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    • 51. 发明授权
    • Binaural compression system
    • 双耳压缩系统
    • US07630507B2
    • 2009-12-08
    • US10353187
    • 2003-01-27
    • James M. Kates
    • James M. Kates
    • H04R25/00
    • H04B1/64H03G7/06H04R25/356H04R25/552
    • A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs. The multi-channel signal processing system further includes peak detecting means detecting signal peaks of the first and second channel input signals for generating first and second channel control signals. Thereafter, gain adjusting means adjusts signal gains of the first and second channel control signals. The first and second channel compressing units then respectively compress the first and second channel input signals to produce the first and second channel compressed outputs in accordance with the adjusted first and second channel control signals, respectively.
    • 提供了一种适用于提供音调输入的双耳压缩的多声道信号处理系统。 这样的系统可以用于例如双耳助听器系统中以提供音调输入的动态范围双耳压缩。 多通道信号处理系统本质上是一种具有通过两个信号通道之间的控制链路连接的两个信号通道的系统,从而允许双耳助听器系统模拟诸如人类的交叉橄榄色丛(COCB)效应的行为 听觉系统包括左耳和右耳之间的神经连接。 多信道信号处理系统包括分别位于多信道信号处理系统的第一和第二信号信道中的第一和第二信道压缩单元。 第一和第二信道压缩单元分别接收第一和第二信道输入信号,以产生第一和第二信道压缩输出。 多通道信号处理系统还包括检测第一和第二信道输入信号的信号峰值的峰值检测装置,用于产生第一和第二信道控制信号。 此后,增益调整装置调整第一和第二信道控制信号的信号增益。 第一和第二信道压缩单元然后分别压缩第一和第二信道输入信号,以分别根据调整的第一和第二信道控制信号产生第一和第二信道压缩输出。
    • 52. 发明申请
    • Binaural Hearing Aid System with Coordinated Sound Processing
    • 具有协调声音处理的双耳助听器系统
    • US20080212810A1
    • 2008-09-04
    • US10561476
    • 2004-06-23
    • Brian Dam Pedersen
    • Brian Dam Pedersen
    • H04R25/00
    • H04R25/552H04R25/407H04R25/558H04R2225/41
    • The present invention relates to a binaural hearing aid system comprising a first hearing aid and a second hearing aid, each of which comprises a microphone and an A/D converter for provision of a digital input signal in response to sound signals received at the respective microphone in a sound environment, a processor that is adapted to process the digital input signals in accordance with a predetermined signal processing algorithm to generate a processed output signal, and a D/A converter and an output transducer for conversion of the respective processed sound signal to an acoustic output signal, and a binaural sound environment detector for binaural determination of the sound environment surrounding a user of the binaural hearing aid system based on at least one signal from the first hearing aid and at least one signal from the second hearing aid for provision of outputs for each of the first and second hearing aids for selection of the signal processing algorithm of each of the respective hearing aid processors so that the hearing aids of the binaural hearing aid system perform coordinated sound processing.
    • 本发明涉及一种双耳助听器系统,包括第一助听器和第二助听器,每个助听器包括麦克风和A / D转换器,用于响应于在相应麦克风处接收到的声音信号来提供数字输入信号 在良好的环境中,处理器适于根据预定的信号处理算法处理数字输入信号以产生经处理的输出信号;以及D / A转换器和输出换能器,用于将各个经处理的声音信号转换为 声输出信号和双耳声音环境检测器,用于基于来自第一助听器的至少一个信号和来自第二助听器的至少一个信号来双耳确定双耳助听器系统的用户周围的声音环境,以提供 的第一和第二助听器中的每一个的输出,用于选择每一个的信号处理算法 辅助助听器处理器,使双耳助听器系统的助听器进行协调的声音处理。
    • 53. 发明授权
    • Binaural signal enhancement system
    • 双耳信号增强系统
    • US07330556B2
    • 2008-02-12
    • US10407305
    • 2003-04-03
    • James M. Kates
    • James M. Kates
    • H04R25/00
    • H04R25/407H04R25/552H04R2225/41
    • A signal processing system, such as a hearing aid system, adapted to enhance binaural input signals is provided. The signal processing system is essentially a system with a first signal channel having a first filter and a second signal channel having a second filter for processing first and second channel inputs and producing first and second channel outputs, respectively. Filter coefficients of at least one of the first and second filters are adjusted to minimize the difference between the first channel input and the second channel input in producing the first and second channel outputs. The resultant signal match processing of the signal processing system gives broader regions of signal suppression than using the Wiener filters alone for frequency regions where the interaural correlation is low, and may be more effective in reducing the effects of interference on the desired speech signal. Modifications to the algorithms can be made to accommodate sound sources located to the sides as well as the front of the listener. Processing artifacts can be reduced by using longer averaging time constants for estimating the signal power and cross-spectra as the signal-to-noise ratio decreases. A stability constant can also be incorporated in the transfer functions of the first and second filters to increase the stability of the signal processing system.
    • 提供了一种适于增强双耳输入信号的信号处理系统,例如助听器系统。 信号处理系统本质上是具有第一信号信道的系统,其具有第一滤波器和具有用于处理第一和第二信道输入的第二滤波器的第二信号信道,并且分别产生第一和第二信道输出。 调整第一和第二滤波器中的至少一个滤波器的滤波器系数,以便在产生第一和第二通道输出时最小化第一通道输入和第二通道输入之间的差异。 信号处理系统的结果信号匹配处理给出了比单独使用维纳滤波器用于频偏相关低的频率区域的更宽的信号抑制区域,并且可以更有效地减少干扰对期望语音信号的影响。 可以对算法进行修改以适应位于侧面以及收听者前面的声源。 通过使用较长的平均时间常数来减少处理伪影,以估计信噪比降低信号功率和交叉谱。 稳定常数也可以并入第一和第二滤波器的传递函数中,以增加信号处理系统的稳定性。
    • 55. 发明申请
    • Hearing aid with suppression of wind noise
    • 助听器抑制风噪声
    • US20070030989A1
    • 2007-02-08
    • US11497664
    • 2006-08-01
    • James Kates
    • James Kates
    • H04R25/00
    • H04R25/505H04R25/502H04R2410/07
    • The present application relates to a hearing aid with suppression of wind noise wherein wind noise detection is provided involving only a single comparison of the input signal power level at first low frequencies with the input signal power level at frequencies that may include the first low frequencies whereby a computational cost effective and simple wind noise detection is provided. The determination of relative power levels of the input signal reflects the shape of the power spectrum of the signal, and the detection scheme is therefore typically capable of distinguishing music from wind noise so that attenuation of desired music is substantially avoided.
    • 本申请涉及一种抑制风噪声的助听器,其中提供的风噪声检测仅涉及第一低频处的输入信号功率电平与可包括第一低频的频率的输入信号功率电平的单一比较,由此 提供了计算成本有效和简单的风噪声检测。 输入信号的相对功率电平的确定反映了信号的功率谱的形状,因此检测方案通常能够区分音乐与风噪声,从而基本上避免了所需音乐的衰减。
    • 56. 发明授权
    • Hearing aid with delayed activation
    • 助听器延迟激活
    • US07031481B2
    • 2006-04-18
    • US10359915
    • 2003-02-07
    • Rene Mortensen
    • Rene Mortensen
    • H04R25/00
    • H04R25/453H04R25/505
    • A hearing aid comprises a microphone, a sound transducer, an amplification signal path configured for coupling the sound transducer to the microphone; and a circuit configured for completely or partly blocking the signal path during a comfort delay period, which may be adjustable, when the hearing aid is switched on. It can hereby be avoided that the hearing aid transmits a disturbing howling tone during the period until it is placed correctly at or in the ear of the user, if the user switches on the hearing aid before it is correctly positioned. The circuit is further configured for generating and transmitting a special acoustic signal to the sound transducer during at least a portion of the comfort delay period. In this way a user will be able to ascertain whether the hearing aid is switched on and that it functions as it should.
    • 助听器包括麦克风,声音换能器,被配置用于将声音传感器耦合到麦克风的放大信号路径; 以及电路,被配置为在助听器接通时在舒适延迟时段期间完全或部分地阻挡信号路径,舒适延迟时段可以是可调节的。 因此,如果用户在正确定位之前接通助听器,则可以避免助听器在该期间内发出干扰啸叫声,直到其被正确地放置在用户的耳朵中或耳朵中。 电路还被配置为在舒适延迟时段的至少一部分期间产生和发送特殊声信号到声换能器。 以这种方式,用户将能够确定助听器是否被接通并且它的功能正常。
    • 57. 发明授权
    • Hearing prosthesis with automatic classification of the listening environment
    • 听觉假体具有听觉环境的自动分类
    • US06862359B2
    • 2005-03-01
    • US10157547
    • 2002-05-29
    • Nils Peter NordqvistArne Leijon
    • Nils Peter NordqvistArne Leijon
    • H04R25/00G10L15/14
    • H04R25/505H04R2225/41
    • A hearing prosthesis that automatically adjusts itself to a surrounding listening environment by applying Hidden Markov Models is provided. In one aspect, classification results are utilized to support automatic parameter adjustment of a parameter or parameters of a predetermined signal processing algorithm executed by processing means of the hearing prosthesis. According to another aspect, features vectors extracted from a digital input signal of the hearing prosthesis and processed by the Hidden Markov Models represent substantially level and/or absolute spectrum shape independent signal features of the digital input signal. This level independent property of the extracted features vectors provides robust classification results in real-life acoustic environments.
    • 提供了通过应用隐马尔可夫模型自动调整到周围听力环境的听觉假体。 在一个方面,分类结果用于支持由听觉假体的处理装置执行的预定信号处理算法的参数或参数的自动参数调整。 根据另一方面,从听觉假体的数字输入信号提取并由隐马尔科夫模型处理的特征矢量表示数字输入信号的基本上水平和/或绝对频谱形状独立的信号特征。 提取的特征向量的这个级别独立性在实际的声学环境中提供了强大的分类结果。
    • 59. 发明申请
    • Fitting methodology and hearing prosthesis based on signal-to-noise ratio loss data
    • 基于信噪比损失数据的拟合方法和听觉假体
    • US20040047474A1
    • 2004-03-11
    • US10422258
    • 2003-04-24
    • GN Resound A/S
    • Aalbert De VriesRob Anton Jurjen De Vries
    • H04R029/00A61B005/12
    • H04R25/70H04R2225/41H04R2460/01
    • An individual with a hearing loss often experiences at least two distinct problems: 1) the hearing loss itself i.e. an increase in hearing threshold level, and 2) a signal-to-noise ratio loss (SNR loss) i.e. a loss of ability to understand high level speech in noise as compared to normal hearing individuals. According to one aspect of the present invention, this problem is solved by selecting parameter values of a noise reduction algorithm or algorithms based on the individual user's SNR loss. Thereby, a degree of restoration/improvement of the SNR of noise-contaminated input signals of the hearing prosthesis has been made dependent on user specific loss data. According to another aspect of the present invention, a hearing prosthesis capable of controlling parameters of a noise reduction algorithms in dependence on the user's current listening environment as recognized and indicated by the environmental classifier has been provided.
    • 具有听力损失的个体经常遇到至少两个明显的问题:1)听力损失本身即听力阈值水平的增加,以及2)信噪比损失(SNR损失),即失去理解能力 与正常听力人士相比,噪音高水平的演讲。 根据本发明的一个方面,通过基于个体用户的SNR损失选择降噪算法或算法的参数值来解决该问题。 因此,依赖于用户特定的损失数据,已经使得听觉假体噪声污染的输入信号的SNR的恢复/改善程度取决于用户特定的损失数据。 根据本发明的另一方面,提供了能够根据由环境分类器识别和指示的用户的当前聆听环境来控制降噪算法的参数的听力假体。