40 results about "Perceptual weighting" patented technology

A perceptual weighting device for producing a perceptually weighted signal in response to a wideband signal comprises a signal pre-emphasis filter, a synthesis filter calculator, and a perceptual weighting filter. The signal pre-emphasis filter enhances the high frequency content of the wideband signal to thereby produce a pre-emphasized signal. The signal pre-emphasis filter has a transfer function of the form: P(z)=1−μz⁻¹, wherein μ is a pre-emphasis factor having a value located between 0 and 1. The synthesis filter calculator is responsive to the pre-emphasized signal for producing synthesis filter coefficients. Finally, the perceptual weighting filter processes the pre-emphasized signal in relation to the synthesis filter coefficients to produce the perceptually weighted signal. The perceptual weighting filter has a transfer function, with fixed denominator, of the form: W(z)=A(z/γ₁)/(1−γ₂z⁻¹) where 0<γ₂<γ₁≦1.

A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution. The method includes pre-computing weighting factors for a perceptual weighting filter optimized to a specific source and destination codec pair, pre-configuring the transcoding strategies, mapping CELP parameters in the CELP parameter space according to the selected coding strategy, performing Linear Prediction analysis if specified by the transcoding strategy, perceptually weighting the speech using with tuned weighting factors, and searching for adaptive codebook and fixed-codebook parameters to obtain a quantized set of destination codec parameters.

The invention relates to the coding/decoding of a signal into several sub-bands, in which at least a first and a second sub-bands which are adjacent are transform coded (601, 602). In particular, in order to apply a perceptual weighting, in the transformed domain, to at least the second sub-band, the method comprises:—determining at least one frequency masking threshold (606) to be applied on the second sub-band; and normalizing said masking threshold in order to provide a spectral continuity between the above-mentioned first and second sub-bands. An advantageous application of the invention involves a perceptual weighting of the high-frequency band in the TDAC transform coding of a hierarchical encoder according to standard G.729.1.

The invention relates to a coding method and a decoding method for ultra wide band expansion, a coder and a decoder as well as a system for the ultra wide band expansion, wherein the coding method and the coder obtain a coding parameter of a reconstructed 7-8 kHz signal through wide band enhancement, perform perception weighting on the base, obtain a frequency band expansion parameter of a reconstructed 8-14 kHz high frequency signal according to a principle of the smallest mean square error; and a decoding end reconstructs a 7-14 kHz high frequency signal through coding transmission so as to achieve the expansion of a 7-14 kHz ultra wide band. The decoding method and the decoder decode a residual error MDCT coding parameter to obtain a 7-8 kHz residual error MDCT recovery coefficient, perform perception weighting on the base, and reconstruct a 8-14 kHz signal according to the principle of the smallest mean square error so as to achieve the reconstruction of the 7-14 kHz high frequency signal and obtain a 0-14 kHz ultra wide band signal.

A perceptually weighted speech coder system samples a speech signal and determines its pitch. The speech signal is characterized as fully voiced, partially voiced or weakly voiced. A Lloyd-Max quantizer is trained with the pitch values of those speech signals characterized as being substantially fully voiced. The quantizer quantizes the trained fully voiced pitch values and the pitch values of the non-fully voiced speech signals. The quantizer can also quantize gain values in a similar manner. Sampling is increased for fully-voice signals to improve coding accuracy. This limits application to non-real time speech storage. Mixed excitation is used to synthesize the speech signal

Layered code-excited linear prediction (CELP) speech encoders have progressively weaker perceptual weighting filters for each of the successive enhancement layers and decoders have progressively weaker short-term postfilters for increased bit rates (increased number of enhancement layers decoded) and a long-term postfilter for all bit rates.

The invention relates to a voice pitch period estimation method and device. The device comprises a signal preprocessing unit, a normalized autocorrelation function computing element and a pitch period postprocessing unit. The method includes the steps of firstly, conducting preprocessing including direct current component removal, perception weighting and under-signal sampling on voice signals; secondly, computing normalized autocorrelation function values of the processed voice signals; thirdly, determining the maximum of the normalized autocorrelation function values in the pitch period searching range, and determining a pitch period candidate value corresponding to the maximum to be a pitch period estimation value of the voice signals. According to the voice pitch period estimation method and device, frequency doubling errors and frequency halving errors in the pitch period estimation are well overcome, the noise resistance performance of the pitch period estimation method is improved, meanwhile, the algorithm complexity of an algorithm is lowered, and the corresponding digital audio/speech coding efficiency is improved. The voice pitch period estimation method and device can be applied to pitch searching of various voice coding and decoding algorithms and have a wide application range.

Disclosed is an apparatus for processing an input signal, having a perceptual weighter and a quantizer. The perceptual weighter has a model provider and a model applicator. The model provider provides a perceptual weighted model based on the input signal. The model applicator provides a perceptually weighted spectrum by applying the perceptual weighted model to a spectrum based on the input signal. The quantizer is configured to quantize the perceptually weighted spectrum and for providing a bitstream. The quantizer has a random matrix applicator and a sign function calculator. The random matrix applicator is configured for applying a random matrix to the perceptually weighted spectrum in order to provide a transformed spectrum. The sign function calculator is configured for calculating a sign function of components of the transformed spectrum in order to provide the bitstream. The invention further refers to an apparatus for processing an encoded signal and to corresponding methods.

The invention discloses a method for perceptual weighted filtering, which comprises: A. to make a spectrum slant filtering processing to the inputted speech or audio signal. B. to make a traditional perceptual weighted filtering on the output speech or audio signal processed by the spectrum slant filtering. C. to output the speech or audio signal processed by the traditional perceptual weighted filtering and takes the partial coefficient of the coefficient in transfer function used in the perceptual weighted filtering as the coefficient in transfer function used in the said perceptual weighted filtering. Besides, the invention also discloses a perceptual weighted filter designed according to the said method for perceptual weighted filtering. The invention is to achieve the purpose of simulating the formant structure of the inputted speech or audio signal spectrum of the broadband to improve the coding efficiency and enhance the subjective auditory effect by taking full advantages of the human auditory masking effect.

A transcoder for use between speech codecs using different Code-Excited Linear Prediction (CELP) type and a method therefor are disclosed. The transcoder includes a decoding unit of an input CELP codec, a transcoding filter, a transcoding filter design unit, and an encoding unit of an output CELP codec. By substituting a post-filter and a perceptual weighting filter of a prior art with one transcoding filter, the calculation amount of the transcoder is reduced, and speech quality decoded at a receiving end is improved.

The present invention discloses a perceptual weighing filtering method. The method mainly comprises treating the input voice or audio signals through spectrum inclined filtering, processing the output voice or audio signals treated through perceptual weighing filtering, selecting the corresponding weighing factor value according to the spectral flatness of the input signals, and outputting the voice or audio signals processed through the traditional perceptual weighing filtering. Moreover, part of the coefficients of the transmission function acquired in the processing of perceptual weighing filtering are directly used as the coefficients of the transmission function acquired in the processing of spectrum inclined filtering. Simultaneously, the present invention also provides a perceptual weighing filter. The perceptual weighing filter can well simulate the formant structure of the frequency spectrum of the input wideband voice or audio signals without additional encoding bit rate and reducing the computation complexity, make full use of he masking effects of human acoustic feelings, and achieve the purposes of improving the encoding efficiency and enhancing the subjective hearing effects.

The invention discloses an audio output method and device. The audio output method includes: determining a masking effect curve of frequency domain noise signals, performing perceptual weighting on energy information of source audio signals to acquire energy information after weighting, performing time domain to frequency domain conversion on the source audio signals to acquire frequency domain source audio signals, determining that the energy information after weighting is energy information corresponding to the frequency domain source audio signals, enhancing or weakening the frequency domain source audio signals according to the masking effect curve and the energy information corresponding to the frequency domain source audio signals, performing frequency domain to time domain conversion on the frequency domain source audio signals after being enhanced or weakened to acquire source audio signals after processing, reversely weighting the source audio signals after processing, and outputting audio signals. By combining and analyzing masking effect and an audio weighting mode, interference resistance of audio output equipment is improved without excessively changing output audio energy.

Primary and alternate optimization procedures are used to improve the ITU-T G.723.1 speech coding standard (the “Standard”) by replacing the Hamming window of the Standard with an optimized window, with two windows, or with two windows and an additional performance of an autocorrelation method. When two windows replace the Hamming window, at least one of which is an optimized window, generally the first is used to determine optimized unquantized LP coefficients which are used to define an optimized perceptual weighting filter, and the second is used to determine optimized unquantized LP coefficients which are used to determine optimized synthesis coefficients. Optimized windows created using the primary and alternate optimization procedures and used in the Standard yield improvements in the objective and subjective quality of synthesized speech produced by the Standard. The improved Standard, methods, and window can all be implemented as computer readable software code.

The invention relates to a method for coding a variable speed audio frequency switching between adjacent high/low speed coding modes, which belongs to the field of audio coding and is particularly suitable for a multi-speed audio encoder. The method is technically characterized by comprising the following steps of: coding and decoding each frame of audio signals at a high speed, making coding input signals and decoding output signals in various coding modes at the speed pass through a sensing and weighting filter, calculating an average segmentation signal to noise ratio (SNR), and selecting a coding mode with the maximum sensing and weighting average segmentation SNR; selecting a coding mode with the maximum sensing and weighting average segmentation SNR of the coding input signals and the decoding output signals at a low coding speed close to the high speed; and finally, calculating average segmentation SNR of the coding input signals and the decoding output signals relative to the coding modes selected at the high speed and the low speed respectively, if the average segmentation SNR in the low-speed coding mode is greater than the average segmentation SNR in the high-speed coding mode, switching to the low-speed coding mode, otherwise, switching to the high-speed coding mode. By the method, each frame of audio signals are switched among various coding modes at adjacent high/low coding speeds according to distortion of the output signals relative to the input signals, so that high coding quality is kept, and the average coding speed of the audio signals is reduced simultaneously; therefore, the coding efficiency of the multi-speed audio encoder is improved.

The embodiment of the invention discloses a multi-rate voice frequency encoding method. The method comprises the following steps: computing frequency spectrum of a perceptual weighting filter and a first ratio of the frequency spectrum of two previous layers of synthetic voice which are firstly encoded and then decoded by an input signal for each adjacent lattice point; and encoding index values of the corresponding lattice point of the first ratio into a code stream according to a descending order of the first ratio. The method can help improve the quality of output voice or sound signals when lacking bits during transform domain encoding.