5 results about "Linear predictive coding" patented technology

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.

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.

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 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.