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    • 64. 发明申请
    • Dynamic Decoding of Binaural Audio Signals
    • 双耳音频信号的动态解码
    • US20080008327A1
    • 2008-01-10
    • US11456191
    • 2006-07-08
    • Pasi OjalaJulia Turku
    • Pasi OjalaJulia Turku
    • H04R5/00
    • H04S3/02H04S2400/01H04S2420/01H04S2420/03
    • Inputting of a parametrically encoded audio signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information describing a multi-channel sound image and including channel configuration information is shown along with deriving, from the channel configuration information, audio source location data describing at least one of horizontal and vertical positions of audio sources in the binaural audio signal; selecting, from a predetermined set of head-related transfer function filters, a left-right pair of head-related transfer function filters matching closest to the audio source location data, wherein the left-right pair of head-related transfer function filters is searched in a stepwise motion in a horizontal plane; and synthesizing a binaural audio signal from the at least one processed signal according to side information and the channel configuration information.
    • 示出了包括多个音频频道的至少一个组合信号和描述多声道声音图像并且包括频道配置信息的一个或多个对应的侧面信息组的参数编码音频信号的输入以及从信道配置 信息,音频源位置数据,描述双耳音频信号中音频源的水平和垂直位置中的至少一个; 从预定的头部相关传递函数滤波器组中选择最接近音频源位置数据的与头部相关的传送函数滤波器对的匹配对象,其中搜索左右对头部相关传递函数滤波器 在水平面上逐步运动; 以及根据侧信息和信道配置信息从至少一个处理信号合成双耳音频信号。
    • 65. 发明授权
    • Bit-rate control in a multimedia device
    • 多媒体设备中的比特率控制
    • US06704281B1
    • 2004-03-09
    • US09483143
    • 2000-01-13
    • Ari HourunrantaMarko LuomiPasi Ojala
    • Ari HourunrantaMarko LuomiPasi Ojala
    • H04L1226
    • H04N21/2368H04N7/52H04N21/2335H04N21/2662H04N21/6587
    • A multimedia terminal comprising: a first encoder (100) for producing a first bit-stream (107) of a first media type and having a first bit-rate; a second encoder (110) for producing a second bit-stream (112) of a second media type and having a second bit-rate; a multiplexer (120) for combining at least the first (106) and the second (112) bit-streams into a third bit-stream (123). The terminal comprises an input element (130) for receiving preference information (131) coupled to the first encoder (100) and the second encoder (110), said preference information (131) indicating a preferred combination of the first and the second media types in the third bit-stream and affecting the first and the second bit-rates. Thus, the transmission capacity is utilised in a more optimised manner and the proportions of different media types are better adjusted to the purpose of the information transfer.
    • 一种多媒体终端,包括:第一编码器(100),用于产生具有第一比特率的第一媒体类型的第一比特流(107) 用于产生具有第二比特率的第二媒体类型的第二比特流(112)的第二编码器(110) 用于将至少第一(106)和第二(112)比特流合并成第三比特流(123)的多路复用器(120)。 终端包括用于接收耦合到第一编码器(100)和第二编码器(110)的偏好信息(131)的输入元件(130),所述偏好信息(131)指示第一和第二媒体类型的优选组合 在第三个比特流中并影响第一个和第二个比特率。 因此,以更优化的方式利用传输容量,并且将不同媒体类型的比例更好地调整到信息传送的目的。
    • 66. 发明授权
    • Adaptive postfilter
    • 自适应后置滤波器
    • US06584441B1
    • 2003-06-24
    • US09234099
    • 1999-01-20
    • Pasi OjalaKari Järvinen
    • Pasi OjalaKari Järvinen
    • G10L1914
    • G10L19/26
    • The invention relates to the coding of speech at a variable bit rate, whereby the bit rates can vary from frame to frame, and more specifically to the methods and filters used for improving the quality of the decoded speech. In the solution according to the invention the weighting factors of the postfilter are not adapted on the basis of the momentary bit rate or the bit rate used in the coding of each frame, but the weighting factors are adapted according to the average bit rate calculated on the basis of a predetermined length of time. In addition to this, the weighting factors of the postfilter are also adjusted on the basis of whether the frame in question contains a voiced speech signal, an unvoiced speech signal or background noise. At frames containing an unvoiced speech signal or background noise, postfiltering is weakened so as to avoid the distortion of the signal tone because the postfiltering is adapted to a voiced signal. The weighting factors of the postfilter can also be adapted on the basis of the error rate or other parameter describing the quality of the signal or the data transfer channel. For example, postfiltering can conveniently be adjusted so that when the channel error rate and the amount of coding error increase, postfiltering is increased, whereby the effect of data transfer errors on the decoded speech signal is reduced and the tolerance of the system with regard to data transfer errors increases.
    • 本发明涉及以可变比特率对语音进行编码,由此比特率可以随帧而变化,更具体地说,涉及用于提高解码语音质量的方法和滤波器。 在根据本发明的解决方案中,后置滤波器的加权因子不是基于在每个帧的编码中使用的瞬时比特率或比特率来适应的,而是根据在 是预定时间的基础。 除此之外,后置滤波器的加权因子也根据所讨论的帧是否包含有声语音信号,无声语音信号或背景噪声进行调整。 在包含无声语音信号或背景噪声的帧中,后滤波被削弱,以避免信号音的失真,因为后滤波适用于有声信号。 后置滤波器的加权因子也可以根据描述信号质量或数据传输通道的错误率或其他参数进行调整。 例如,可以方便地调整后置滤波,使得当信道误码率和编码误差量增加时,后置滤波增加,从而降低了对解码语音信号的数据传输误差的影响,并且系统对于 数据传输错误增加。
    • 67. 发明授权
    • Speech coding
    • 语音编码
    • US06470313B1
    • 2002-10-22
    • US09263439
    • 1999-03-04
    • Pasi Ojala
    • Pasi Ojala
    • G01L1912
    • G10L19/002G10L19/06
    • A variable bit-rate speech coding method determines for each subframe a quantised vector d(i) comprising a variable number of pulses. An excitation vector c(i) for exciting LTP and LPC synthesis filters is derived by filtering the quantised vector d(i), and a gain value gc is determined for scaling the pulse amplitude excitation vector c(i) such that the scaled excitation vector represents the weighted residual signal {tilde over (s)} remaining in the subframe speech signal after removal of redundant information by LPC and LTP analysis. A predicted gain value ĝc is determined from previously processed subframes, and as a function of the energy Ec contained in the excitation vector c(i) when the amplitude of that vector is scaled in dependence upon the number of pulses m in the quantised vector d(i). A quantised gain correction factor {circumflex over (&ggr;)}gc is then determined using the gain value gc and the predicted gain value ĝc.
    • 可变比特率语音编码方法为每个子帧确定包括可变数目的脉冲的量化矢量d(i)。 通过对量化矢量d(i)进行滤波来得到用于激发LTP和LPC合成滤波器的激励矢量c(i),并且确定用于缩放脉冲振幅激励矢量c(i)的增益值gc,使得缩放的激励矢量 表示通过LPC和LTP分析去除冗余信息之后残留在子帧语音信号中的加权残差信号{波形(s)}}。 根据先前处理的子帧确定预测的增益值ĝc,并且根据量子化矢量d中的脉冲数m对该向量的幅度进行缩放时,作为激励矢量c(i)中包含的能量Ec的函数 (一世)。 然后使用增益值gc和预测增益值ĝc来确定量化的增益校正因子(回旋(γ)} gc)。
    • 70. 发明授权
    • Apparatus and method for adjusting spatial cue information of a multichannel audio signal
    • 用于调整多声道音频信号的空间提示信息的装置和方法
    • US09025775B2
    • 2015-05-05
    • US13002486
    • 2008-07-01
    • Pasi Ojala
    • Pasi Ojala
    • H04R5/00G10L19/008
    • G10L19/008
    • An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.
    • 一种用于增强多声道音频信号的装置,包括至少两个通道,所述至少两个通道被配置为:从至少第一通道估计表示与第一音频信号相关联的到达方向的值,以及至少来自至少第二信道的第二音频信号 两声道的多声道音频信号; 确定取决于与第一音频信号和第二音频信号相关联的到达方向的缩放因子; 并将缩放因子应用于与第一音频信号和第二音频符号之间的音频信号电平的差异相关联的参数。