公开(公告)号：US08812923B2

公开(公告)日：2014-08-19

申请号：US13302094

申请日：2011-11-22

Applicant: Christophe Marc Macours

Inventor： Christophe Marc Macours

IPC: H04L1/18 ,  G10L15/00 ,  G10L19/005 ,  H03M7/30 ,  G10L19/02 ,  H04L1/00

CPC classification number: G10L19/005 ,  G10L19/0204 ,  H03M7/3046 ,  H04L1/0072

Abstract: A decoder and method of decoding a sub-band coded digital audio signal. The decoder comprises: an input, for receiving sub-band coefficients for a plurality of sub-bands of the audio signal; an error detection unit, adapted to analyze the content of a sequence of coefficients in one of the sub-bands, to derive for each coefficient an indication of whether the coefficient has been corrupted by an error of a predefined type; an error masking unit, adapted to generate from the sequence a modified sequence of coefficients for the sub-band, wherein errors of the predefined type are attenuated; a coefficient combination unit, adapted to combine the received coefficients and the modified coefficients, in dependence upon the indication of error; and a signal reconstruction unit, adapted to reconstruct the audio signal using the combined coefficients.

Abstract translation: 一种对子带编码的数字音频信号进行解码和解码的方法。 解码器包括：用于接收音频信号的多个子带的子带系数的输入; 错误检测单元，适于分析一个子带中的一系列系数的内容，为每个系数导出该系数是否已被预定类型的错误破坏的指示; 一个误差屏蔽单元，适于从该序列产生经修改的子带序列，其中预定义类型的误差被衰减; 系数组合单元，其适于根据误差的指示来组合所接收的系数和修正的系数; 以及信号重建单元，其适于使用所述组合系数来重构所述音频信号。

公开(公告)号：US08787602B2

公开(公告)日：2014-07-22

申请号：US12855557

申请日：2010-08-12

Applicant: Christophe Marc Macours

Inventor： Christophe Marc Macours

IPC: H04R5/02 ,  H04R5/033 ,  H04R29/00 ,  H04R3/12

CPC classification number: H04R5/04 ,  H04R25/43 ,  H04R25/552 ,  H04R2201/109 ,  H04R2420/01 ,  H04R2420/03 ,  H04R2460/01 ,  H04S1/005 ,  H04S7/304

Abstract: A device (100) for processing audio data, wherein the device (100) comprises a first audio reproduction unit (102) adapted for reproducing a first part of the audio data and adapted to be attached to a left ear (106) of a user (110), a second audio reproduction unit (104) adapted for reproducing a second part of the audio data and adapted to be attached to a right ear (108) of the user (110), a detection unit (112) adapted for detecting a left/right inversion of the first audio reproduction unit (102) and the second audio reproduction unit (104), and a control unit (114) adapted for controlling the first audio reproduction unit (102) for reproducing the second part of the audio data and for controlling the second audio reproduction unit (104) for reproducing the first part of the audio data upon detecting the left/right inversion.

Abstract translation: 一种用于处理音频数据的设备（100），其中所述设备（100）包括适于再现所述音频数据的第一部分并适于附接到用户的左耳（106）的第一音频再现单元（102） （110），适于再现所述音频数据的第二部分并适于附接到所述用户（110）的右耳（108）的第二音频再现单元（104），适于检测所述用户 第一音频再现单元（102）和第二音频再现单元（104）的左/右反转，以及适于控制用于再现音频的第二部分的第一音频再现单元（102）的控制单元（114） 数据和用于控制第二音频再现单元（104），用于在检测到左/右反转时再现音频数据的第一部分。

公开(公告)号：US08682010B2

公开(公告)日：2014-03-25

申请号：US12970905

申请日：2010-12-16

Applicant: Christophe Marc Macours

Inventor： Christophe Marc Macours

IPC: H04R5/02

CPC classification number: H04S7/306

Abstract: A headphone system includes sound processor which calculates properties of the environment from signals from an internal microphone and an external microphone. The impulse response of the environment may be calculated from the signals received from the internal and external microphones as the user speaks.

Abstract translation: 耳机系统包括声音处理器，其根据来自内部麦克风和外部麦克风的信号来计算环境的属性。 可以根据用户说话从内部和外部麦克风接收的信号来计算环境的脉冲响应。

公开(公告)号：US07539614B2

公开(公告)日：2009-05-26

申请号：US10736939

申请日：2004-05-17

Applicant: Christophe Marc Macours

Inventor： Christophe Marc Macours

IPC: G01L19/14 ,  G01L21/02

CPC classification number: G10L21/038 ,  H03G7/007

Abstract: In a sound reproduction or recording system an audio signal is multiplied by a gain factor (z) which is dependent on the input level (y). The dependence of the gain factor on input level is chosen such that unvoiced phonemes are at least 6 dB, preferably at least 12 dB more enhanced than voiced phonemes, where preferably the average gain is less than 6 dB. This improves the intelligibility.

Abstract translation: 在声音再现或记录系统中，音频信号乘以取决于输入电平（y）的增益因子（z）。 选择增益因子对输入电平的依赖性使得清音音素比浊音素增强至少6dB，优选至少12dB，其中平均增益最好小于6dB。 这提高了清晰度。