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    レコーダ、情報処理装置、情報処理システム、および、情報処理方法
    3.
    发明申请
    レコーダ、情報処理装置、情報処理システム、および、情報処理方法 审中-公开

    公开(公告)号：WO2021161834A1

    公开(公告)日：2021-08-19

    申请号：PCT/JP2021/003498

    申请日：2021-02-01

    Applicant: 株式会社時空テクノロジーズ

    Inventor： 橋本　善久 ,  春日　秀之 ,  林　優一

    IPC: H04R3/00 ,  G10L21/0316 ,  G10L15/30

    Abstract: 本実施形態に係るレコーダは、複数のコネクタと、アナログ・デジタルコンバータと、コントローラと、記憶装置とを備える。コントローラは、複数のマイクロフォンと接続可能である。アナログ・デジタルコンバータは、複数のコネクタのそれぞれから受信した複数のアナログ信号をデジタル信号へ変換する。コントローラは、デジタル信号に基づいて音データを生成する。記憶装置は、コントローラによって生成された音データを記憶する。コントローラは、アナログ・デジタルコンバータから、複数のアナログ信号のレベルを含む解析情報を受信し、解析情報に基づいて複数のアナログ信号のレベルを調整するための制御コマンドを、アナログ・デジタルコンバータへ送信する。

    AUDIO METADATA SMOOTHING
    4.
    发明申请
    AUDIO METADATA SMOOTHING 审中-公开

    公开(公告)号：WO2021061656A1

    公开(公告)日：2021-04-01

    申请号：PCT/US2020/052017

    申请日：2020-09-22

    Applicant: NETFLIX, INC.

    Inventor： ZHENG, Weiguo ,  CHING, Rex ,  NI, Weibo ,  MIYAGI, Kensuke ,  MUNDAY, Sean ,  TAO, Teresa

    IPC: H04N21/236 ,  H04N21/84 ,  H04N21/242 ,  H04N21/2343 ,  H04N21/44 ,  H04N21/845 ,  G11B27/02 ,  G10L19/16 ,  G10L21/0316 ,  G10L21/0356 ,  G10L21/055

    Abstract: The disclosed computer-implemented method for smoothing audio gaps using adaptive metadata identifies an initial audio segment and a subsequent audio segment that follows the initial audio segment. The method accesses a first set of metadata that corresponds to a last audio frame of the initial audio segment and accesses a second set of metadata that corresponds to the first audio frame of the subsequent audio segment. The first and second sets of metadata include audio characteristic information for the two audio segments. The method then generates a new set of metadata that is based on both sets of audio characteristics. The method further inserts a new audio frame between the last audio frame of the initial audio segment and the first audio frame of the subsequent audio segment and applies the new set of metadata to the new audio frame. Various other methods, systems, and computer-readable media are also disclosed.

    DIALOG ENHANCEMENT USING ADAPTIVE SMOOTHING
    5.
    发明申请
    DIALOG ENHANCEMENT USING ADAPTIVE SMOOTHING 审中-公开

    公开(公告)号：WO2021041568A1

    公开(公告)日：2021-03-04

    申请号：PCT/US2020/048034

    申请日：2020-08-26

    Applicant: DOLBY LABORATORIES LICENSING CORPORATION

    Inventor： YU, Xuemei

    IPC: G10L21/0316 ,  H03G3/30 ,  H03G7/00 ,  G10L25/81

    Abstract: A method of enhancing dialog intelligibility in an audio signal, comprising determining a speech confidence score that the audio content includes speech content, determining a music confidence score that the audio content includes music correlated content, in response to the speech confidence score, and applying a user selected gain of selected frequency bands of the audio signal to obtain a dialogue enhanced audio signal. The user selected gain is smoothed by an adaptive smoothing algorithm, an impact of past frames in said smoothing algorithm being determined by a smoothing factor, the smoothing factor being calculated in response to the music confidence score, and having a relatively higher value for content having a relatively higher music confidence score and a relatively lower value for speech content having a relatively lower music confidence score, so as to increase the impact of past frames on the dialogue enhancement of music correlated content.

    오디오 신호의 주파수의 변화에 따른 위상 변화율에 기반하여 노이즈가 감쇠된 오디오 신호를 생성하는 장치 및 방법
    6.
    发明申请
    오디오 신호의 주파수의 변화에 따른 위상 변화율에 기반하여 노이즈가 감쇠된 오디오 신호를 생성하는 장치 및 방법 审中-公开

    公开(公告)号：WO2019156339A1

    公开(公告)日：2019-08-15

    申请号：PCT/KR2018/016121

    申请日：2018-12-18

    Applicant: 삼성전자 주식회사

    Inventor： 백순호 ,  문한길 ,  손백권 ,  양재모 ,  조기호 ,  김준태

    IPC: G10L21/0208 ,  G10L25/30 ,  G10L21/0316 ,  G10L21/0216

    Abstract: 전자 장치가 개시된다. 이 외에도 명세서를 통해 파악되는 다양한 실시 예가 가능하다. 전자 장치는, 음성 신호와 잡음 신호를 포함하는 복수의 입력 신호들을 수신하는 복수의 입력 장치들, 및 상기 입력 장치들과 전기적으로 연결되는 프로세서를 포함하고, 상기 프로세서는, 상기 복수의 입력 신호들에 대한 신호 대 잡음 비(signal to ratio, SNR) 값을 주파수 대역 별로 결정하고, 상기 SNR 값이 지정된 임계 값 이상인 제1 주파수 대역에서 상기 복수의 입력 신호들의 주파수 대비 위상의 변화를 나타내는 제1 파라미터를 결정하고, 상기 제1 파라미터에 기반하여, 상기 SNR 값이 상기 임계 값 미만인 제2 주파수 대역에서 상기 복수의 입력 신호들의 주파수 대비 위상의 변화를 나타내는 제2 파라미터를 결정하고, 상기 제1 파라미터 및 상기 제2 파라미터에 기반하여 상기 복수의 입력 신호들에 대한 빔포밍을 수행하도록 설정될 수 있다.

    電気式人工喉頭装置
    7.
    发明申请
    電気式人工喉頭装置 审中-公开
    电动人造LARYNX装置

    公开(公告)号：WO2015019835A1

    公开(公告)日：2015-02-12

    申请号：PCT/JP2014/069274

    申请日：2014-07-22

    Applicant: 国立大学法人奈良先端科学技術大学院大学

    Inventor： 戸田　智基 ,  田中　宏 ,  中村　哲 ,  サクティ　サクリアニ　 ,  ニュービッグ　グラム　

    IPC: G10L15/00 ,  A61F2/20 ,  G10L21/013 ,  G10L21/0316 ,  G10L25/90 ,  H04R1/14

    CPC classification number: A61F2/20 ,  A61F2002/206 ,  G10L21/013 ,  G10L21/0316 ,  G10L25/90 ,  H04R1/14

    Abstract: 　使用者が発する発声音に適合した音源音を円滑に出力することが可能な電気式人工喉頭装置を提供する。電気式人工喉頭装置１は、使用者Ｐの声道に入力された音源音が調音処理されて発せられる発声音を集音して、発声信号を生成する集音部１０と、集音部１０が生成する発声信号に対応した音源信号を生成する信号処理部２０と、信号処理部２０が生成する音源信号を再生して音源音を出力する音源信号再生部３０と、を備える。

    Abstract translation: 提供一种能够平滑地输出与用户发出的语音对应的声源声音的电动人造喉装置。 一种电动人造喉装置（1）具有：声音收集单元（10），其收集通过输入到用户（P）的声道中的声源声音处理并发出的语音发音，并产生语音 信号; 信号处理单元（20），其生成与由所述声音采集单元（10）生成的语音信号相对应的声源信号; 以及再现由信号处理单元（20）生成的声源信号并输出​​声源声音的声源信号再现单元（30）。

    AUDIO CLIPPING DETECTION
    8.
    发明申请
    AUDIO CLIPPING DETECTION 审中-公开
    音频剪辑检测

    公开(公告)号：WO2014126842A1

    公开(公告)日：2014-08-21

    申请号：PCT/US2014/015533

    申请日：2014-02-10

    Applicant: GOOGLE INC.

    Inventor： SKOGLUND, Jan ,  LINDEN, Jan Thomas

    IPC: G10L21/02 ,  G10L21/0316

    CPC classification number: H04R29/00 ,  G10L25/69

    Abstract: Methods and systems for detecting the presence and frequency of clipping in an audio signal are provided. A clipping detection algorithm detects the presence of hard and soft clipping using histograms with intervals of samples, rather than attempting to identify the clipping value. Therefore, it is not essential to the algorithm that there be a large number of bins. Furthermore, the bins may be non-uniformly distributed since the number of samples belonging to lower amplitudes is of little importance. The detection algorithm is also configured to determine the severity and/or perceptual effect of any clipping found to be present in the signal by calculating the ratio of clipped samples to non-clipped samples. Temporal information on the occurrence of clipping in the signal is also used to evaluate perceptual effect.

    Abstract translation: 提供了用于检测音频信号中的限幅的存在和频率的方法和系统。 剪辑检测算法使用具有样本间隔的直方图来检测是否存在硬和软剪辑，而不是尝试识别剪切值。 因此，算法不是必须存在大量的分组。 此外，由于属于较低振幅的样本数量几乎不重要，因此可能不均匀分布。 检测算法还被配置为通过计算剪切样本与非剪切样本的比率来确定发现存在于信号中的任何限幅的严重性和/或感知效果。 关于信号中剪辑发生的时间信息也用于评估感知效果。

    METHOD AND SYSTEM FOR BIAS CORRECTED SPEECH LEVEL DETERMINATION
    9.
    发明申请
    METHOD AND SYSTEM FOR BIAS CORRECTED SPEECH LEVEL DETERMINATION 审中-公开
    用于偏差校正语音级别确定的方法和系统

    公开(公告)号：WO2013142695A1

    公开(公告)日：2013-09-26

    申请号：PCT/US2013/033312

    申请日：2013-03-21

    Applicant: DOLBY LABORATORIES LICENSING CORPORATION

    Inventor： GUNAWAN, David ,  DICKINS, Glenn

    IPC: G10L21/0316 ,  G10L25/48 ,  G10L25/78 ,  G10L25/21 ,  G10L25/18

    CPC classification number: G10L21/0316 ,  G10L25/18 ,  G10L25/21 ,  G10L25/48 ,  G10L25/78

    Abstract: Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.

    Abstract translation: 一种用音频信号测量的语音水平的方法，该方法通过增加噪声对其进行修正和/或降低其变化的影响，并且降低其变化的影响，以及被配置为执行该方法的任何实施例的系统 。 在一些实施例中，该方法包括以下步骤：产生表示输入语音信号的频带，频域数据，根据数据确定语音信号的高斯参数频谱模型，以及从参数频谱模型确定估计的平均语音电平 和数据的每个频带的标准偏差值; 以及生成指示针对每个频带的偏置校正的平均语音电平的语音电平数据，包括使用至少一个校正值来校正所述频带的估计平均语音电平，其中每个校正值已经使用参考语音模型预先确定。

    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT
    10.
    发明申请
    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT 审中-公开
    后处理增益信号增强

    公开(公告)号：WO2013142661A1

    公开(公告)日：2013-09-26

    申请号：PCT/US2013/033251

    申请日：2013-03-21

    Applicant: DOLBY LABORATORIES LICENSING CORPORATION

    Inventor： SUN, Xuejing ,  DICKINS, Glenn N.

    IPC: G10L21/0316

    CPC classification number: G10L21/0364 ,  G10K11/16 ,  G10L21/0224 ,  G10L21/0316 ,  G10L21/034 ,  G10L25/78 ,  H03G3/301 ,  H03G3/32

    Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

    Abstract translation: 一种用于后处理由输入处理确定的原始增益以产生后处理增益的方法，装置和逻辑，包括使用增量增益平滑和决策导向增益平滑中的一者或两者。 Δ增益平滑包括以平滑因子对原始增益应用平滑滤波器，该平滑因子取决于增益增量：当前帧的原始增益与前一帧的后处理增益之间的差的绝对值。 决策导向增益平滑包括将原始增益转换为信噪比，将具有平滑因子的平滑滤波器应用于信噪比以计算平滑的信噪比，并将 平滑的信噪比以确定第二平滑增益，平滑因子可能取决于增益增量。

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