33results about How to "Improve hearing quality" patented technology

The invention discloses a prior signal-to-noise ratio estimating method based on an MMSE error criterion and used for voice enhancement, and belongs to the technical field of voice signal processing. Aimed at the prior signal-to-noise ratio estimating problem in the voice enhancement technology, the method comprises the steps of: firstly carrying out preliminary estimation on a prior signal-to-noise ratio of noised voices based on the MMSE error criterion, carrying out Wiener filtering calculation on an obtained prior signal-to-noise ratio estimated value to obtain a first system gain factor, carrying out calculation on the first system gain factor an amplitude spectrum value of the noised voices to obtain a voice power spectrum estimated value, then utilizing the obtained voice power spectrum estimated value and a power spectrum estimated value of noise to carry out estimation once again, and obtaining a final prior signal-to-noise ratio estimated value. The prior signal-to-noise ratio estimated value is substituted into a subsequent voice enhancing step for processing, and de-noised estimated voice clearing signals are obtained. The prior signal-to-noise ratio estimating method based on the MMSE error criterion can effectively inhibit background noise components in estimated cleared voices, and excessive damages to the cleared voice components are avoided, so that the hearing quality of the estimated cleared voices is improved, and the performance of a voice enhancement algorithm is improved.

Push button insertion tool is a device used to help clean and install a receiver-in-the-canal hearing instrument into the ear. This product may be in the shape of a tapered tube, resembling a writing pen. The far end of the tube may be shaped to fit receivers of various sizes, and may include a slot on the side through which the microphone wire may extend. On the other end of the tube may be a push button. When pushed, it moves a rod through the tube to a push plunger, which inserts the receiver out of the tube and into the ear canal. A return spring and guide pin on the far end is designed to retract the push rod back to starting position.

The present invention discloses a digital audio camouflage and reconstruction method based on segmented sequences. The method comprises the steps of taking sub segment sequences obtained by dividing a published audio and a secret audio as a secret sequence and a published sequence, adding random perturbations to the sequences with the same element value, directly carrying out least squares matching on an equidistant transformation sequence obtained by the rotate right of the secret sequence and a corresponding position published sequence, finding a rotate right step length with a minimum residual error and a matched parameter, and thus camouflaging the secret audio as the published audio and further reconstructing the camouflaged audio as the secret audio. The method is easy to realize, since only the corresponding position equidistant transformation sequence matching is carried out, while the encoding time is reduced, the equidistant transformation number is raised, the matching precision is improved, the detail loss and overflow problems brought by the deformation from a smooth block to a complex block and the restoration from the smooth block to a complex texture block can be effectively avoided, and thus the auditory quality of camouflaged and reconstructed audios is improved further.

The invention provides an encoding method of a multi-sound-track signal and an encoder. The encoding method comprises the steps of obtaining the multi-sound-track signal of a current frame; determining an initial ITD value of the current frame; controlling the quantity of target frames permitted to continuously occur according to characteristic information of the multi-sound-track signal, whereinthe characteristic information comprises at least one of signal-to-noise ratio parameters of the multi-sound-track signal and peak features of the mutual relation number of the multi-sound-track signal, and the ITD values of the target frames reuse the ITD value of the previous frames of the target frames; determining the ITD value of the current value according to the ITD value of the current frame and the quantity of the target frames permitted to continuously occur; encoding the multi-sound-track signal according to the ITD value of the current frame. The encoding quality of the multi-sound-track signal can be improved.

A prediction method and a coding/decoding device for a high frequency band signal. The method comprises: acquiring a signal type of an audio signal and a low frequency band signal (100); the audio signal comprises the low frequency band signal and a high frequency band signal; acquiring a frequency domain envelope of the high frequency band signal on the basis of the signal type (101); predicting an excitation signal of the high frequency band signal on the basis of the low frequency band signal (102); and, restoring the high frequency band signal on the basis of the frequency domain envelope of the high frequency band signal and of the excitation signal of the high frequency band signal. The method and device allow for effective reduction in errors found between the high frequency band signal acquired by prediction and an actual high frequency band signal, and increases the accuracy of the predicted high frequency band signal.

The application discloses a quantification noise reducing method and device. After an original digital signal is subjected to framing operation, digital signal sampling points with a maximum amplitude are found in two continuous frames of signals which are respectively a current frame and a next frame, and maximum object digital gain which prevents the maximum amplitude from overflowing is obtained via calculation; object sampling points having minimum own energy and adjacent energy in the current frame are used as gain switching points; according to a gain switching position and a gain switching step number, original digital gain is gradually switched to object digital gain at the gain switching points, and analog gain is correspondingly adjusted. According to the quantification noise reducing method and device, a plurality of object sampling points with the maximum energy are used as the gain switching points; thus, compared with technologies of the prior art, the quantification noise reducing method and device effectively optimize selection of the gain switching points, the gain switching points are enabled to have the minimum adjacent energy, switching noise is lowered to the minimum, and therefore quantification noise is lowered to the minimum and audio quality of signals can be improved.

The invention provides a bone conduction speech enhancement method based on joint dictionary learning and sparse representation. In a training stage, in an indoor noise-free environment, a special-shaped double-microphone array composed of a bone conduction microphone and an air conduction microphone is used for synchronously collecting training speech, and a joint training set of the bone conduction speech and the air conduction speech is constructed; and short-time inverse Fourier transform is performed on the training signals of the bone conduction speech and the air conduction speech to obtain a time-frequency spectrum amplitude, and a joint speech dictionary of the bone conduction speech and the air conduction speech is learnt on a time-frequency spectrum. In a detection stage, short-time Fourier transform is performed on the bone conduction speech to obtain a time-frequency spectrum amplitude and a phase; the amplitude is projected on a bone conduction speech sub-dictionary of the joint speech dictionary; the air-guided speech time-frequency spectrum amplitude is reconstructed by using the obtained sparse representation coefficient and the air-guided speech sub-dictionary ofthe joint speech dictionary, two methods are provided for enhancing the time-frequency spectrum of the bone conduction speech, and finally short-time inverse Fourier transform is performed to obtain an enhanced bone conduction speech time-domain signal, so that the speech sharpness is improved.

A hearing aid and a method performed by a hearing aid including at least one microphone unit, an accelerometer unit, and a processor. The method performed by the hearing aid includes generating a processed microphone signal including attenuating the microphone signal in accordance with an attenuation value; wherein the attenuation value is based on a measure of correlation between a microphone signal from the microphone unit and an acceleration signal from the accelerometer unit. The hearing aid instantly reduces noise arising from mechanical handling of the hearing aid based on input from the microphone unit and the accelerometer unit.

A predicting method and apparatus for frequency domain pulse decoding and a decoder are provided. The method includes: dividing current frame and previous frame into spectral blocks according to spectral coefficient of the previous frame (11), judging whether the prediction between frames for the spectral block divided in the current frame is needed according to the correlation between the spectral blocks divided in the current frame and previous frame (12), for the spectral block of the current frame which is judged that the prediction between frames is needed, predicting the spectral coefficient which is not decoded of the spectral block of the current frame by using the decoded spectral coefficient in the spectral block corresponding to the previous frame and the decoded spectral coefficient in the current frame (13)..

The invention provides a speech enhancement system based on a time modeling generative adversarial network, which belongs to the technical field of speech signal processing and comprises a data acquisition unit used for acquiring noisy speech signals and downsampling the noisy speech signals; and the signal enhancement unit is used for inputting the noisy voice signal into a generative adversarial network based on time modeling, compressing and extracting a global time domain feature of the voice signal, linking the time domain feature and random noise into a feature vector, and decoding the feature vector to obtain an enhanced voice signal. According to the method, the problem that time dependence and global consideration of voice time domain features are insufficient is solved, the noise influence in voice signals is reduced, and therefore the auditory quality of enhanced voice is improved.