公开(公告)号：US20070288235A1

公开(公告)日：2007-12-13

申请号：US11450919

申请日：2006-06-09

申请人： Riitta Vaananen ,  Jarmo Hiipakka

发明人： Riitta Vaananen ,  Jarmo Hiipakka

IPC分类号： G10L19/02

CPC分类号： H04S1/007 ,  H04S1/005

摘要： This invention relates to a device, a method, a software application program, a software application program product and an audio device for processing a digital signal, wherein the digital signal is separated and downsampled into at least two downsampled subband signals, wherein at least one of the at least two downsampled subband signals is equalized, and wherein the at least two downsampled subband signals are upsampled and combined into a digital output signal.

摘要翻译： 本发明涉及一种用于处理数字信号的设备，方法，软件应用程序，软件应用程序产品和音频设备，其中数字信号被分离和下采样为至少两个下采样子带信号，其中至少一个 所述至少两个下采样子带信号被均衡，并且其中所述至少两个下采样子带信号被上采样并组合成数字输出信号。

公开(公告)号：US20060015196A1

公开(公告)日：2006-01-19

申请号：US10537931

申请日：2003-10-08

申请人： Jarmo Hiipakka ,  Samu Kaajas

发明人： Jarmo Hiipakka ,  Samu Kaajas

IPC分类号： G06F17/00

CPC分类号： G06F9/4887

摘要： The invention relates to an audio processing system 1. order to improve the audio processing, the system comprises at least one audio processing component 11, 12, 13 with a group of real-time functions 14 for processing audio data and a group of control functions 15 for processing control signals. The system further comprises at least one processor 16 providing a first process 20 for executing real-time functions 14 of the at least one audio processing component 11, 12, 13 using a basically constant processing power and at least one further process 30 for executing control functions 15 of the at least one audio processing component 11, 12, 13 when needed without affecting the processing power employed for the first process 20. The invention relates equally to a corresponding method and to a corresponding software program product.

摘要翻译： 本发明涉及音频处理系统1.为了改进音频处理，该系统包括至少一个音频处理组件11,12,13，其具有用于处理音频数据的一组实时功能14和一组控制功能 15用于处理控制信号。 该系统还包括至少一个处理器16，其提供第一处理20，用于使用基本上恒定的处理能力来执行至少一个音频处理组件11,12,13的实时功能14，以及用于执行控制的至少一个另外的进程30 当需要时至少一个音频处理组件11,12,13的功能15，而不影响用于第一处理20的处理能力。本发明同样涉及相应的方法和相应的软件程序产品。

公开(公告)号：US08180062B2

公开(公告)日：2012-05-15

申请号：US11755383

申请日：2007-05-30

申请人： Julia Turku ,  Ole Kirkeby ,  Jarmo Hiipakka

发明人： Julia Turku ,  Ole Kirkeby ,  Jarmo Hiipakka

IPC分类号： H04R5/00 ,  H04R5/02

CPC分类号： H04S5/00 ,  H04S7/30 ,  H04S2400/05 ,  H04S2420/07 ,  H04S2420/11

摘要： Aspects of the invention provide methods, computer-readable media, and apparatuses for digital processing of acoustic signals to create a reproduction of a natural or an artificial spatial sound environment. An aspect of the invention supports spatial audio processing such as extracting a center channel in up-mixing stereo sound for multi-channel loudspeaker setup or headphone virtualization. An aspect of the invention also supports directional listening in which sound sources in a desired direction may be amplified or attenuated. Direction and diffuseness parameters for regions of input channels are determined and an extracted channel is extracted from the input channels according to the direction and diffuseness parameters. A gain estimate is estimated for each signal component being fed into the extracted channel and an extracted channel may be synthesized from a base signal and the gain estimate. The input channels may be partitioned into a plurality of time-frequency regions.

摘要翻译： 本发明的方面提供了用于数字处理声信号以产生自然或人造空间声音环境的再现的方法，计算机可读介质和装置。 本发明的一个方面支持空间音频处理，例如提取用于多声道扬声器建立或耳机虚拟化的上混合立体声的中心声道。 本发明的一个方面还支持定向收听，其中期望方向上的声源可以被放大或衰减。 确定输入通道区域的方向和扩散参数，并根据方向和扩散参数从输入通道提取提取的通道。 对于馈送到所提取的信道中的每个信号分量估计增益估计，并且可以从基本信号和增益估计合成提取的信道。 输入通道可以被划分成多个时频区域。

公开(公告)号：US07587254B2

公开(公告)日：2009-09-08

申请号：US10830715

申请日：2004-04-23

申请人： Ole Kirkeby ,  Jarmo Hiipakka

发明人： Ole Kirkeby ,  Jarmo Hiipakka

IPC分类号： G06F17/00 ,  H03G9/00 ,  H03G3/00 ,  G10L19/00

CPC分类号： H04L25/03159 ,  G10L21/02 ,  G10L21/0264

摘要： This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor λ of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

摘要翻译： 本发明描述了一种使用翘曲频谱滤波实时调整数字音频信号的响度和频谱含量的方法。 翘曲处理模块通过由翘曲延迟线的翘曲因子λ确定的多个非线性缩放频带的一组增益来修改数字音频信号的频谱内容。 由翘曲延迟线产生的翘曲延迟线信号由包含多个弯曲有限脉冲响应滤波器的翘曲滤波器块（例如，M频带滤波器）使用在所述多个非线性缩放频带中的单独翘曲频谱滤波来处理 ，其后是动态范围控制/均衡块的常规处理。 本发明描述了另一个创新，即将翘曲处理模块嵌入在双通道正交镜滤波器（QMF）库中，以提高在高采样率下的处理效率。

公开(公告)号：US07363095B2

公开(公告)日：2008-04-22

申请号：US10537931

申请日：2003-10-08

申请人： Jarmo Hiipakka ,  Samu Kaajas

发明人： Jarmo Hiipakka ,  Samu Kaajas

IPC分类号： G06F17/00

CPC分类号： G06F9/4887

摘要： The invention relates to an audio processing system 1. In order to improve the audio processing, the system comprises at least one audio processing component 11, 12, 13 with a group of real-time functions 14 for processing audio data and a group of control functions 15 for processing control signals. The system further comprises at least one processor 16 providing a first process 20 for executing real-time functions 14 of the at least one audio processing component 11, 12, 13 using a basically constant processing power and at least one further process 30 for executing control functions 15 of the at least one audio processing component 11, 12, 13 when needed without affecting the processing power employed for the first process 20. The invention relates equally to a corresponding method and to a corresponding software program product.

摘要翻译： 本发明涉及音频处理系统1。 为了改善音频处理，该系统包括至少一个音频处理组件11,12,13，其具有用于处理音频数据的一组实时功能14和用于处理控制信号的一组控制功能15。 该系统还包括至少一个处理器16，其提供第一处理20，用于使用基本上恒定的处理能力来执行至少一个音频处理组件11,12,13的实时功能14，以及用于执行控制的至少一个另外的进程30 当需要时至少一个音频处理组件11,12,13的功能15，而不影响第一处理20所使用的处理能力。 本发明同样涉及相应的方法和相应的软件程序产品。

公开(公告)号：US20080004729A1

公开(公告)日：2008-01-03

申请号：US11478792

申请日：2006-06-30

申请人： Jarmo Hiipakka

发明人： Jarmo Hiipakka

IPC分类号： G06F17/00 ,  H04R5/00

CPC分类号： H04R5/04

摘要： Provided are improved systems, methods, and computer program products for direct encoding of spatial sound into a directional audio coding format. The direct encoding may also include providing spatial information for a monophonic sound source. The direct encoding of spatial information may be used, for example, in interactive audio applications such as gaming environments and in teleconferencing applications such as multi-party teleconferencing.

摘要翻译： 提供了用于将空间声音直接编码为定向音频编码格式的改进的系统，方法和计算机程序产品。 直接编码还可以包括为单声道声源提供空间信息。 空间信息的直接编码可以用于例如诸如游戏环境的交互式音频应用和诸如多方电话会议的电话会议应用中。

公开(公告)号：US20070299655A1

公开(公告)日：2007-12-27

申请号：US11425809

申请日：2006-06-22

申请人： Laura Laaksonen ,  Jarmo Hiipakka ,  Ville Myllyla ,  Kalle I. Makinen

发明人： Laura Laaksonen ,  Jarmo Hiipakka ,  Ville Myllyla ,  Kalle I. Makinen

IPC分类号： G10L19/14

CPC分类号： G10L21/038

摘要： An apparatus for providing low frequency expansion of speech includes a nonlinear function element, a band-pass filter element and a level control element. The non-linear function element is configured to receive a signal including at least two harmonic components and to produce a signal including at least one lower frequency harmonic component having a lower frequency than a highest frequency component of the at least two harmonic components responsive to the signal including at least two harmonic components. The band-pass filter element is in communication with the non-linear function element and configured to filter the signal including the at least one lower frequency harmonic component. The level control element is configured to apply a level control to alter the filtered signal based on a feature vector associated with an input speech signal.

摘要翻译： 用于提供语音的低频扩展的装置包括非线性功能元件，带通滤波器元件和电平控制元件。 非线性功能元件被配置为接收包括至少两个谐波分量的信号，并产生包括至少一个低频谐波分量的信号，该低频谐波分量的频率低于至少两个谐波分量的最高频率分量 信号包括至少两个谐波分量。 带通滤波器元件与非线性功能元件通信，并且被配置为对包括至少一个较低频率谐波分量的信号进行滤波。 电平控制元件被配置为施加电平控制以基于与输入语音信号相关联的特征向量来改变滤波信号。

公开(公告)号：US20070260780A1

公开(公告)日：2007-11-08

申请号：US11321796

申请日：2006-04-11

申请人： Jarmo Hiipakka ,  Seppo Ingalsuo ,  Pekka Wallius

发明人： Jarmo Hiipakka ,  Seppo Ingalsuo ,  Pekka Wallius

IPC分类号： G06F5/00

CPC分类号： G06F9/544 ,  G06F1/3203

摘要： A media subsystem of a processing element includes a plurality of elements and a latency manager. The plurality of elements are capable of processing media data including a plurality of instances wherein a first element inserts a length of media data into buffer(s) from which a second element thereafter reads the length of media data for subsequent output from the media subsystem. The latency manager is capable of determining a latency requirement of the media subsystem, and then dynamically tuning the length of media data inserted into the buffer(s) based upon the latency requirement, including increasing or decreasing the length of media data inserted into the buffer(s) during one or more instances(s).

摘要翻译： 处理元件的媒体子系统包括多个元件和延迟管理器。 多个元件能够处理包括多个实例的媒体数据，其中第一元素将一定长度的媒体数据插入到缓冲器中，然后第二元素从其中读取媒体数据的长度以便随后从媒体子系统输出。 延迟管理器能够确定媒体子系统的等待时间要求，然后基于等待时间要求动态调整插入到缓冲器中的媒体数据的长度，包括增加或减少插入到缓冲器中的媒体数据的长度 在一个或多个实例中。