Patents
Literature
Patsnap Eureka AI that helps you search prior art, draft patents, and assess FTO risks, powered by patent and scientific literature data.

4 results about "Speech enhancement algorithm" patented technology

Near-end speech intelligibility enhancement with minimal artifacts

ActiveUS12614559B2Speech analysisIntelligibility (communication)Noise
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.
Owner:RTX AS CO

A single microphone long distance sound pickup method and device

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.
Owner:SHANGHAI WEIJING SEMICONDUCTOR CO LTD

Robustness / performance improvements for deep learning-based speech enhancement against artifacts and distortion.

ActiveJP7863597B2Speech analysisNeural learning methodsAcousticsSpeech enhancement algorithm
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
Owner:DOLBY LABORATORIES LICENSING CORP

A two-branch speech enhancement algorithm based on structured state-space sequential model

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.
Owner:NANJING UNIV OF POSTS & TELECOMM