable for scenes such as public security technical investigation and the like.

The invention provides a tinnitus patient intelligent social assistance system and method based on multi-modal fusion, and relates to the technical field of medical assistance, and the method comprises the steps: collecting environment sound signals, voice signals and image data, and carrying out the primary processing; a personalized adaptive noise reduction algorithm and a voice enhancement algorithm are formulated, and then an enhanced voice signal is obtained; constructing a lip language recognition model, and performing conversion correction processing on the enhanced voice signal; constructing a tinnitus feature database, and screening tinnitus masking sound effects from the tinnitus feature database according to the individual tinnitus parameters; obtaining an association relationship between tinnitus and environmental factors during social contact of the tinnitus patient, and providing personalized prevention suggestions according to the association relationship and the tinnitus masking sound effect; and uploading the corresponding enhanced voice signal and the preliminarily processed image data to a cloud server so as to optimize a personalized adaptive noise reduction algorithm and a voice enhancement algorithm.

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 application discloses a speech enhancement method, comprising: acquiring audio information collected by a robot; generating a speech presence probability of speech included in the audio information according to the audio information; and enhancing the audio information according to the speech presence probability to generate enhanced speech. According to the embodiment of the application, the speech presence probability of speech is generated according to the audio information, and the audio information can be enhanced according to the speech presence probability when the speech is enhanced, so that the enhancement effect is improved. In the embodiment of the application, the precision of the speech enhancement algorithm can be improved through the speech presence probability, so that the speech enhancement effect is realized.

The application discloses a single microphone long-distance sound pickup method and device, which comprises the following steps: determining a microphone bias voltage according to the microphone type, circuit and characteristics of the single microphone to be used, directly picking up sound from the sound source by the single microphone to obtain a to-be-processed audio signal; automatically adjusting the overall level of the to-be-processed audio signal by using an automatic level control algorithm; converting the to-be-processed audio signal after the level adjustment into a digital signal; sequentially performing noise reduction, speech enhancement and gain adjustment processing on the digital signal by using a speech enhancement algorithm and an automatic gain control algorithm, that is, using a method combining an analog domain and a digital domain to improve the sound pickup distance; and using an automatic level control (ALC) technology to optimize the quality of the audio signal collected by the single microphone.

To provide a method for processing audio signals, as well as a corresponding device, a computer program, and computer-readable storage media.SOLUTION: A method 1000 includes the steps of: applying emphasis to a first component of an audio signal and / or applying suppression to a second component of the audio signal with respect to the first component S1010; and modifying an output of the step S1010 by applying a deep learning based model to the output to remove artifacts and / or distortions introduced into the audio signal in the step S1010, and perceptually improving the first component of the audio signal S1020.SELECTED DRAWING: Figure 10

The application discloses a double-branch speech enhancement algorithm based on a structured state space sequence model, which comprises the following steps: obtaining amplitude spectrum and complex spectrum features of noisy speech, and inputting the features into an amplitude rough estimation branch and a complex refinement estimation branch respectively to obtain real and imaginary components of the rough estimated speech and the refined speech; introducing an interaction module to realize the flow of the amplitude spectrum and the complex spectrum features between the two branches; superimposing the real and imaginary components of the rough estimated speech and the refined speech to reconstruct a target signal complex spectrum; and evaluating the performance of the double-branch enhancement algorithm based on the structured state space sequence model. The amplitude spectrum and the complex spectrum are estimated simultaneously, and the interaction module is introduced to promote information exchange, so that the features learned from one branch can supplement the missing information of the other branch; and a diagonalized state space model is used to model the speech feature sequence, so that the parameter quantity of the model is reduced, and the algorithm performance is improved.