| 序号 | 专利名 | 申请号 | 申请日 | 公开（公告）号 | 公开（公告）日 | 发明人 |
|---|---|---|---|---|---|---|
| 1 | 兴趣区间抽取装置、兴趣区间抽取方法 | CN201180012516.2 | 2011-10-28 | CN102782750B | 2015-04-01 | 小沼知浩; 川西亮一; 上野山努 |
| 一种兴趣区间抽取装置（104），根据动态图像文件中所包含的音频信号来抽取包括指定时刻（T0）的用户的兴趣区间，该兴趣区间抽取装置（104）具有：接口装置（109），取得指定时刻（T0）；似然度向量生成部（202），对于音频信号的每个第1单位区间计算表现多种音素各自的特征的各个锚模型（Ar）的似然度，并生成以计算出的各个似然度为分量的似然度向量（F）；以及兴趣区间抽取部（209），根据似然度向量（F）计算成为兴趣区间的候选的第1特征区间，并抽取包括指定时刻（T0）的第1特征区间的一部分区间作为兴趣区间。 | ||||||
| 2 | 音频播放器支持的音频文件格式的测试设备及方法 | CN200810304752.1 | 2008-10-06 | CN101714380A | 2010-05-26 | 蒲小满; 邱金泉 |
| 本发明提供一种音频播放器支持的音频文件格式的测试方法及测试设备。该测试方法包括步骤：在该测试设备上输出每一音频文件至该音频播放器，每一音频文件具有一初始标识符对应每一音频文件所属的音频格式；利用该测试设备接收该音频播放器播放每一音频文件时输出的声音信号；在该测试设备上将接收的该音频播放器播放每一音频文件时输出的声音信号录制为每一声音文件；处理每一录制的声音文件以获取每一录制的声音文件的标识符；比较每一录制的声音文件的标识符与每一音频文件的初始标识符，根据比较结果确定该音频播放器支持的音频格式；以及输出该音频播放器支持的音频格式。本发明提供的测试设备及方法能够降低人力成本，提高测试效率。 | ||||||
| 3 | 兴趣区间抽取装置、兴趣区间抽取方法 | CN201180012516.2 | 2011-10-28 | CN102782750A | 2012-11-14 | 小沼知浩; 川西亮一; 上野山努 |
| 一种兴趣区间抽取装置（104），根据动态图像文件中所包含的音频信号来抽取包括指定时刻（T0）的用户的兴趣区间，该兴趣区间抽取装置（104）具有：接口装置（109），取得指定时刻（T0）；似然度向量生成部（202），对于音频信号的每个第1单位区间计算表现多种音素各自的特征的各个锚模型（Ar）的似然度，并生成以计算出的各个似然度为分量的似然度向量（F）；以及兴趣区间抽取部（209），根据似然度向量（F）计算成为兴趣区间的候选的第1特征区间，并抽取包括指定时刻（T0）的第1特征区间的一部分区间作为兴趣区间。 | ||||||
| 4 | 音乐片段检测设备和方法以及音乐信号检测设备 | CN201210107008.9 | 2012-04-12 | CN102750947A | 2012-10-24 | 东山惠祐; 安部素嗣 |
| 本公开涉及音乐片段检测设备和方法以及音乐信号检测设备。基于被变换到时间频率域中的输入信号的每个区域的信号分量的强度(例如功率频谱)和通过逼近所述信号分量的强度而获得的函数(二次函数)，指标计算单元计算所述信号分量的音调指标。音乐确定单元基于音调指标确定输入信号的每个区域是否包括音乐。本技术可以应用于音乐片段检测设备，该音乐片段检测设备从其中音乐与噪声相混合的输入信号中检测音乐部分。 | ||||||
| 5 | 音频播放器支持的音频文件格式的测试设备及方法 | CN200810304752.1 | 2008-10-06 | CN101714380B | 2011-09-28 | 蒲小满; 邱金泉 |
| 本发明提供一种音频播放器支持的音频文件格式的测试方法及测试设备。该测试方法包括步骤：在该测试设备上输出每一音频文件至该音频播放器，每一音频文件具有一初始标识符对应每一音频文件所属的音频格式；利用该测试设备接收该音频播放器播放每一音频文件时输出的声音信号；在该测试设备上将接收的该音频播放器播放每一音频文件时输出的声音信号录制为每一声音文件；处理每一录制的声音文件以获取每一录制的声音文件的标识符；比较每一录制的声音文件的标识符与每一音频文件的初始标识符，根据比较结果确定该音频播放器支持的音频格式；以及输出该音频播放器支持的音频格式。本发明提供的测试设备及方法能够降低人力成本，提高测试效率。 | ||||||
| 6 | Device for vehicle | JP2011069888 | 2011-03-28 | JP2012203799A | 2012-10-22 | KAMIYA SHINJI |
| PROBLEM TO BE SOLVED: To provide a device for vehicle for receiving vehicle information even when an information center is able to receive only analog data.SOLUTION: The device for vehicle is configured to, when an emergency situation occurs in a vehicle, and it is connected to an information center 31 other than an emergency information center, transmit the vehicle information to the information center 31 by the voice prompt of the analog data. Thus, even when the information center 31 is the police headquarter, the fire department headquarter, or an individual telephone set (cellular phone) which is able to receive only the analog data, the operator of the information center 31 is able to appropriately cope with the emergency situation by confirming the circumstances of the vehicle side. | ||||||
| 7 | 興味区間抽出装置、興味区間抽出方法 | JP2012551746 | 2011-10-28 | JP5658285B2 | 2015-01-21 | 小沼 知浩; 知浩 小沼; 亮一 川西; 上野山 努; 上野山 努 |
| 8 | Music section detecting apparatus and method, program, recording medium, and music signal detecting apparatus | EP12160281.7 | 2012-03-20 | EP2544175A1 | 2013-01-09 | Touyama, Keisuke; Abe, Mototsugu |
An index calculating unit calculates a tonality index of a signal component of each area of an input signal transformed into a time frequency domain based on intensity (for example, power spectrum) of the signal component and a function (quadratic function) obtained by approximating the intensity of the signal component. A music determining unit determines whether or not each area of the input signal includes music based on the tonality index. The present technology can be applied to a music section detecting apparatus that detects a music part from an input signal in which music is mixed with noise.
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| 9 | Sample rate converter with automatic anti-aliasing filter | US14963228 | 2015-12-08 | US10002618B2 | 2018-06-19 | Thomas Craig Savell |
| The subject disclosure is directed towards dynamically computing anti-aliasing filter coefficients for sample rate conversion in digital audio. In one aspect, for each input-to-output sampling rate ratio (pitch) obtained, anti-aliasing filter coefficients are interpolated based upon the pitch (e.g., using the fractional part of the ratio) from two filters (coefficient sets) selected based upon the pitch (e.g., using the integer part of the ratio). The interpolation provides for fine-grained cutoff frequencies, and by re-computation for each pitch, smooth anti-aliasing with dynamically changing ratios. | ||||||
| 10 | Methods and systems for managing adaptation data | US13662125 | 2012-10-26 | US09899040B2 | 2018-02-20 | Royce A. Levien; Richard T. Lord; Robert W. Lord; Mark A. Malamud |
| Computationally implemented methods and systems include managing adaptation data, wherein the adaptation data is correlated to at least one aspect of speech of a particular party, facilitating transmission of the adaptation data to a target device, in response to an indicator related to a speech-facilitated transaction of a particular party, wherein the adaptation data is correlated to at least one aspect of speech of the particular party, and determining whether to update the adaptation data, said determination at least partly based on a result of at least a portion of the speech-facilitated transaction In addition to the foregoing, other aspects are described in the claims, drawings, and text. | ||||||
| 11 | Music section detecting apparatus and method, program, recording medium, and music signal detecting apparatus | US13443047 | 2012-04-10 | US08901407B2 | 2014-12-02 | Keisuke Touyama; Mototsugu Abe |
| An index calculating unit calculates a tonality index of a signal component of each area of an input signal transformed into a time frequency domain based on intensity (for example, power spectrum) of the signal component and a function (quadratic function) obtained by approximating the intensity of the signal component. A music determining unit determines whether or not each area of the input signal includes music based on the tonality index. The present technology can be applied to a music section detecting apparatus that detects a music part from an input signal in which music is mixed with noise. | ||||||
| 12 | 暗号化データを通信装置に伝達する方法及びシステム | JP2017561928 | 2016-06-01 | JP2018524860A | 2018-08-30 | フルニエ,ジャンクロード; ブノア,ベルナール; ヴェントリング,ベルトラン; クデルスキー,アンドレ |
| 実施形態は、通信装置に取付けるアクセサリという形をとる。アクセサリは、音を検出するためにアクセサリに配置されるマイクロフォン、検出音に基づいて暗号化音声データを生成するためにマイクロフォンと通信するアクセサリに配置される暗号モジュール、暗号化音声データを通信装置に伝達するように構成される暗号モジュールと通信するアクセサリに配置される通信インタフェース、及び通信装置音声センサに隣接して配置されるように構成される音声センサ抑止装置を備える。 【選択図】図1 |
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| 13 | 興味区間抽出装置、興味区間抽出方法 | JP2012551746 | 2011-10-28 | JPWO2012093430A1 | 2014-06-09 | 小沼 知浩; 知浩 小沼; 亮一 川西; 上野山 努; 上野山 努 |
| 興味区間抽出装置104は、動画ファイルに含まれるオーディオ信号に基づいて、指定時刻T0を含むユーザの興味区間を抽出する興味区間抽出装置104であって、指定時刻T0を取得するインターフェース装置109と、オーディオ信号の第1単位区間毎に複数種類のサウンド素片それぞれの特徴を表現するアンカーモデルArそれぞれの尤度を算出し、算出された尤度それぞれを成分とする尤度ベクトルFを生成する尤度ベクトル生成部202と、尤度ベクトルFに基づいて興味区間の候補となる第1特徴区間を算出し、指定時刻T0を含む第1特徴区間の一部を興味区間として抽出する興味区間抽出部209とを備える。 | ||||||
| 14 | Music-piece section detection device and method, program, recording medium, and music-piece signal detection device | JP2011093441 | 2011-04-19 | JP2012226106A | 2012-11-15 | HIGASHIYAMA KEISUKE; ABE MOTOTSUGU |
| PROBLEM TO BE SOLVED: To correctly detect a music-piece part from an input signal.SOLUTION: An indicator calculation part calculates an indicator of tone-likelihood of a signal component on the basis of strength of a signal component (e.g., a power spectrum) of an input signal in each of zones produced by converting the input signal into time-frequency zones and on a function (a quadratic function) approximating the strength of the signal component; a music-piece determination part determines whether a music-piece is included in each zone of the input signal on the basis of the indicator of the tone-likelihood. The present technique is applicable to a music-piece section detection device for detecting a music-piece part from an input signal containing a music-piece and noise intermixed. | ||||||
| 15 | Digital Voice Processing Method And System For Headset Computer | US16107390 | 2018-08-21 | US20180359550A1 | 2018-12-13 | Dashen Fan; Jang Ho Kim; Yong Seok Seo; John C. C. Fan |
| The invention is a multi-microphone voice processing SoC primarily for head worn applications. It bypasses the use of conventional pre-amp voice CODEC (ADC/DAC) chips all together by replacing their functionality with digital MEMS microphone(s) and digital speaker driver (DSD). Functionality necessary for speech recognition such as noise/echo cancellation, speech compression, speech feature extraction and lossless speech transmission are also integrated into the SoC. One embodiment is a noise cancellation chip for wired, battery powered headsets and earphones, as smart-phone accessory. Another embodiment is as a wireless Bluetooth noise cancellation companion chip. The invention can be used in headwear, eyewear glass, mobile wearable computing, heavy duty military, aviation and industrial headsets and other speech recognition applications in noisy environments. | ||||||
| 16 | Digital voice processing method and system for headset computer | US14318235 | 2014-06-27 | US10070211B2 | 2018-09-04 | Dashen Fan; Jang Ho Kim; Yong Seok Seo; John C. C. Fan |
| The invention is a multi-microphone voice processing SoC primarily for head worn applications. It bypasses the use of conventional pre-amp voice CODEC (ADC/DAC) chips all together by replacing their functionality with digital MEMS microphone(s) and digital speaker driver (DSD). Functionality necessary for speech recognition such as noise/echo cancellation, speech compression, speech feature extraction and lossless speech transmission are also integrated into the SoC. One embodiment is a noise cancellation chip for wired, battery powered headsets and earphones, as smart-phone accessory. Another embodiment is as a wireless Bluetooth noise cancellation companion chip. The invention can be used in headwear, eyewear glass, mobile wearable computing, heavy duty military, aviation and industrial headsets and other speech recognition applications in noisy environments. | ||||||
| 17 | MIX BUFFERS AND COMMAND QUEUES FOR AUDIO BLOCKS | US15469480 | 2017-03-24 | US20170200466A1 | 2017-07-13 | John A. Tardif; Brian Lloyd Schmidt; Sunil Kumar Vemula; Robert N. Heitkamp |
| The subject disclosure is directed towards a technology that may be used in an audio processing environment. Nodes of an audio flow graph are associated with virtual mix buffers. As the flow graph is processed, commands and virtual mix buffer data are provided to audio fixed-function processing blocks. Each virtual mix buffer is mapped to a physical mix buffer, and the associated command is executed with respect to the physical mix buffer. One physical mix buffer mix buffer may be used as an input data buffer for the audio fixed-function processing block, and another physical mix buffer as an output data buffer, for example. | ||||||
| 18 | System and method for audio scene understanding of physical object sound sources | US14877680 | 2015-10-07 | US09668073B2 | 2017-05-30 | Samarjit Das; Joao P. Sousa |
| A method of operating an audio monitoring system includes generating with a sound sensor audio data corresponding to a sound event generated by an object in a scene around the sound sensor, identifying with a processor a type and action of the object in the scene that generated the sound with reference to the audio data, generating with the processor a timestamp corresponding to a time of the detection of the sound event, and updating a scene state model corresponding to sound events generated by a plurality of objects in the scene with reference to the identified type of object, action taken by the object, and the timestamp. The method further includes identifying a sound event in the scene with reference to the scene state model and a predetermined scene grammar stored in a memory, and generating with the processor an output corresponding to the sound event. | ||||||
| 19 | Mix buffers and command queues for audio blocks | US13766128 | 2013-02-13 | US09646623B2 | 2017-05-09 | John A. Tardif; Brian Lloyd Schmidt; Sunil Kumar Vemula; Robert N. Heitkamp |
| The subject disclosure is directed towards a technology that may be used in an audio processing environment. Nodes of an audio flow graph are associated with virtual mix buffers. As the flow graph is processed, commands and virtual mix buffer data are provided to audio fixed-function processing blocks. Each virtual mix buffer is mapped to a physical mix buffer, and the associated command is executed with respect to the physical mix buffer. One physical mix buffer mix buffer may be used as an input data buffer for the audio fixed-function processing block, and another physical mix buffer as an output data buffer, for example. | ||||||
| 20 | SAMPLE RATE CONVERTER WITH AUTOMATIC ANTI-ALIASING FILTER | US14963228 | 2015-12-08 | US20160217802A1 | 2016-07-28 | Thomas Craig Savell |
| The subject disclosure is directed towards dynamically computing anti-aliasing filter coefficients for sample rate conversion in digital audio. In one aspect, for each input-to-output sampling rate ratio (pitch) obtained, anti-aliasing filter coefficients are interpolated based upon the pitch (e.g., using the fractional part of the ratio) from two filters (coefficient sets) selected based upon the pitch (e.g., using the integer part of the ratio). The interpolation provides for fine-grained cutoff frequencies, and by re-computation for each pitch, smooth anti-aliasing with dynamically changing ratios. | ||||||
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