子分类：
| 序号 | 专利名 | 申请号 | 申请日 | 公开（公告）号 | 公开（公告）日 | 发明人 |
|---|---|---|---|---|---|---|
| 61 | 具有相位模糊和相位解抹滤波器的自适应差分脉码调制语音编码系统 | CN02801128.7 | 2002-03-27 | CN1221941C | 2005-10-05 | E·F·吉吉 |
| 在一种语音编码系统中，该系统具有语音编码器以及与所述语音编码器配合使用的语音解码器，语音编码器包括预处理器以及ADPCM(自适应差分脉码调制)编码器，ADPCM编码器具有量化器和步长适配装置，而语音解码器包括具有与ADPCM编码器中类似的步长适配装置的ADPCM解码器、解码器和后处理器。预处理器配置了相位模糊滤波装置，用于对量化器的输入处高和/或快速的能量变化效应进行平滑，而后处理器配置了与所述相位模糊滤波装置相反的滤波装置。 | ||||||
| 62 | 数字线路传送装置 | CN02807611.7 | 2002-02-04 | CN1500322A | 2004-05-26 | 原田良尚 |
| 其目的在于获得一种为能同时实现线路的有效使用与高音质，即使在同一呼叫内切换语音编解码器，通话者也不会产生不适感的数字线路传送装置。在编码器内具有高音质高位速率的第1语音编解码器7、音质虽然不高但位速率低的第2语音编解码器8，基于来自运用监视控制部4的判别了承载线路111的通信量大小的控制信息来进行这些语音编解码器的切换。在信号检测部3内的有音检测部31中检测所输入的语音信号的无音部分，在无音时进行语音编解码器的切换。 | ||||||
| 63 | 具有相位模糊和相位解抹滤波器的自适应差分脉码调制语音编码系统 | CN02801128.7 | 2002-03-27 | CN1461469A | 2003-12-10 | E·F·吉吉 |
| 在一种语音编码系统中，该系统具有语音编码器以及与所述语音编码器配合使用的语音解码器，语音编码器包括预处理器以及ADPCM(自适应差分脉码调制)编码器，ADPCM编码器具有量化器和步长适配装置，而语音解码器包括具有与ADPCM编码器中类似的步长适配装置的ADPCM解码器、解码器和后处理器。预处理器配置了相位模糊滤波装置，用于对量化器的输入处高和/或快速的能量变化效应进行平滑，而后处理器配置了与所述相位模糊滤波装置相反的滤波装置。 | ||||||
| 64 | 音频信号的编码、解码方法及音频传输方法 | CN03120486.4 | 1999-10-12 | CN1440133A | 2003-09-03 | 渕上德彦; 植野昭治; 田中美昭 |
| 本发明公开一种对音频信号进行预测编码时改善压缩率的装置。加法电路1a计算出立体声2声道信号L、R的和信号(L+R)，减法电路1b计算出差信号(L－R)。由差分运算电路11D1、11D2计算出本次与上次的差分Δ(L+R)、Δ(L－R)，由于预测编码电路(15D1、15D2、16D1、16D2)计算出差分Δ(L+R)、Δ(L－R)的多个预测值，计算出多个预测值与差分Δ(L+R)、Δ(L－R)的各预测残差，选择最小的预测残差。 | ||||||
| 65 | 音频信号的编码、解码方法及音频传输方法 | CN03120483.X | 1999-10-12 | CN1440130A | 2003-09-03 | 渕上德彦; 植野昭治; 田中美昭 |
| 本发明公开一种对音频信号进行预测编码时改善压缩率的装置。加法电路1a计算出立体声2声道信号L、R的和信号(L+R)，减法电路1b计算出差信号(L－R)。由差分运算电路11D1、11D2计算出本次与上次的差分Δ(L+R)、Δ(L－R)，由于预测编码电路(15D1、15D2、16D1、16D2)计算出差分Δ(L+R)、Δ(L－R)的多个预测值，计算出多个预测值与差分Δ(L+R)、Δ(L－R)的各预测残差，选择最小的预测残差。 | ||||||
| 66 | 音频信号的编码、解码方法及音频传输方法 | CN03120482.1 | 1999-10-12 | CN1440129A | 2003-09-03 | 渕上德彦; 植野昭治; 田中美昭 |
| 本发明公开一种对音频信号进行预测编码时改善压缩率的装置。加法电路1a计算出立体声2声道信号L、R的和信号(L+R)，减法电路1b计算出差信号(L－R)。由差分运算电路11D1、11D2计算出本次与上次的差分Δ(L+R)、Δ(L－R)，由于预测编码电路(15D1、15D2、16D1、16D2)计算出差分Δ(L+R)、Δ(L－R)的多个预测值，计算出多个预测值与差分Δ(L+R)、Δ(L－R)的各预测残差，选择最小的预测残差。 | ||||||
| 67 | 音频信号的编码、解码方法及音频传输方法 | CN03120480.5 | 1999-10-12 | CN1440127A | 2003-09-03 | 渕上德彦; 植野昭治; 田中美昭 |
| 本发明公开一种对音频信号进行预测编码时改善压缩率的装置。加法电路1a计算出立体声2声道信号L、R的和信号(L+R)，减法电路1b计算出差信号(L－R)。由差分运算电路11D1、11D2计算出本次与上次的差分Δ(L+R)、Δ(L－R)，由于预测编码电路(15D1、15D2、16D1、16D2)计算出差分Δ(L+R)、Δ(L－R)的多个预测值，计算出多个预测值与差分Δ(L+R)、Δ(L－R)的各预测残差，选择最小的预测残差。 | ||||||
| 68 | 将语音信号压缩成可变速率数据的方法、设备和电路 | CN92104618.9 | 1992-06-11 | CN1091535C | 2002-09-25 | 保罗·E·雅各布; 威廉·R·加德纳; 冲·U·李; 克莱恩·S·吉豪森; S·凯瑟琳·兰姆; 民昌·蔡 |
| 一种通过对数字化语音取样的帧作可变速率编码而将语音信号压缩的装置和方法。对于数字化语音取样的每个帧确定语音动作级别，依据所确定的帧语音动作级别，从一组速率中选出一个输出数据包速率。该组速率中的最低速率对应于检测到的最低语音动作级别，例如对应于语音中的背景噪声或停顿，而最高速率对应于检测到的最大语音动作级别，如对应于现有发音。每个帧根据选中速率的预定编码格式进行编码，各个速率具有代表编码帧的相应位数。对每个编码帧形成一个数据包，各输出数据包的位速率对应于选中的速率。 | ||||||
| 69 | 具有改进的丢失部分重构功能的传输系统 | CN98801792.X | 1998-08-17 | CN1243621A | 2000-02-02 | J·拉佩利 |
| 在传输系统中,要由第一节点(2)发送的信号被施加到编码器(8)并由发送器(10)发射到第二节点(6)。在该第二节点(6)中,来自传输介质(4)的信号被接收器(12)所接收,并直接通过内插器(14)传递给选择器(18)。在发生导致帧丢失的传输错误的情况下,通过延迟信号的解码,有可能用内插法完成丢失的帧。 | ||||||
| 70 | 音频信号编码装置和译码装置以及音频信号编码和译码方法 | CN98800527.1 | 1998-05-14 | CN1224523A | 1999-07-28 | 津岛峰生; 则松武志; 石川智一 |
| 一种音频信号编码装置和译码装置,提高在译码装置中即使不使用编码装置的全部信息也可以进行音频信号的再生的编码装置和与其对应的译码装置。将构成编码装置1的量化部采用具有由低频、中频、高频的小量化部构成的第1小量化部501、进而将第1小量化部501的量化误差进行量化处理的第2小量化部502和进而将由第1小量化部501和第2小量化部502进行处理后的量化误差进行量化处理的第3小量化部。 | ||||||
| 71 | 可变速率声码器 | CN96123201.3 | 1992-06-11 | CN1167309A | 1997-12-10 | 保罗E·雅各布; 威廉·R·加德纳; 冲·U·李; 克莱恩·S·吉豪森; S·凯瑟琳·兰姆; 民昌·蔡 |
| 一种通过对数字化语音取样的帧作可变速率编码而将语音信号压缩的装置和方法。对于数字化语音取样的每个帧确定语音动作级别，依据所确定的帧语音动作级别，从一组速率中选出一个输出数据包速率。该组速率中的最低速率对应于检测到的最低语音动作级别，例如对应于语音中的背景噪声或停顿，而最高速率对应于检测到的最大语音动作级别，如对应于现有发音。每个帧根据选中速率的预定编码格式进行编码，各个速率具有代表编码帧的相应位数。对每个编码帧形成一个数据包，各输出数据包的位速率对应于选中的速率。 | ||||||
| 72 | 数字传输系统，发射机、接收机对等的模拟信号、和传输方法 | CN96190953.6 | 1996-07-08 | CN1164942A | 1997-11-12 | 肖群 |
| 已知有一种数字传输系统，它把已数字化的模拟信号的比特组，在以一连串PAM信号的方式串行传输之前，变换成对等的模拟信号。在接收机方，从所接收到的PAM信号重新构成模拟信号。这种传输仍要求相当大的带宽，两者在接收机方还要求帧同步和码元同步。因此提出了一种要求较少带宽并仅要求码元同步的数字传输系统。为达此目的，比特组经分开的传输通道并行传送，而在接收机方，从并行接收到的对等模拟信号组分中，重新构成所发送的信号。为解决传输通道噪音，放大最低有效比特组。为解决传输通道的有限的动态范围，缩小最高有效比特组。 | ||||||
| 73 | 用于产生高质量声音信号的自适应块长、自适应变换、及自适应窗变换代码、解码和编码/解码 | CN91102167.1 | 1991-04-06 | CN1055830A | 1991-10-30 | 格兰特·阿兰·戴维森 |
| 一种音乐信号等声音信号的高质量低速率数字变换编码，通过对每一声音采样段自适应地选择一种最佳变换，窗函数及变换块长，实现变换代码中时间分辨力和频率分辨力之间的最佳权衡。它适用所有离散的正交变换。其正交性保证通过正/反变换精确地再现信号。自适应地选择正交变换的块长，而不丢失信息，即无量化误差。本发明的最佳实施例中，也将自适应技术与非正交变换一起使用，该自适应块长选择保持了变换特性；(1)在无系数量化误差的情况下完全消除了假频；(2)临界采样。选择变换相位项以消除时域假频。 | ||||||
| 74 | 极值编码数字转换信号的处理方法和设备 | CN85109755 | 1985-12-19 | CN85109755A | 1987-04-08 | 阿里·维瑟 |
| 将模拟输入波形转换成数字信号的系统。它将传输信号速率降低，使接收机接到的再现模拟信号对人的听觉仍有主观的高质量。它包括极值编码器，将极值出现的时间或输入波形的极大极小值，包括自然产生或引入的基本上是随机宽带噪声出现时间，进行编码。将极值编码器的输出耦合到采用Δ调制器的连接线路，将连接线路的输出送到数字转换装置，将数字转换装置的输出耦合到传输信道，以传输到接收机。接收机有译码电路以再现原来的模拟信号。 | ||||||
| 75 | SUB-BAND SPLITTER UNIT AND ENVLOPE CURVES DETECTOR PROVIDED WITH A SUB-BAND SPLITTER UNIT | PCT/EP2013062649 | 2013-06-18 | WO2013189942A3 | 2014-03-06 | GROH JENS |
| The invention relates to a sub-band splitter unit for splitting a broadband input signal (G1), e.g., an audio signal, in K narrowband sub-band signals (L1,....., Lk,....., LK), wherein K is an integer larger than 1, which sub-band splitter unit (100) is provided with an input terminal (100) for receiving the broadband input signal and K-1 sub-band filter circuits (SBF1,....., SBFk,....., SBFK-1), each of the sub-band filter circuits (SBFk) is provided with an input (103.k) and a first (104.k) and a second output (105.k), a first filter arrangement (LPFk) coupled between the input (103.k) and the first output (104.k), a second filter arrangement (HPFk) coupled between the input (103.k) and the second output (105.k), and the sub-band splitter unit is provided with K output terminals (102.1,....., 102.k,....., 102.K) for supplying the K sub-band signals. The first output (104.k) of a k-th sub-band filter circuit (SBFk) is coupled to an input (103.k+1) of the (k+1)-th sub-band filter circuit. The input (103.1) of the first sub-band filter circuit (SBF1) is coupled to the input (101) of the sub-band splitter unit. The second output (105.k) of a k-th sub-band filter circuit is coupled to the k-th output (102.k) of the sub-band splitter unit. The first output (104.K-1) of the (K-1)-th sub-band filter circuit is coupled to the K-th output (104.K-1) of the sub-band splitter unit. The first filter arrangement (LPFk) of a sub-band filter circuit is adapted to carry out a lowpass filtering on the signal at the input of the first filter arrangement, and the second filter arrangement (HPFk) is adapted to carry out a highpass filtering on the signal at the input of the second filter arrangement. In addition, the sub-band splitter unit is furthermore devoid of any down-sampling means. (Fig.1). Furthermore, the invention relates to a sub-band combining unit (110), for which holds that it is provided with K input terminals (111.1,....., 1111.k,....., 111.K) for receiving the K narrowband sub-band signals and an output terminal (112). The sub-band combining unit is adapted for combining the K narrowband sub-band signals for generating a broadband output signal which is a replica of the broadband input signal of the sub-band splitter unit, and for supplying the broadband output signal to its output terminal. The sub-band combining unit is adapted to add time equivalent samples of the K narrowband sub-band signals to generate the broadband output signal. Thereby, the subband combining unit is devoid of any up-sampling means (Fig.1,2). Further, the invention proposes an envelope curves detector unit, wherein the envelope curves detector unit comprises the K-1 sub-band filter circuits of the sub-band splitter unit. | ||||||
| 76 | TRANSMITTER AND RECEIVER FOR A WIRELESS AUDIO TRANSMISSION SYSTEM | PCT/EP2005011495 | 2005-10-27 | WO2006045605A2 | 2006-05-04 | MEYER ROLF; PEISSIG JUERGEN; BUHE GERRIT |
| A transmitter for a wireless audio transmission system has at least one analogue/digital conversion unit for the analogue/digital conversion of the analogue audio signals to be transmitted, at least one digital signal processing unit with compressing/encoding means for compressing and encoding the digitised signals to be transmitted, a digital/analogue conversion unit for the digital/analogue conversion of the digital output signals from the digital signal processing unit into analogue signals, and a transmission unit for wirelessly transmitting the output signals from the analogue/digital conversion unit. Also disclosed is a receiver for a wireless audio transmission system having a reception unit for receiving wirelessly transmitted analogue H.F. signals, an intermediate frequency unit for mixing the H.F. signals into intermediate frequency signals, at least one analogue/digital conversion unit for the analogue/digital conversion of the wirelessly received signals, at least one digital signal processing unit with expanding/decoding means for expanding and decoding the signals digitised by the at least one analogue/digital conversion unit, and at least one digital/analogue conversion unit for converting the digital output signals from the digital signal processing unit into analogue signals. | ||||||
| 77 | VIDEO DEBLOCKING FILTER | PCT/US2005033782 | 2005-09-20 | WO2006034331A3 | 2009-04-09 | |
| Deblocking filters are disclosed (Fig. 2, item 20), where the nature of the filter is determined based upon the level of detail of a reconstructed video frame in the region in which the block boundary is located (Fig. 2, item 22). One embodiment of the method of the invention includes indentifying a boundary between two blocks of the reconstructed video frame, determining the level of detail of the reconstructed video frame in a region in which the block boundary is located (Fig. 2, item 24), wherein the region includes pixels from multiple columns of the reconstructed frame and includes at least one pixel that is not immediately adjacent the block boundary (See Fig. 2, item 26), and selecting a filter to apply to predetermined pixels on either side of the block boundary based upon the determined level of detail (See Fig. 2, item 30). | ||||||
| 78 | MULTI-CAMERA SURVEILLANCE SYSTEM AND METHOD FOR USING THE SAME | PCT/US2005004505 | 2005-02-10 | WO2006022824A3 | 2009-04-09 | BENGOECHEA XAVIER; FINIZIO FRANCESCO; GRICH RICHARD |
| An improved multi-camera surveillance system for use on a vehicle such as an aircraft, and a method for implementing the same. The multi-camera surveillance system is capable of displaying a user-selected image from any camera or user-selected images from multiple cameras at multiple viewing stations by flight and cabin crew, as well as recording and maintaining the images at storage locations on the aircraft and making the images available for viewing and recording at locations external to the aircraft by gate personnel, security officers, and incident investigators. | ||||||
| 79 | COMPRESSION RATE CONTROL SYSTEM AND METHOD WITH VARIABLE SUBBAND PROCESSING | PCT/US2005034025 | 2005-09-21 | WO2006034416A3 | 2006-11-30 | SAUNDERS STEVEN E; LYNCH WILLIAM C; KOLAROV KRASIMIR D |
| A system, method, and computer program product provide finer rate control in data compression by processing a data stream (201) tliroug a plurality of parallel subbands wherein a first subband processes the data differently than a second subband. Separate shift quantization (204) parameters can be provided for each subband. The parameter value can be recorded in the compressed output file (208). | ||||||
| 80 | 향상된 스펙트럼 확장을 사용하여 양자화 잡음을 감소시키기 위한 압신 장치 및 방법 | KR1020167015589 | 2014-04-01 | KR1020160075805A | 2016-06-29 | 헤델린,페르; 비스와스,아리짓; 슈그,미하엘; 멜코트,비나이 |
| 실시예들은오디오코덱에서코딩잡음을감소시키기위한압신방법및 시스템에관한것이다. 압축프로세스는정의된윈도우형태를사용하여초기오디오신호를복수의세그먼트들로분할하고, 상기초기오디오신호의주파수도메인샘플들의비-에너지기반평균을사용하여주파수도메인에서광대역이득을산출하고, 비교적낮은강도의세그먼트들을증폭시키며비교적높은강도의세그먼트들을감쇠시키기위해개개의이득값들을적용하는압축프로세스를통해초기오디오신호의원래동적범위를감소시킨다. 상기압축된오디오신호는그 후비교적높은강도의세그먼트들을증폭시키며비교적낮은강도의세그먼트들을감쇠시키기위해역 이득값들을적용하는실질적으로원래동적범위로다시확대된다. QMF 필터뱅크는주파수도메인표현을획득하도록초기오디오신호를분석하기위해사용된다. | ||||||
QQ群二维码
意见反馈
意见反馈