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    • 1. 发明授权
    • Method and apparatus for processing two or more initially decoded audio signals received or replayed from a bitstream
    • 用于处理从比特流接收或重放的两个或更多个初始解码的音频信号的方法和装置
    • US08082050B2
    • 2011-12-20
    • US10536539
    • 2003-11-24
    • Jürgen SchmidtJens SpilleErnst F. SchröderJohannes Böhm
    • Jürgen SchmidtJens SpilleErnst F. SchröderJohannes Böhm
    • G06F17/00
    • H04S3/008H04S1/007
    • In the MPEG-4 standard ISO/IEC 14496:2001 several audio objects that can be coded with different MPEG-4 format coding types can together form a composed audio system representing a single soundtrack from the several audio substreams. In a receiver the multiple audio objects are decoded separately, but not directly played back to a listener. Instead, transmitted instructions for mixdown are used to prepare a single soundtrack. Mixdown conflicts can occur in case the audio signals to be combined have different channel numbers or configurations. According to the invention an additional audio channel configuration node is used that tags the correct channel configuration information items to the decoded audio data streams to be presented. The invention enables the content provider to set the channel configuration in such a way that the presenter at receiver side can produce a correct channel presentation under all circumstances. An escape code value in the channel configuration data facilitates correct handling of not yet defined channel combinations.
    • 在MPEG-4标准ISO / IEC 14496:2001中,可以用不同MPEG-4格式编码类型编码的几个音频对象可以一起形成代表来自多个音频子流的单个音轨的组合音频系统。 在接收机中,多个音频对象被单独解码,但不直接回放到收听者。 相反,用于混音的传输指令用于准备单个音轨。 要组合的音频信号具有不同的通道号或配置时,可能会发生混频冲突。 根据本发明,使用附加的音频通道配置节点,其将正确的信道配置信息项标记到要呈现的解码音频数据流。 本发明使得内容提供商能够以这样一种方式设置频道配置,使得接收机侧的演示者可以在任何情况下产生正确的频道呈现。 通道配置数据中的转义码值有助于正确处理尚未定义的通道组合。
    • 2. 发明授权
    • System for storing and transmitting home network system data
    • 用于存储和发送家庭网络系统数据的系统
    • US06996613B1
    • 2006-02-07
    • US09830104
    • 1999-10-21
    • Ernst F. Schröder
    • Ernst F. Schröder
    • G06F15/173
    • H04N5/77H04N21/43615H04N21/43632
    • The invention specifies a system having a plurality of devices which are connected to one another via an IEEE 1394 interface and one of which contains a control unit which, when operated appropriately by a user, polls system data for devices in this system via the interface and passes this system data to an output unit of this device. The device having the output unit is, by way of example, a set-top box having a microprocessor which a user uses to poll system data for the devices, which contains, in particular the input and output characteristics of the latter, via the interface and which the user can use to store this system data on a smart card by means of a write/read device. Alternatively or at the same time, the system data can be shown on a display or transmitted to a desired address via a modem connection. As a result, the user of the system can receive expert advice from a specialist dealer or a customer service point regarding which devices he can best add to his system; or if a point of failure or faults arise.
    • 本发明规定了一种具有多个设备的系统,该多个设备经由IEEE 1394接口相互连接,并且其中一个设备包含一个控制单元,该控制单元当由用户适当地操作时,经由接口轮询该系统中的设备的系统数据, 将该系统数据传送到该设备的输出单元。 具有输出单元的装置例如是具有微处理器的机顶盒,其中用户使用微处理器来轮询用于装置的系统数据,其特别包括后者的输入和输出特性经由接口 并且用户可以通过写/读设备将该系统数据存储在智能卡上。 或者或同时,系统数据可以在显示器上显示或通过调制解调器连接发送到期望的地址。 因此,系统的用户可以从专业经销商或客户服务点接收有关他最能添加到他的系统中的哪些设备的专家建议; 或者如果出现故障或故障。
    • 3. 发明授权
    • Determination of the presence of additional coded data in a data frame
    • 确定数据帧中附加编码数据的存在
    • US07334176B2
    • 2008-02-19
    • US10495838
    • 2002-11-02
    • Ernst F. Schröder
    • Ernst F. Schröder
    • H04L1/14
    • G10L19/167
    • An mp3-standard bitstream is formatted into a sequence of fixed-length data frames. These include headers, side information, main information and a remaining data field without generally defined information denoted as ‘ancillary data’. The mp3PRO format is an extension of the mp3 format, wherein the additional mp3PRO data are transferred in the ancillary data fields. In various applications, e.g. Internet music search machines, a necessity arises for a fast determination of the bitstream types. Such determination is normally executed using an mp3PRO decoder. However, because the frame header does not contain a corresponding pointer to the start address of the ancillary data field, an mp3PRO decoder must first completely decode at least one data frame according to the mp3 standard in order to find the end address of the mp3 data and thereby the following start address of the mp3PRO data in that data frame. Thereafter the mp3PRO decoder must examine the data following in the data stream for characteristics that are typical for mp3PRO additional information. The invention discloses how the bitstream type can be determined without using mp3 decoding and without using an mp3PRO decoder.
    • mp3标准比特流被格式化成固定长度数据帧的序列。 这些包括头部,侧面信息,主要信息和没有被表示为“辅助数据”的通常被定义的信息的剩余数据字段。 mp3PRO格式是mp3格式的扩展,其中附加的mp3PRO数据在辅助数据字段中传输。 在各种应用中,例如 互联网音乐搜索机,需要快速确定比特流类型。 这种确定通常使用mp3PRO解码器来执行。 然而,由于帧头不包含与辅助数据字段的起始地址相对应的指针,所以mp3PRO解码器必须首先根据mp3标准完全解码至少一个数据帧,以便找到mp3数据的结束地址 从而在该数据帧中的mp3PRO数据的起始地址。 此后,mp3PRO解码器必须检查数据流中跟随的数据,以获取典型的mp3PRO附加信息的特性。 本发明公开了如何在不使用mp3解码而不使用mp3PRO解码器的情况下确定比特流类型。
    • 4. 发明授权
    • Voice control system with a microphone array
    • 带麦克风阵列的语音控制系统
    • US06868045B1
    • 2005-03-15
    • US09660381
    • 2000-09-12
    • Ernst F. Schröder
    • Ernst F. Schröder
    • G10L15/20G10L15/26G10L21/0216H04R3/00G10K11/00
    • H04R3/005G10L15/26G10L2021/02166H04R2201/401
    • Voice control systems are used in diverse technical fields. In this case, the spoken words are detected by one or more microphones and then fed to a speech recognition system. In order to enable voice control even from a relatively great distance, the voice signal must be separated from interfering background signals. This can be effected by spatial separation using microphone arrays comprising two or more microphones. In this case, it is advantageous for the individual microphones of the microphone array to be distributed spatially over the greatest possible distance. In an individual consumer electronics appliance, however, the distances between the individual microphones are limited on account of the dimensions of the appliance. Therefore, the voice control system according to the invention comprises a microphone array having a plurality of microphones which are distributed between different appliances, in which case the signals generated by the microphones can be transmitted to the central speech recognition unit, advantageously via a bidirectional network based on an IEEE 1394 bus.
    • 语音控制系统用于多种技术领域。 在这种情况下,口语由一个或多个麦克风检测,然后馈送到语音识别系统。 为了即使在距离较远的情况下也能进行语音控制,语音信号必须与干扰背景信号分开。 这可以通过使用包括两个或更多个麦克风的麦克风阵列的空间分离来实现。 在这种情况下,有利的是麦克风阵列的各个麦克风在最大可能的距离上在空间上分布。 然而,在个人消费电子设备中,由于设备的尺寸,各个麦克风之间的距离受到限制。 因此,根据本发明的语音控制系统包括具有分布在不同设备之间的多个麦克风的麦克风阵列,在这种情况下,由麦克风产生的信号可以有利地通过双向网络传输到中央语音识别单元 基于IEEE 1394总线。
    • 5. 发明授权
    • Method and apparatus for decoding a coded digital audio signal which is arranged in frames containing headers
    • 用于对编码的数字音频信号进行解码的方法和装置,其被布置在包含报头的帧中
    • US07342944B2
    • 2008-03-11
    • US10493286
    • 2002-10-11
    • Ernst F SchröderJohannes Böhm
    • Ernst F SchröderJohannes Böhm
    • H04J3/06
    • G10L19/167
    • With audio data reduction on the basis of ISO/IEC standard 11172-3, a frame length varying by 8 bits is used at a sampling frequency of 44.1 kHz in order to arrive, on average, at a particular fixed data rate. The lengthening of a data frame is signalled by a padding bit in the header of the frames. The invention dispenses with evaluation of the padding bit. Instead, the mean frame length L is calculated, L is rounded down to the next integer, for the subsequent frame it is first established whether the expected sync word for this frame appears, and, if this is so, this frame is decoded without taking into account the padding bit, but if the expected sync word for this frame does not appear, the decoding of the frame is started one 8-bit later without taking into account the padding bit.
    • 基于ISO / IEC标准11172-3的音频数据缩减,以44.1kHz的采样频率使用8比特的帧长度,以平均以特定的固定数据速率到达。 数据帧的延长由帧的报头中的填充位发出信号。 本发明省略了对填充位的评估。 相反,平均帧长度L被计算,L被舍入到下一个整数,对于首先建立的后续帧,是否出现该帧的预期同步字,并且如果是这样,则该帧被解码而不采取 考虑到填充位,但是如果没有出现该帧的预期同步字,则在不考虑填充位的情况下,8位后来开始帧的解码。
    • 7. 发明授权
    • Synchronization of a two-channel audio decoder with a multichannel audio decoder
    • 双通道音频解码器与多声道音频解码器的同步
    • US06230141B1
    • 2001-05-08
    • US09205464
    • 1998-12-04
    • Johannes BöhmErnst F. Schröder
    • Johannes BöhmErnst F. Schröder
    • G11B27031
    • G11B27/10G11B27/105G11B2220/2562H04N7/56H04N21/4307H04N21/8106
    • Dolby AC-3 and MPEG-2 audio permit the transmission of audio signals with more than two independent audio channels. If a reproduction device has only a two-channel audio decoder (DEC1) then an external multi-channel audio decoder (DEC2) can be used for multi-channel sound reproduction. If the audio reproduction at the same time accompanies video reproduction, then a synchronization method is required in order to achieve lip synchronism between picture and sound. According to the invention, for the synchronization of a first decoder (DEC1), which merely has two-channel compatibility, with a second decoder, which has multi-channel compatibility, a counting variable is allocated a value (F) which is produced from system parameters such as the data coding method used, the transmission speed and/or the data rate. The data are received by the first decoder and output by the latter to the second decoder, the counting variable being decremented or incremented respectively for a specific volume of data. When the counting variable reaches a value corresponding to the beginning of the decoding by the second decoder, the first decoder then begins decoding the data.
    • 杜比AC-3和MPEG-2音频允许具有两个以上独立音频通道的音频信号传输。 如果再现设备仅具有双声道音频解码器(DEC1),则外部多声道音频解码器(DEC2)可以用于多声道声音再现。 如果音频再现同时伴随视频再现,则需要同步方法来实现图像和声音之间的唇部同步。 根据本发明,对于具有双信道兼容性的第一解码器(DEC1)与具有多信道兼容性的第二解码器的同步,计数变量被分配为(F)从 诸如使用的数据编码方法,传输速度和/或数据速率的系统参数。 数据由第一解码器接收并由后者输出到第二解码器,对于特定数据量,计数变量分别递减或递增。 当计数变量达到与第二解码器解码开始对应的值时,第一解码器开始对数据进行解码。
    • 8. 发明授权
    • Method and apparatus for encoding and for decoding a digital information signal
    • 用于对数字信息信号进行编码和解码的方法和装置
    • US06903664B2
    • 2005-06-07
    • US10372515
    • 2003-02-24
    • Ernst F. SchröderJohannes Böhm
    • Ernst F. SchröderJohannes Böhm
    • G11B20/12G10L19/00G10L19/16H03M7/30H03M7/00
    • G10L19/167
    • Original digital audio signals are represented as PCM sample values wherein the distance between the values corresponds to the sampling frequency. Digital signals can have a length that is an integer multiple only of this time element. In particular coded digital audio signals are processed block-based, leading to a total length that is a multiple only of the block unit. According to the invention, information about the exact length of the original signal is transferred together with the encoded audio information. Additionally, an information value can be transferred that represents the total encoder and/or decoder delay. The decoder extracts these items of information and adjusts the total length of the decoded signal by cutting off samples from the decoded program or track.
    • 原始数字音频信号被表示为PCM采样值,其中值之间的距离对应于采样频率。 数字信号的长度只能是这个时间元素的整数倍。 特别地,编码的数字音频信号是基于块的,导致总长度仅为块单元的倍数。 根据本发明,关于原始信号的精确长度的信息与编码的音频信息一起传送。 另外,可以传送表示总编码器和/或解码器延迟的信息值。 解码器提取这些信息项,并通过从解码的程序或轨道切断样本来调整解码信号的总长度。
    • 9. 发明授权
    • Method and an apparatus for sampling-rate conversion of audio signals
    • 用于对音频信号进行采样率转换的方法和装置
    • US06681209B1
    • 2004-01-20
    • US09309704
    • 1999-05-11
    • Jürgen SchmidtErnst F. Schröder
    • Jürgen SchmidtErnst F. Schröder
    • G10L1900
    • H03H17/0621
    • Generally, performing sampling-rate conversion from a higher sampling frequency fs1 to a lower sampling frequency fs2 results in aliasing. It is known to use a low-pass filter, known as anti-alias filter, for avoiding this alias distortion. Its effect is to remove spectral contents above fs2/2 from the digital signal. According to the invention these signal parts are suppressed at spectral decoding resulting in a bandwidth of the signal to be re-sampled which is less than half of the second sampling frequency fs2. This can be done for MPEG encoded audio signals by limiting the decoding to a certain number of subbands, for DOLBY AC-3 encoded audio signals by setting certain spectral lines to zero at decoding. The inventive method not only totally removes the processing power needed for calculating an anti-alias filter, but also limits the decoding work needed.
    • 通常,从较高采样频率fs1到较低采样频率fs2执行采样率转换导致混叠。 已知使用称为抗混叠滤波器的低通滤波器来避免这种混叠失真。 其效果是从数字信号中去除fs2 / 2以上的频谱内容。 根据本发明,这些信号部分在频谱解码时被抑制,导致待重新采样的信号的带宽小于第二采样频率fs2的一半。 这可以通过将解码限制到一定数量的子带,对于DOLBY AC-3编码的音频信号,通过在解码时将某些谱线设置为零来对MPEG编码的音频信号进行。 本发明的方法不仅完全消除了计算抗混叠滤波器所需的处理能力,而且限制了所需的解码工作。