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
    • 12. 发明授权
    • Encoder/decoder for multidimensional sound fields
    • 用于多维声场的编码器/解码器
    • US5583962A
    • 1996-12-10
    • US927429
    • 1992-09-04
    • Mark F. DavisCraig C. ToddRay M. Dolby
    • Mark F. DavisCraig C. ToddRay M. Dolby
    • G10L19/02G06F9/445G10L19/00G11B20/00G11B20/10H03M1/12H03M7/30H04B14/04H04H1/00H04H20/91H04S3/00G10L9/18G10L7/00
    • H04S3/008G10L19/008G11B20/00G11B20/10527H04H20/91
    • Two or more audio channels (i.e.--stereo, 4-channel surround, etc.) are each divided into frequency subbands to be coarsely quantized. An adaptive bit allocation scheme is then applied to subbands which are combined across channels such that equivalent subbands (i.e.--same frequency band) from each channel are grouped together to form a steered subband. The power from each subband is averaged across the channels to form a steered subband level. Bits are conserved by forming a vector in which each channel is represented by the difference between the steered subband level (average) and the actual subband level. Subbands which are not steered are represented by the coarse quantization and are considered unsteered channel subbands. Subbands which are steered are represented by vectors and are considered composite channel subbands. Vectors may be represented using a lookup table and the steered subband levels may be calculated using alternatives such as a peak level rather than an average.
    • PCT No.PCT / US92 / 00133 Sec。 371日期:1992年9月4日 102(e)1992年9月4日PCT PCT 1992年1月8日PCT公布。 公开号WO92 / 12607 日期1992年7月23日将两个或更多个音频通道(即,立体声,4声道环绕等)分别划分成要被粗量化的频率子带。 然后,自适应位分配方案被应用于跨信道组合的子带,使得来自每个信道的等同子带(即,相同的频带)被分组在一起以形成转向子带。 每个子带的功率在通道之间被平均以形成转向子带电平。 通过形成矢量来保存位,其中每个信道由转向子带电平(平均)和实际子带电平之间的差异表示。 未被引导的子带由粗量化表示,并且被认为是非导向信道子带。 被转向的子带由矢量表示,并被认为是复合信道子带。 可以使用查找表来表示向量,并且可以使用诸如峰值级别而不是平均值的替代来计算转向子带级别。
    • 13. 发明授权
    • Low bit-rate high-resolution spectral envelope coding for audio encoder
and decoder
    • 用于音频编码器和解码器的低比特率高分辨率频谱包络编码
    • US5581653A
    • 1996-12-03
    • US115513
    • 1993-08-31
    • Craig C. Todd
    • Craig C. Todd
    • H04N7/26H03M7/30H04B1/66H04B14/04H04N1/41G10L3/02G10L9/00
    • G10L19/035H04B1/667
    • A split-band encoder prepares an estimate of an input signal spectral envelope by splitting the input signal into frequency subband signals, generates a scaled representation of the subband signals comprising scaling factors and scaled values, generates a differential coded representation of the scaling factors, and assembles the differential coded representation and scaled values into an encoded signal. The scale factors represent a spectral envelope of the input signal and spectral leakage between filterbank subbands limits the change in value between adjacent scaling factors. This limitation in change can be exploited to reduce the informational requirements of the differential coded representation. Adaptive selection between high- and low-resolution spectral envelopes and adaptive scaling factor reuse are described.
    • 分频编码器通过将输入信号分成频率子带信号来准备输入信号频谱包络的​​估计,产生包括缩放因子和缩放值的子带信号的缩放表示,产生缩放因子的差分编码表示,以及 将差分编码表示和缩放值组合成编码信号。 比例因子表示输入信号的频谱包络,滤波器组子带之间的频谱泄漏限制了相邻缩放因子之间的值的变化。 可以利用这种变化限制来减少差分编码表示的信息要求。 描述了高分辨率和低分辨率频谱包络之间的自适应选择以及自适应缩放因子重用。
    • 15. 发明授权
    • Compatible digital audio for NTSC television
    • NTSC电视兼容的数字音频
    • US5357284A
    • 1994-10-18
    • US501608
    • 1990-03-29
    • Craig C. Todd
    • Craig C. Todd
    • H04N5/455H04N5/60H04N7/04H04N7/06H04N7/00
    • H04N5/455H04N5/605H04N7/04H04N7/06
    • A method and apparatus for transmitting a digitally modulated Quadrature Phase Keyed (QPSK) audio carrier signal, or a digitally modulated Quadrature Partial Response System (QPRS) audio carrier signal, wherein the audio carrier signal is located 1.2 MHz below the video carrier of an NTSC signal. This places the digital audio signal 300 KHz above the analog FM sound center frequency of the NTSC signal in the adjacent lower channel, 4.8 MHz above the video carrier of the adjacent lower channel NTSC signal and at the edge of the lower vestigal video sideband components of the NTSC signal with which the digital audio signal is associated.
    • 一种用于发送数字调制正交相位键控(QPSK)音频载波信号或数字调制正交部分响应系统(QPRS)音频载波信号的方法和装置,其中音频载波信号位于NTSC的视频载波之下1.2MHz 信号。 这样将数字音频信号在NTSC信号的模拟FM声音中心频率上方300KHz放置在邻近的下一个信道中,相邻下一个信道NTSC信号的视频载波上方4.8MHz,并且在较低的背景视频边带分量 与数字音频信号相关联的NTSC信号。
    • 17. 发明授权
    • Computationally efficient adaptive bit allocation for coding method and
apparatus
    • 用于编码方法和装置的计算有效的自适应比特分配
    • US5632003A
    • 1997-05-20
    • US145975
    • 1993-11-01
    • Grant A. DavidsonCraig C. ToddMark F. DavisBrian D. LinkLouis D. Fielder
    • Grant A. DavidsonCraig C. ToddMark F. DavisBrian D. LinkLouis D. Fielder
    • H03M7/30H04B1/66H04B14/06G10L9/00
    • H04B1/665H04B1/667
    • The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to computationally efficient adaptive bit allocation and quantization of encoded information useful in high-quality low bit-rate coding systems.In one embodiment, an audio split-band encoder splits an input signal into frequency subband signals, quantizes the subband signals according to values established by an allocation function, and assembles the quantized subband signals into an encoded signal. The allocation function establishes allocation values in accordance with psychoacoustic principles based upon a masking threshold. The masking threshold is established by estimating the power spectral density (PSD) of the input signal, generating an excitation pattern by applying a spreading function to the PSD, adjusting the excitation pattern by an amount equal to a signal-to-noise ratio (SNR) offset sufficient to achieve psychoacoustic masking, comparing the level of the adjusted pattern to the threshold of hearing and generating a masking threshold which is equal to the larger of the two. The spreading function may be implemented by applying one or more IIR filters to the input signal PSD.
    • 本发明一般涉及诸如音频信息的信息的低比特率编码和解码。 更具体地,本发明涉及在高质量低比特率编码系统中有用的编码信息的计算上有效的自适应比特分配和量化。 在一个实施例中,音频分离带编码器将输入信号分离成频率子带信号,根据由分配函数建立的值量化子带信号,并将量化的子带信号组合成编码信号。 分配功能基于掩蔽阈值根据心理声学原理建立分配值。 通过估计输入信号的功率谱密度(PSD)来建立屏蔽阈值,通过向PSD施加扩展函数来产生激励模式,将激励模式调整等于信噪比(SNR)的量 )偏移足以实现心理声学掩蔽,将调整模式的电平与听力阈值进行比较,并产生等于两者中较大者的掩蔽阈值。 可以通过将一个或多个IIR滤波器应用于输入信号PSD来实现扩展功能。