会员体验
专利管家(专利管理)
工作空间(专利管理)
风险监控(情报监控)
数据分析(专利分析)
侵权分析(诉讼无效)
联系我们
交流群
官方交流:
QQ群: 891211   
微信请扫码    >>>
现在联系顾问~
热词
    • 3. 发明公开
    • Switched adaptive predictive coding with noise shaping
    • 开关自适应预测编码与噪声形成
    • EP0481707A3
    • 1992-08-05
    • EP91309444.7
    • 1991-10-15
    • SONY CORPORATION
    • Mitsuhashi, SatoshiNishiguchi, Masayuki
    • H03M7/30
    • H03M7/3046
    • 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.
    • 4. 发明公开
    • System zum Übertragen oder Speichern von Eingangssignalen
    • 系统,用于传输或存储的输入信号。
    • EP0264999A2
    • 1988-04-27
    • EP87201914.6
    • 1987-10-07
    • Philips Patentverwaltung GmbHPhilips Electronics N.V.
    • Rosebrock, Jens
    • H03M3/04
    • H03M7/3044H03M7/3046
    • 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.
    • 在已知的系统中,在时间关键的环的位置处差分脉冲编码调制器,一个数量的处理步骤必须在单个样品,即求和,乘法与加法,减法和量化内进行。 用于特定几乎总是发生在位于用于预测值的材料之间的图像处理的情况下信号的多个不用于预测值的信号,溶液被指定时,在此期间,时间关键的环路内只有一个映射和加法所需 是。 这种布置对应于以下方程的预测误差信号成各个术语的特定划分。 给定的溶液可以以各种方式,以节省成本并使用自适应调制器进行修改。 同样的措施也可以在解调器使用。
    • 7. 发明公开
    • Error concealment for sub-band coded audio signals
    • FehlerverdeckungfürSubband-codierte Audiosignale
    • EP2458585A1
    • 2012-05-30
    • EP10192870.3
    • 2010-11-29
    • NXP B.V.
    • Macours, Christophe Marc
    • G10L19/00
    • G10L19/005G10L19/0204H03M7/3046H04L1/0072
    • 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.
    • 解码器和解码子带编码数字音频信号的方法。 解码器包括:用于接收音频信号的多个子带的子带系数的输入; 一个错误检测单元(20),用于分析一个子带中的一系列系数的内容,为每个系数导出该系数是否已被预定义类型的错误破坏的指示; 一个误差屏蔽单元,适于从该序列产生经修改的子带序列,其中预定类型的误差被衰减; 系数组合单元(40),其适于根据误差的指示组合所接收的系数和修改的系数; 以及信号重建单元(50),其适于使用所述组合系数来重构所述音频信号。
    • 10. 发明公开
    • ADAPTIVE DIFFERENTIAL PULSE CODE MODULATION ENCODING APPARATUS AND DECODING APPARATUS
    • 反刍动物解剖异常ÜATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION ATION
    • EP2383730A1
    • 2011-11-02
    • EP09834996.2
    • 2009-12-25
    • Kyushu Institute of Technology
    • SATO YasushiRYU Atsuko
    • G10L19/00
    • G10L19/04H03M7/3046
    • 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.
    • 检测对应于短周期变化的信号和对应于声音信号的长周期变化的信号,并且基于两个信号的组合来执行最佳量化。 在ADPCM编码装置(100)中,通过减法器(102)计算16位输入信号X n与一个采样前的解码信号Y n-1之间的差分值d n。 此后,通过自适应量化部分(103)自适应地量化16位微分值d n,以便转换成(1至8)位长度可变ADPCM值D n。 此后,ADPCM值D n由压缩编码部(108)进行压缩编码,生成信号D'n,信号D'n由成帧部(130)构成并被输出。 此外,在ADPCM解码装置中,对成帧的输入信号进行与上述处理相反的处理,以进行解码。