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. 发明申请
    • CODING GENERIC AUDIO SIGNALS AT LOW BITRATES AND LOW DELAY
    • 编码低频和低延迟的一般音频信号
    • WO2012055016A1
    • 2012-05-03
    • PCT/CA2011001182
    • 2011-10-24
    • VOICEAGE CORPVAILLANCOURT TOMMYJELINEK MILAN
    • VAILLANCOURT TOMMYJELINEK MILAN
    • G10L19/12
    • G10L19/20G10L19/02G10L19/08
    • A mixed time-domain / frequency-domain coding device and method for coding an input sound signal, wherein a time-domain excitation contribution is calculated in response to the input sound signal. A cut-off frequency for the time-domain excitation contribution is also calculated in response to the input sound signal, and a frequency extent of the time-domain excitation contribution is adjusted in relation to this cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to the input sound signal, the adjusted time-domain excitation contribution and the frequency-domain excitation contribution are added to form a mixed time-domain / frequency-domain excitation constituting a coded version of the input sound signal. In the calculation of the time-domain excitation contribution, the input sound signal may be processed in successive frames of the input sound signal and a number of sub-frames to be used in a current frame may be calculated. Corresponding encoder and decoder using the mixed time-domain / frequency-domain coding device are also described.
    • 一种用于编码输入声音信号的混合时域/频域编码装置和方法,其中响应于输入声音信号计算时域激励贡献。 还响应于输入声音信号计算时域激励贡献的截止频率,并且相对于该截止频率调整时域激励贡献的频率范围。 在响应于输入声音信号计算频域激励贡献之后,调整调整的时域激励贡献和频域激励贡献以形成构成编码版本的混合时域/频域激励 输入声音信号。 在时域激励贡献的计算中,可以在输入声音信号的连续帧中处理输入声音信号,并且可以计算要在当前帧中使用的多个子帧。 还描述了使用混合时域/频域编码装置的对应编码器和解码器。
    • 5. 发明申请
    • VARIABLE BIT RATE LPC FILTER QUANTIZING AND INVERSE QUANTIZING DEVICE AND METHOD
    • 可变位速率LPC滤波器量子和反相量化器件及方法
    • WO2010003253A1
    • 2010-01-14
    • PCT/CA2009000980
    • 2009-07-10
    • VOICEAGE CORPBESSETTE BRUNOGOURNAY PHILIPPESALAMI REDWAN
    • BESSETTE BRUNOGOURNAY PHILIPPESALAMI REDWAN
    • G10L19/12
    • G10L19/06G10L19/18H03M7/3082
    • A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector. A device and a method for inverse quantizing of a LPC filter, comprises means for receiving coded indices representative of a first-stage approximation of a vector representative of the LPC filter in a quantization domain and of a quantized weighted residual version of the vector, a calculator of an inverse weighting function from the first-stage approximation, an inverse quantizer of the quantized weighted residual version of the vector to produce a weighted residual vector, a multiplier of the weighted residual vector by the inverse weighting function to produce a residual vector, and an adder of the first-stage approximation with the residual vector to produce the vector representative of the LPC filter in the quantization domain.
    • 在量化域中以输入向量的形式对LPC滤波器进行量化的装置和方法包括:输入向量的第一级逼近的计算器,从输入向量产生的第一级逼近的减法器 一个残差矢量,一个来自第一阶段逼近的加权函数的计算器,具有加权函数的残差向量的整经器,以及加权残差向量的量化器,以提供量化的加权残差向量。 用于LPC滤波器的逆量化的装置和方法包括用于接收代表代表量化域中的LPC滤波器的矢量的第一级逼近的编码索引和矢量的量化加权残差版本的装置, 来自第一阶段近似的逆加权函数的计算器,用于产生加权残差向量的向量的量化加权残差版本的逆量化器,通过逆加权函数的加权残差向量的乘数以产生残差向量, 以及具有残差矢量的第一级近似的加法器,以产生代表量化域中的LPC滤波器的向量。
    • 8. 发明申请
    • SYSTEM AND METHOD FOR ENHANCING A DECODED TONAL SOUND SIGNAL
    • 用于增强解码的声音信号的系统和方法
    • WO2009109050A1
    • 2009-09-11
    • PCT/CA2009000276
    • 2009-03-05
    • VOICEAGE CORPVAILLANCOURT TOMMYJELINEK MILANMALENOVSKY VLADIMIRSALAMI REDWAN
    • VAILLANCOURT TOMMYJELINEK MILANMALENOVSKY VLADIMIRSALAMI REDWAN
    • G10L21/02G10L19/12
    • G10L19/26G10L25/18
    • A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.
    • 一种用于响应于接收的编码比特流来增强由语音专用编解码器的解码器解码的音调声音信号的系统和方法,其中频谱分析仪响应于解码的音调声音信号以产生表示解码的频谱参数 音调声信号。 响应于由光谱分析仪产生的光谱参数,解码的音调声音信号的低能谱区域中的量化噪声被减小。 光谱分析仪将由光谱分析得到的光谱分成一组包括多个频率仓的临界频带,并且量化噪声的衰减器包括噪声衰减器,其对每个关键频带的解码音调声音信号的频谱进行缩放, 每个频率仓,或每个临界频带和频率仓。
    • 9. 发明申请
    • METHODS FOR INTEROPERATION BETWEEN ADAPTIVE MULTI-RATE WIDEBAND (AMR-WB) AND MULTI-MODE VARIABLE BIT-RATE WIDEBAND (WMR-WB) SPEECH CODECS
    • 用于自适应多速率宽带(AMR-WB)和多模式可变比特率宽带(WMR-WB)语音编码器之间的交互的方法
    • WO2004034376A3
    • 2004-06-10
    • PCT/CA0301572
    • 2003-10-10
    • VOICEAGE CORPJELINEK MILANSALAMI REDWAN
    • JELINEK MILANSALAMI REDWAN
    • G01L19/14G10L11/04G10L19/02G10L19/14G10L21/02
    • G10L19/24G10L19/012G10L19/173
    • A source-controlled Variable bit-rate Multi-mode WideBand (VMR-WB) speech codec, having a mode of operation that is interoperable with the Adaptive Multi-Rate wideband (AMR-WB) codec, the codec comprising: at least one Interoperable full-rate (1-FR) mode, having a first bit allocation structure based an one of a AMR-WB codec coding types; and at least one comfort noise generator (CNG) coding type for encoding inactive speech frame having a second bit allocation structure based on AMR-WB SID_UPDATE coding type. Methods for i) digitally encoding a sound using a source-controlled Variable bit rate multi-mode wideband (VMR-WB) speech codec for interoperation with an adaptative multi-rate wideband (AMR-WB) codec, ii) translating a Variable bit rate multi-mode wideband (VMR-WB) speech codec-signal frame into an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame, iii) translating an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame into a Variable bit rate multi-mode wideband (VMR-WB) speech signal frame, and iv) translating an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame into a Variable bit rate multi-mode wideband (VMR-WB) speech signal frame are also provided.
    • 一种具有与自适应多速率宽带(AMR-WB)编解码器相互操作的操作模式的源控制的可变比特率多模式宽带(VMR-WB)语音编解码器,所述编解码器包括:至少一个可互操作的 全速率(1-FR)模式,具有基于AMR-WB编解码器类型之一的第一比特分配结构; 以及至少一种用于编码基于AMR-WB SID_UPDATE编码类型的具有第二位分配结构的无效语音帧的舒适噪声发生器(CNG)编码类型。 用于i)使用源控制的可变比特率多模宽带(VMR-WB)语音编解码器对数字编码声音的方法,用于与适应性多速率宽带(AMR-WB)编解码器进行互操作,ii)将可变比特率 多模宽带(VMR-WB)语音编解码信号帧转换为自适应多速率宽带(AMR-WB)语音信号帧,iii)将自适应多速率宽带(AMR-WB)语音信号帧转换为变量 比特率多模宽带(VMR-WB)语音信号帧,以及iv)将自适应多速率宽带(AMR-WB)语音信号帧转换为可变比特率多模宽带(VMR-WB)语音信号帧 也提供。
    • 10. 发明申请
    • METHOD AND DEVICE FOR EFFICIENT IN-BAND DIM-AND-BURST SIGNALING AND HALF-RATE MAX OPERATION IN VARIABLE BIT-RATE WIDEBAND SPEECH CODING FOR CDMA WIRELESS SYSTEMS
    • 用于CDMA无线系统的可变位速率宽带语音编码中的有效带内DIM-AND-BURST信令和高达率最大值操作的方法和设备
    • WO2004006226B1
    • 2004-03-04
    • PCT/CA0300980
    • 2003-06-27
    • VOICEAGE CORPJELINEK MILANSALAMI REDWAN
    • JELINEK MILANSALAMI REDWAN
    • G10L19/12G10L19/24H03M7/30H04B1/707H04B7/24H04B7/26G10L19/14H04Q7/30
    • G10L19/24
    • In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.
    • 在用于使用第一通信方案互操作第一站的方法和设备中,包括第一编码器和具有第二站的第一解码器,并且包括第二编码器和第二解码器,第一和第二站之间的通信 通过将与来自第一和第二站中的一个的编码器的声音信号相关的信号编码参数发送到另一站的解码器来进行。 声音信号被分类以确定信号编码参数是否应当使用全位比特率用于传输信号编码参数的第一通信模式从一个站的编码器发送到另一站的解码器。 当声音信号的分类确定应当使用第一通信模式发送信号编码参数时,以及当使用第二通信模式从一个站的编码器向另一站的解码器发送信号编码参数的请求时 被设计为在信号编码参数的传输期间降低比特率被接收到,来自编码器一个站的信号编码参数的一部分被丢弃,剩下的信号编码参数使用第二个信号编码参数传送到另一台的解码器 通讯模式。 信号编码参数的丢弃部分在另一站的解码器解码信号编码参数之前被再生。