摘要:
On procède à l'estimation de valeurs de hauteur de résolution à sous-nombre entier lors de l'estimation initiale de la hauteur, les valeurs de hauteur à sous-nombre entier étant de préférence estimées par interpolation de variables intermédiaires entre des valeurs de nombre entier. On utilise des régions de hauteur afin de réduire la quantité de calculs nécessaires à l'estimation initiale de la hauteur. On utilise une résolution dépendante de la hauteur lors de l'estimation initiale de ladite hauteur, une résolution supérieure étant utilisée pour des valeurs plus petites de hauteur.
摘要:
A device for signal processing includes a memory and a processor. The memory is configured to store a parameter associated with a bandwidth-extended audio stream. The processor is configured to select a plurality of non-linear processing functions based at least in part on a value of the parameter. The processor is also configured to generate a high-band excitation signal based on the plurality of non-linear processing functions.
摘要:
The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.
摘要:
A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook (202) used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described.
摘要:
This invention relates to an audio decoder for providing a decoded audio information on the basis of an encoded audio information comprising linear prediction coefficients (LPC), a respective method, a respective computer program for performing such a method and an audio signal for a storage medium having stored such an audio signal, the audio signal having been treated with such a method. The audio decoder comprises a tilt adjuster configured to adjust a tilt of a noise using linear prediction coefficients of a current frame to obtain a tilt information and a noise inserter configured to add the noise to the current frame in dependence on the tilt information obtained by the tilt calculator. Another audio decoder according to the invention comprises a noise level estimator configured to estimate a noise level for a current frame using a linear prediction coefficient of at least one previous frame to obtain a noise level information; and a noise inserter configured to add a noise to the current frame in dependence on the noise level information provided by the noise level estimator. Thus, side information about a background noise in the bitstream may be omitted.
摘要:
In a method of smoothing stationary background noise in a telecommunication speech session, initially receiving and decoding S10 a signal representative of a speech session, where the signal comprises both a speech component and a background noise component. Subsequently, providing S20 a noisiness measure for the signal, and adaptively S30 smoothing the background noise component based on the provided noisiness measure.
摘要:
In a method of smoothing stationary background noise in a telecommunication speech session, initially receiving and decoding S10 a signal representative of a speech session, where the signal comprises both a speech component and a background noise component. Subsequently, providing S20 a noisiness measure for the signal, and adaptively S30 smoothing the background noise component based on the provided noisiness measure.
摘要:
In a speech coding apparatus (111), a band-pass filter unit (133) separates a residual signal generated by a predictive analyzer (131) into band by band components. Then, a gain calculation unit (135) and a voiced/unvoiced discrimination and pitch extraction unit (137) acquire information on an intensity characterizing each band, information on a result of discrimination as to whether each band-by-band component is a voiced sound or unvoiced sound, and information on a pitch frequency when it is a voiced sound. The acquired information is coded together with a predictive coefficient and is transmitted to a speech decoding apparatus (211). The speech decoding apparatus (211) generates an excitation signal while reflecting the feature of each band of the original residual signal. This makes the excitation signal an efficient replica of the original residual signal.
摘要:
Vector quantization techniques reduce the effective bit rate to 600 bps while maintaining intelligible speech. Four frames of speech are combined into one frame. The system uses mixed excitation linear prediction speech model parameters to quantized the frame and achieve a fixed rate of 600 bps. The system allows voice communication over bandwidth constrained channels.