15 results about "Linear predictive coding" patented technology

An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first and second encoding branches, the second encoding branch having a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and the second encoding branch having a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder, and a third domain decoder as well as two cascaded switches for switching between the decoders.

The invention discloses a two-stage voice noise reduction method and device based on enhanced spectral subtraction and Kalman filtering, and a storage medium. The method comprises the following steps: performing preliminary noise reduction on a voice signal in a frequency domain through enhanced spectral subtraction; speech signal state space model parameters are identified based on a linear predictive coding algorithm, and secondary noise reduction is performed on the sound signals in a time domain by using a Kalman filtering algorithm; and repeatedly carrying out multiple parameter identification and secondary noise reduction processes on the frame-by-frame signals, and then outputting noise-reduced voice. According to the method, the environmental noise can be effectively filtered in a complex noise environment, and the voice signal quality is improved. The method is suitable for single-microphone noise reduction of the artificial cochlea, efficient noise reduction can be achieved in the artificial cochlea with limited computing resources, and the sound perception ability of a wearer is improved.

Provided are an apparatus for encoding an audio signal and a method of an operation thereof. An audio signal encoding method includes obtaining quantized linear prediction (LP) coefficients by performing a linear predictive coding (LPC) analysis and quantization on an input audio signal, generating a reference signal by applying discrete Fourier transform (DFT) to the input audio signal, obtaining LP residual coefficients from the reference signal, scaling magnitudes of the LP residual coefficients using the quantized LP coefficients and the reference signal, and quantizing phases of the LP residual coefficients and the scaled magnitudes of the LP residual coefficients.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

The invention belongs to the technical field of communication, and particularly relates to an interphone remote communication method and system, and the method comprises the steps: carrying out the framing processing of a collected original voice signal at a source end interphone, and generating a voice frame sequence with a fixed time length; performing forward error correction coding on each voice frame and adding a timestamp mark to form a coded voice packet with a time sequence identifier; injecting the coded voice packet into a multi-hop network formed by a plurality of relay nodes, and executing dynamic path selection and voice packet priority scheduling based on link state prediction at each relay node; according to the system, forward error correction coding voice frames with timestamps are introduced at a source end, dynamic path selection based on link state prediction and voice frame priority preemptive scheduling are implemented in a relay network, and a multi-path redundancy receiving and missing frame interpolation reconstruction mechanism based on linear predictive coding are adopted at a destination end. And transmission time delay and jitter accumulated in the multi-hop forwarding process are effectively eliminated.

The application describes a method and apparatus for decoding voice speech. The method may include receiving encoded data including linear predication coding (LPC) coefficients, an indication of a fixed codebook (FCB), and an indication of an adaptive codebook (ACB). The encoded data may be decoded into a excitation signal. A spectral analysis may performed on the excitation signal. A time envelope of the excitation signal may be applied to a white noise signal and filtered to obtain a complementary noise signal. The complementary noise signal may be summated with the excitation signal to recover a modulated signal. The application also describes a method and apparatus for decoding unvoiced speech.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

The application relates to the technical field of speech recognition, in particular to a speech-based emotion recognition method, which comprises the following steps: according to an input speech signal, frame division and windowing are carried out. The application can obtain more profound insights into emotional expression by decomposing the input speech signal into two physical sources of glottal excitation and vocal tract response for independent modeling and analysis, obtaining a vocal tract transfer function set via linear predictive coding operation, and applying inverse filtering to the original speech signal to reconstruct an approximate glottal pulse sequence, effectively stripping the influence of vocal tract resonance on the signal, so that perturbation parameters, open quotient and closed quotient and the like representing vocal cord vibration patterns can be directly calculated, at the same time, the vocal tract transfer function set is used to identify and track the dynamic trajectory of the formant, and the phase difference cosine mean between adjacent frames is combined to quantify the sound production stability, and subtle dynamic adjustment and control stability of sound production organs such as the oral cavity and tongue position caused by emotional changes are captured.

The present invention discloses a rail transit transformer fault diagnosis method, comprising obtaining noise signals emitted by the rail transit transformer under various operating conditions; using wavelet thresholding for denoising; recursively transferring linear predictive coding to the cepstral domain to obtain linear predictive cepstral coefficients; using fast Fourier transform to obtain a spectrum graph to extract Mel-type cepstral coefficients, and combining the Mel-type cepstral coefficients with first-order difference coefficients and second-order difference coefficients to obtain optimized Mel-type cepstral coefficients; combining the characteristic parameter linear predictive cepstral coefficients and the optimized Mel-type cepstral coefficients to obtain a feature set; further performing feature learning on the feature set using a deep learning architecture to train and establish a rail transit transformer fault identification model; and using the trained rail transit transformer fault identification model to detect rail transit transformer noise. The present invention can quickly, accurately, and effectively identify rail transit transformer faults.

The present invention discloses a two-stage speech noise reduction method based on enhanced spectral subtraction and Kalman filtering, as well as its device and storage medium. The method comprises the following steps: performing preliminary noise reduction on the speech signal in the frequency domain by using enhanced spectral subtraction; identifying the state space model parameters of the speech signal based on a linear predictive coding algorithm, and performing secondary noise reduction on the sound signal in the time domain by using a Kalman filter algorithm; and outputting noise-reduced speech after repeating the parameter identification and secondary noise reduction process on the frame-by-frame signal multiple times. The method of the present invention can effectively filter out environmental noise in a complex noise environment and improve the quality of the speech signal. The method is suitable for single-microphone noise reduction of a cochlear implant and can achieve efficient noise reduction in a cochlear implant with limited computing resources, thereby improving the wearer's sound perception ability.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

Disclosed in embodiments of the present invention are a UWB-based audio transmission method and device, a terminal, and a storage medium. The method comprises: acquiring an audio digital signal, and compressing the digital signal on the basis of linear predictive coding; converting the compressed digital signal into a pulse signal, and controlling a sending end to send the pulse signal in a UWB channel; upon receiving the pulse signal, a receiving end demodulating the pulse signal into a digital signal and decompressing same; and finally performing packet loss detection processing, and compensating for packet loss data, so as to obtain a complete audio digital signal.

The invention discloses a multi-channel sound mixing processing method and system, and belongs to the technical field of multi-channel sound mixing processing, and the method comprises the step of carrying out millisecond-level delay calibration between channels through dynamic phase alignment according to multi-channel original audios. A multi-dimensional frequency deconstruction technology realizes high-precision separation of a basic frequency body, a formant matrix and a transient profile, and extraction of the basic frequency body can independently enhance fundamental frequency energy and improve voice definition or plumpness of musical instrument timbre; the accurate positioning of the formant matrix can simulate the spatial reflection characteristics of different musical instruments; the impact force of the sound is reserved by the capture of the wavelet packet decomposition on the transient event, the self-adaptive window length design is combined with the energy weighted filtering, so that the calculation complexity is effectively reduced, the multi-scene requirement is compatible, the gain or delay of the three types of separated spectrum components can be independently adjusted, and the controllability of audio processing is remarkably improved.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.