公开(公告)号：US5890126A

公开(公告)日：1999-03-30

申请号：US815318

申请日：1997-03-10

Applicant: Eric Lindemann

Inventor： Eric Lindemann

IPC: G10L7/06

CPC classification number: G10L21/003 ,  G10H7/04

Abstract: Apparatus for simultaneously decompressing and interpolating compressed audio data. The compressed audio data is stored in differential log format, meaning that the difference between each two consecutive data points is taken and the log of the difference calculated to form each compressed data point. To efficiently decompress and interpolate the compressed data, advantage is taken of the fact that addition of logs is equivalent to multiplication of linear values. Thus the log of an interpolation factor is added to each compressed data point prior to taking the inverse log of the sum. An integrator block completes the interpolation and decompression of the data.

Abstract translation: 用于同时解压缩和内插压缩音频数据的装置。 压缩音频数据以差分日志格式存储，这意味着采用每两个连续数据点之间的差异，并计算差异日志以形成每个压缩数据点。 为了有效地解压缩和内插压缩数据，优点在于添加日志等价于线性值的乘法。 因此，在获取总和的逆对数之前，将内插因子的对数添加到每个压缩数据点。 积分器块完成数据的插值和解压缩。

公开(公告)号：US5651071A

公开(公告)日：1997-07-22

申请号：US123503

申请日：1993-09-17

Applicant: Eric Lindemann ,  John Laurence Melanson

Inventor： Eric Lindemann ,  John Laurence Melanson

IPC: H04R25/00

CPC classification number: H04R25/552 ,  H04R25/505

Abstract: In this invention noise in a binaural hearing aid is reduced by analyzing the left and right digital audio signals to produce left and right signal frequency domain vectors and thereafter using digital signal encoding techniques to produce a noise reduction gain vector. The gain vector can then be multiplied against the left and right signal vectors to produce a noise reduced left and right signal vector. The cues used in the digital encoding techniques include directionality, short term amplitude deviation from long term average, and pitch. In addition, a multidimensional gain function based on directionality estimate and amplitude deviation estimate is used that is more effective in noise reduction than simply summing the noise reduction results of directionality alone and amplitude deviations alone. As further features of the invention, the noise reduction is scaled based on pitch-estimates and based on voice detection.

Abstract translation: 在本发明中，通过分析左和右数字音频信号以产生左和右信号频域矢量并且此后使用数字信号编码技术产生降噪增益矢量来减少双耳助听器中的噪声。 然后可以将增益矢量与左和右信号矢量相乘以产生噪声减小的左和右信号矢量。 在数字编码技术中使用的线索包括方向性，短期幅度偏离长期平均值和音调。 另外，使用基于方向性估计和幅度偏差估计的多维增益函数，其在降噪方面比简单地将方向性的降噪结果和单独的幅度偏差相加得更有效。 作为本发明的进一步特征，基于音调估计并基于语音检测来缩放噪声。

公开(公告)号：US5479522A

公开(公告)日：1995-12-26

申请号：US123499

申请日：1993-09-17

Applicant: Eric Lindemann ,  John L. Melanson

Inventor： Eric Lindemann ,  John L. Melanson

IPC: H04R25/00

CPC classification number: H04R25/356 ,  H04R25/552 ,  H04R25/558 ,  G10L21/0364 ,  H04R25/505 ,  H04R25/554

Abstract: This invention relates to a hearing enhancement system having an ear device for each of the wearer's ears, each ear device has a sound transducer, or microphone, and a sound reproducer, or speaker, and associated electronics for the microphone and speaker. Further, the electronic enhancement of the audio signals is performed at a remote digital signal processor (DSP) likely located in a body pack worn somewhere on the body by the user. There is a down-link from each ear device to the (DSP) and an up-link from the DSP to each ear device. The DSP digitally interactively processes the audio signals for each ear based on both of the audio signals received from each ear device. In other words, the enhancement of the audio signal for the left ear is based on the both the right and left audio signals received by the DSP.In addition digital filters implemented at the DSP have a linear phase response so that time relationships at different frequencies are preserved. The digital filters have a magnitude and phase response to compensate for phase distortions due to analog filters in the signal path and due to the resonances and nulls of the ear canal.

Abstract translation: 本发明涉及一种听力增强系统，其具有用于佩戴者耳朵中的每一个的耳朵装置，每个耳朵装置具有声音换能器或麦克风，以及声音再现器或扬声器以及用于麦克风和扬声器的相关电子装置。 此外，音频信号的电子增强在可能位于用户身体某处的身体组件中的远程数字信号处理器（DSP）处执行。 从每个耳朵设备到（DSP）的下行链路以及从DSP到每个耳朵设备的上行链路。 DSP基于从每个耳朵设备接收的两个音频信号数字地交互地处理每个耳朵的音频信号。 换句话说，左耳音频信号的增强是基于由DSP接收的右和左音频信号。 此外，在DSP处实现的数字滤波器具有线性相位响应，从而保持不同频率的时间关系。 数字滤波器具有幅度和相位响应，以补偿由信号路径中的模拟滤波器引起的相位失真，并且由于耳道的共振和零点。

公开(公告)号：US20150352491A1

公开(公告)日：2015-12-10

申请号：US14731133

申请日：2015-06-04

Applicant: James M. Valentine ,  Jeffrey Michael Broderick ,  Thomas V. Eldredge ,  Saiprasad Jangiti ,  Eric Lindemann

Inventor： James M. Valentine ,  Jeffrey Michael Broderick ,  Thomas V. Eldredge ,  Saiprasad Jangiti ,  Eric Lindemann

IPC: B01D53/86 ,  B01D53/90 ,  B01J19/24

CPC classification number: B01D53/8631 ,  B01D53/8625 ,  B01D53/90 ,  B01D53/9409 ,  B01D2251/2062 ,  B01D2251/2067 ,  B01D2257/40 ,  B01D2259/124 ,  B01J4/002 ,  B01J4/008 ,  B01J19/2405 ,  B01J2219/00063 ,  B01J2219/00164 ,  B01J2219/00186 ,  B01J2219/00202 ,  B01J2219/00231

Abstract: A system for vaporizing and optionally decomposing a reagent, such as aqueous ammonia or urea, which is useful for NOx reduction, includes a cyclonic decomposition duct, wherein the duct at its inlet end is connected to an air inlet port and a reagent injection lance. The air inlet port is in a tangential orientation to the central axis of the duct. The system further includes a metering valve for controlling the reagent injection rate. A method for vaporizing and optionally decomposing a reagent includes providing a cyclonic decomposition duct which is connected to an air inlet port and an injection lance, introducing hot gas through the air inlet port in a tangential orientation to the central axis of the duct, injecting the reagent axially through the injection lance into the duct; and adjusting the reagent injection rate through a metering valve.

Abstract translation: 用于蒸发和任选分解用于NOx还原的试剂例如氨水或尿素的系统包括气旋分解管道，其中其入口端处的管道连接到空气入口和试剂注入喷枪。 进气口与导管的中心轴线相切。 该系统还包括用于控制试剂注射速率的计量阀。 一种蒸发和任选地分解试剂的方法包括提供一个气流分解管道，其连接到空气入口和喷射枪，将热气体以切线方向引导通过空气入口到管道的中心轴线， 试剂轴向穿过注射枪进入管道; 并通过计量阀调节试剂注射速率。

公开(公告)号：US20070137466A1

公开(公告)日：2007-06-21

申请号：US11637596

申请日：2006-12-12

Applicant: Eric Lindemann

Inventor： Eric Lindemann

IPC: G10H1/02 ,  G10H7/00 ,  G01P3/00

CPC classification number: G10H7/006 ,  G10H1/0066 ,  G10H2210/201 ,  G10H2210/225 ,  G10H2250/211 ,  G10H2250/495 ,  G10H2250/615

Abstract: The present synthesizer generates an underlying spectrum, pitch and loudness for a sound to be synthesized, and then combines the underlying spectrum, pitch and loudness with stored Spectral, Pitch, and Loudness Fluctuations and noise elements. The input to the synthesizer is typically a MIDI stream. A MIDI preprocess block processes the MIDI input and generates the signals needed by the synthesizer to generate output sound phrases. The synthesizer comprises a harmonic synthesizer block (which generates an output representing the tonal audio portion of the output sound), an Underlying Spectrum, Pitch, and Loudness (which takes pitch and loudness and uses stored algorithms to generate the slowly varying portion of the output sound) and a Spectral, Pitch, and Loudness Fluctuation portion (which generates the quickly varying portion of the output sound by selecting and combining Spectral, Pitch, and Loudness Fluctuation segments stored in a database). A specialized analysis process is used to derive the formulas used by the Underlying Spectrum, Pitch, and Loudness and to generate and store the Spectral, Pitch, and Loudness Fluctuation segments stored in the database.

Abstract translation: 本合成器为合成的声音生成基础频谱，音调和响度，然后将基础频谱，音调和响度与存储的频谱，音高和响度波动和噪声元素相结合。 合成器的输入通常是MIDI流。 MIDI预处理块处理MIDI输入并生成合成器所需的信号以产生输出声音短语。 合成器包括谐波合成器块（其产生表示输出声音的音调音频部分的输出），基础频谱，音高和响度（其​​采用音调和响度，并且使用存储的算法来生成输出的缓慢变化部分 声音）和光谱，音高和响度波动部分（通过选择和组合存储在数据库中的光谱，音高和响度波动段来产生输出声音的快速变化部分）。 使用专门的分析过程来导出基础频谱，音高和响度所使用的公式，并生成和存储数据库中存储的谱，音高和响度波动段。

公开(公告)号：US5744742A

公开(公告)日：1998-04-28

申请号：US808676

申请日：1997-02-28

Applicant: Eric Lindemann ,  Jeffrey Barish

Inventor： Eric Lindemann ,  Jeffrey Barish

IPC: G10H5/00 ,  G10H7/10 ,  G10H1/00 ,  G10H1/12

CPC classification number: G10H5/007 ,  G10H7/10 ,  G10H2250/015 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/081 ,  G10H2250/095 ,  G10H2250/101 ,  G10H2250/235 ,  G10H2250/285 ,  G10H2250/481 ,  G10H2250/545 ,  G10H2250/611 ,  Y10S84/09

Abstract: A parametric signal modeling musical tone synthesizer utilizes a multidimensional filter coefficient space consisting of many sets of filter coefficients to model an instrument. These sets are smoothly interpolated over pitch, intensity, and time. The filter excitation for a particular note is derived from a collection of single period excitations, which form a multidimensional excitation space, which is also smoothly interpolated over pitch, intensity and time. The synthesizer includes effective modeling of attacks of tones, and the noise component of a tone is modelled separately from the pitched component. The input control signals may include initial pitch and intensity, or the intensity may be time-varying. A variety of instruments may be specified.

Abstract translation: 参数信号建模乐音合成器利用由许多滤波器系数组成的多维滤波器系数空间来对仪器建模。 这些组在音高，强度和时间上顺利插值。 用于特定音符的滤波器激励源自单周期激励的集合，其形成多维激励空间，其也在音调，强度和时间上平滑地内插。 合成器包括对音调的有效建模，并且音调的噪声分量与俯仰分量分开建模。 输入控制信号可以包括初始音高和强度，或者强度可以是时变的。 可以指定各种仪器。

公开(公告)号：US20070137465A1

公开(公告)日：2007-06-21

申请号：US11633675

申请日：2006-12-04

Applicant: Eric Lindemann

Inventor： Eric Lindemann

IPC: G10H1/02 ,  G10H7/00 ,  G01P3/00

CPC classification number: G10H7/008 ,  G10H1/0066 ,  G10H2240/131

Abstract: The present synthesizer includes functionality for changing over from a current note to the following notes that results in natural and expressive combinations and transitions. The method of the present invention incorporates an delay (actual, functional, or look ahead) between receiving control data inputs and generating an output sound. This period of delay is used to modify how notes will be played according to control data inputs for later notes. The input to the synthesizer is typically a time-varying MIDI stream, which may be provided by a musician or a MIDI sequencer from stored data. An actual delay occurs when the synthesizer receives a MIDI stream and buffers it while looking ahead for changeovers between notes. A functional delay occurs in a system in which the synthesizer has knowledge of note changeovers ahead of time. A look ahead delay occurs when the synthesizer queries the sequencer for information about the stored sequence ahead of when the synthesizer needs to generate the output for the sequence.

Abstract translation: 本合成器包括从当前音符切换到以下音符的功能，导致自然和表达的组合和转换。 本发明的方法包括在接收控制数据输入和产生输出声音之间的延迟（实际的，功能性的或前瞻性的）。 这段延迟用于修改如何根据后续笔记的控制数据输入播放音符。 合成器的输入通常是时变MIDI流，其可以由音乐家或MIDI定序器从存储的数据提供。 当合成器接收到MIDI流并缓冲它，同时向前看，在音符之间切换时，会发生实际的延迟。 在合成器提前了解笔记转换的系统中发生功能延迟。 当合成器需要生成序列的输出时，合成器在序列发生器查询有关存储的序列的信息前，会出现一个提前延迟。

公开(公告)号：US06792114B1

公开(公告)日：2004-09-14

申请号：US09413732

申请日：1999-10-06

Applicant: James Mitchell Kates ,  John Laurence Melanson ,  Eric Lindemann

Inventor： James Mitchell Kates ,  John Laurence Melanson ,  Eric Lindemann

IPC: H04R2900

CPC classification number: H04R25/70 ,  A61B5/121 ,  H04R25/305 ,  H04R25/505

Abstract: A digital hearing aid according to the present invention is capable of measuring its own performance. The hearing aid includes a test signal generator for feeding a test signal into the hearing aid amplifier. The response to the test signal is acquired at a specific point in the hearing aid, depending upon what aspect of performance is to be measured. Various elements of the hearing aid and/or the hearing aid feedback may be bypassed. The hearing aid further includes the capability of initializing hearing aid parameters based upon the performance measurements. The measurement and initialization capability may be entirely integral to the hearing aid, or an external processor may be used to download the measurement program and the run time program, and assist in computing the parameters.

Abstract translation: 根据本发明的数字助听器能够测量其自身的性能。 助听器包括用于将测试信号馈送到助听放大器中的测试信号发生器。 取决于要测量的性能方面，在助听器的特定点获取对测试信号的响应。 可以绕过助听器和/或助听器反馈的各种元件。 助听器还包括基于性能测量来初始化助听器参数的能力。 测量和初始化能力可以完全与助听器成一体，或者可以使用外部处理器来下载测量程序和运行时程序，并协助计算参数。

公开(公告)号：US06316710B1

公开(公告)日：2001-11-13

申请号：US09406459

申请日：1999-09-27

Applicant: Eric Lindemann

Inventor： Eric Lindemann

IPC: G10H700

CPC classification number: G10H7/02 ,  G10H2210/095 ,  G10H2240/056

Abstract: The present invention describes a device and methods for synthesizing a musical audio signal. The invention includes a device for storing a collection of sound segments taken from idiomatic musical performances. Some of these sound segments include transitions between musical notes such as the slur from the end of one note to the beginning of the next. Much of the complexity and expressivity in musical phrasing is associated with the complex behavior of these transition segments. The invention further includes a device for generating a sequence of sound segments in response to an input control sequence—e.g. a MIDI sequence. The sound segments are associated with musical gesture types. The gesture types include attack, release, transition, and sustain. The sound segments are further associated with musical gesture subtypes. Large upward slur, small upward slur, large downward slur, and small downward slur are examples of subtypes of the transition gesture type. Event patterns in the input control sequence lead to the generation of a sequence of musical gesture types and subtypes, which in turn leads to the selection of a sequence of sound segments. The sound segments are combined to form an audio signal and played out by a sound segment player. The sound segment player pitch-shifts and intensity-shifts the sound segments in response to the input control sequence.

Abstract translation: 本发明描述了一种用于合成音乐音频信号的装置和方法。 本发明包括一种用于存储从惯用音乐表演中取出的声音段的集合的装置。 这些声音段中的一些包括音符之间的转换，例如从一个音符的结尾到下一个音符的开始的音调。 音乐短语中的许多复杂性和表现力与这些过渡段的复杂行为有关。 本发明还包括一种用于响应于输入控制序列产生声音段序列的装置， 一个MIDI序列。 声音段与音乐手势类型相关联。 手势类型包括攻击，释放，转换和维持。 声音段进一步与音乐手势子类型相关联。 大型向上的sl ur，小向上的sl ur，大的向下的sl ur和小的向下的sl ur是过渡手势类型的亚型的例子。 输入控制序列中的事件模式导致产生一系列音乐手势类型和子类型，这又导致对一段声音段的选择。 声音段被组合以形成音频信号并由声音段播放器播放。 响应于输入控制序列，声音段播放器对音段进行间距移位和强度移位。

公开(公告)号：US06298322B1

公开(公告)日：2001-10-02

申请号：US09306256

申请日：1999-05-06

Applicant: Eric Lindemann

Inventor： Eric Lindemann

IPC: G10L1902

CPC classification number: G10L19/02

Abstract: Tonal audio signals can be modeled as a sum of sinusoids with time-varying frequencies, amplitudes, and phases. An efficient encoder and synthesizer of tonal audio signals is disclosed. The encoder determines time-varying frequencies, amplitudes, and, optionally, phases for a restricted number of dominant sinusoid components of the tonal audio signal to form a dominant sinusoid parameter sequence. These components are removed from the tonal audio signal to form a residual tonal signal. The residual tonal signal is encoded using a residual tonal signal encoder (RTSE). In one embodiment, the RTSE generates a vector quantization codebook (VQC) and residual codebook sequence (RCS). The VQC may contain time-domain residual waveforms selected from the residual tonal signal, synthetic time-domain residual waveforms with magnitude spectra related to the residual tonal signal, magnitude spectrum encoding vectors, or a combination of time-domain waveforms and magnitude spectrum encoding vectors. The tonal audio signal synthesizer uses a sinusoidal oscillator bank to synthesize a set of dominant sinusoid components from the dominant sinusoid parameter sequence generated during encoding. In one embodiment, a residual tonal signal is synthesized using a VQC and RCS generated by the RTSE during encoding. If the VQC includes time-domain waveforms, an interpolating residual waveform oscillator may be used to synthesize the residual tonal signal. The synthesized dominant sinusoids and synthesized residual tonal signal are summed to form the synthesized tonal audio signal.

Abstract translation: 色调音频信号可以被建模为具有时变频率，幅度和相位的正弦曲线的总和。 公开了一种有效的音调信号编码器和合成器。 编码器确定音调音频信号的有限数量的主要正弦分量的时变频率，振幅以及可选的相位，以形成主要的正弦参数序列。 这些组件从音调音频信号中去除以形成残余音调信号。 剩余音调信号使用残差音调信号编码器（RTSE）进行编码。 在一个实施例中，RTSE生成矢量量化码本（VQC）和残留码本序列（RCS）。 VQC可以包含从残差音调信号中选择的时域残差波形，与残差音调信号相关的幅度谱的合成时域残差波形，幅度谱编码矢量，或时域波形和幅度谱编码矢量的组合 。 色调音频信号合成器使用正弦振荡器组来从编码期间产生的主要正弦参数序列合成一组主要的正弦分量。 在一个实施例中，使用在编码期间由RTSE生成的VQC和RCS来合成残余音调信号。 如果VQC包括时域波形，则可以使用内插残差波形振荡器来合成残余音调信号。 将合成的主要正弦曲线和合成的残差音调信号相加，形成合成的音调信号。