摘要:
A circuit for applying a predetermined algorithm to an input signal, comprises an input for receiving the input signal, a signal processing device for processing the input signal in accordance with the predetermined algorithm, and a device for outputting the processed signal, the signal processing device comprising distributed bit-serial logic circuits to implement the predetermined algorithm.
摘要:
A digital audio signal processing apparatus comprising a predictive error generator means (2) for generating predictive error data by processing input digital data to acquire a plurality of different frequency characteristic; a selector means (13) for selecting one of the plural predictive error data; a requantizer means (16) for requantizing the selected predictive error data; a corrector means for processing (18), with a predetermined frequency characteristic, the requantization error induced during the operation of the requantizer means, thereby correcting the requantization error caused in the requantizer means; and a frequency characteristic control means (12) for selecting at least two of the predictive error data obtained with the plural frequency characteristics, then calculating the selected predictive error data and controlling the frequency characteristic in the corrector means in accordance with the result of such calculation. In this apparatus, the ratio or the difference between at least two predictive error data obtained with a plurality of frequency characteristics is calculated and then is compared with a predetermined reference value. And the frequency characteristic in the corrector means is controlled in conformity with the numerical relation between the calculated value and the reference value. Therefore two or more frequency characteristics in the corrector means are selectively rendered conformable with one frequency characteristic in the predictive error generator means, hence achieving an enhanced effect of further improving the signal-to-noise ratio.
摘要:
Bei bekannten Systemen müssen im Differenzpulscodemodulator an einer Stelle in einer zeitkritischen Schleife eine Anzahl Verarbeitungsschritte innerhalb eines Abtasttaktes durchgeführt werden, nämlich eine Summenbildung, eine Multiplikation mit Addition, eine Differenzbildung und eine Quantisierung. Für den insbesondere bei der Bildverarbeitung fast immer auftretenden Fall, daß zwischen den für den Vorhersagewert verwendeten Signalen eine Anzahl Signale liegt, die nicht für den Vorhersagewert verwendet werden, wird eine Lösung angegeben, bei der innerhalb der zeitkritischen Schleife nur eine Zuordnung und eine Addition erforderlich ist. Eine solche Anordnung entspricht einer speziellen Aufteilung der Gleichung für das Vorhersagefehlersignal in einzelne Terme. Die angegebene Lösung läßt sich in verschiedener Weise zur Einsparung von Aufwand abwandeln und auch für adaptive Modulatoren verwenden. Die gleichen Maßnahmen können auch im Demodulator verwendet werden.
摘要:
A signal corresponding to a short-period change and a signal corresponding to a long-period change of a sound signal are detected, and optimal quantization is performed based on the combination, of the two signals. In an ADPCM encoding apparatus (100), a differential value d n between a 16-bit input signal X n and a decoded signal Y n-1 of one sample ago is calculated by a subtractor (102). Thereafter, the 16-bit differential value d n is adaptively quantized by an adaptive quantizing section (103), so as to be converted to a (1 to 8)-bit length-variable ADPCM value D n . Thereafter, the ADPCM value D n is compression-encoded by a compression-encoding section (108) to generate a signal D' n , and the signal D' n is framed by a framing section (130) and outputted. Further, in an ADPCM decoding apparatus, a framed input signal is subjected to a reverse of the aforesaid process so as to be decoded.
摘要:
A device divides a sampled sound signal into a high frequency signal and a low frequency signal, individually encodes the high frequency signal and the low frequency signal, and generates error detection code pertinent to the high frequency ADPCM data and the low frequency ADPCM data. The device replaces data pertinent to some of a plurality of bits which configure the low frequency ADPCM data with the error detection code and transmits them. A receiver side receives the high frequency ADPCM data and the low frequency ADPCM data, and individually processes the high frequency ADPCM data and the low frequency ADPCM data in accordance with a value of the error detection code.
摘要:
A decoder and method of decoding a sub-band coded digital audio signal. The decoder comprises: an input, for receiving sub-band coefficients for a plurality of sub-bands of the audio signal; an error detection unit (20), adapted to analyze the content of a sequence of coefficients in one of the sub-bands, to derive for each coefficient an indication of whether the coefficient has been corrupted by an error of a predefined type; an error masking unit (30), adapted to generate from the sequence a modified sequence of coefficients for the sub-band, wherein errors of the predefined type are attenuated; a coefficient combination unit (40), adapted to combine the received coefficients and the modified coefficients, in dependence upon the indication of error; and a signal reconstruction unit (50), adapted to reconstruct the audio signal using the combined coefficients.
摘要:
The invention relates to a method for high-resolution, wave-form maintaining digitization of analog signals. The usual scalar logarithmic quantization is transferred to multidimensional spherical coordinates whereby the resulting advantages such as a constant signal-to-noise ratio arise over an extremely high dynamic range with very little loss in relation to the distortion-theory rate. In order to use the statistic dependencies available in the source signal for another gain in the signal-to-noise ratio, the differential pulse code modulation (DPCM) is combined with spherically logarithmic quantization. The resulting method makes it possible to reduce data in an effective manner with a high long-term distance for an extremely small signal delay.
摘要:
The invention relates to a method for high-resolution, wave-form maintaining digitization of analog signals. The usual scalar logarithmic quantization is transferred to multidimensional spherical coordinates whereby the resulting advantages such as a constant signal-to-noise ratio arise over an extremely high dynamic range with very little loss in relation to the distortion-theory rate. In order to use the statistic dependencies available in the source signal for another gain in the signal-to-noise ratio, the differential pulse code modulation (DPCM) is combined with spherically logarithmic quantization. The resulting method makes it possible to reduce data in an effective manner with a high long-term distance for an extremely small signal delay.
摘要:
A signal corresponding to a short-period change and a signal corresponding to a long-period change of a sound signal are detected, and optimal quantization is performed based on the combination, of the two signals. In an ADPCM encoding apparatus (100), a differential value d n between a 16-bit input signal X n and a decoded signal Y n-1 of one sample ago is calculated by a subtractor (102). Thereafter, the 16-bit differential value d n is adaptively quantized by an adaptive quantizing section (103), so as to be converted to a (1 to 8)-bit length-variable ADPCM value D n . Thereafter, the ADPCM value D n is compression-encoded by a compression-encoding section (108) to generate a signal D' n , and the signal D' n is framed by a framing section (130) and outputted. Further, in an ADPCM decoding apparatus, a framed input signal is subjected to a reverse of the aforesaid process so as to be decoded.