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公开(公告)号：US10854207B2
公开(公告)日：2020-12-01
申请号：US16231913
申请日：2018-12-24
Inventor： Wei Li , Binghua Qian , Xingming Jin , Ke Li , Fuzhang Wu , Yongjian Wu , Feiyue Huang
Abstract: A method and an apparatus for training a voiceprint recognition system are provided. The method includes obtaining a voice training data set comprising voice segments of users; determining identity vectors of all the voice segments; identifying identity vectors of voice segments of a same user in the determined identity vectors; placing the recognized identity vectors of the same user in the users into one of user categories; and determining an identity vector in the user category as a first identity vector. The method further includes normalizing the first identity vector by using a normalization matrix, a first value being a sum of similarity degrees between the first identity vector in the corresponding category and other identity vectors in the corresponding category; training the normalization matrix, and outputting a training value of the normalization matrix when the normalization matrix maximizes a sum of first values of all the user categories.
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公开(公告)号：US10692503B2
公开(公告)日：2020-06-23
申请号：US15764573
申请日：2017-03-03
Inventor： Xingming Jin , Wei Li , Fangmai Zheng , Fuzhang Wu , Bilei Zhu , Binghua Qian , Ke Li , Yongjian Wu , Feiyue Huang
Abstract: A voice data processing method and apparatus are provided. The method includes obtaining an I-Vector vector of each of voice samples, and determining a target seed sample in the voice samples. A first cosine distance is calculated between an I-Vector vector of the target seed sample and an I-Vector vector of a target remaining voice sample, where the target remaining voice sample is a voice sample other than the target seed sample in the voice samples. A target voice sample is filtered from the voice samples or the target remaining voice sample according to the first cosine distance, to obtain a target voice sample whose first cosine distance is greater than a first threshold.
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公开(公告)号：US10770050B2
公开(公告)日：2020-09-08
申请号：US15775460
申请日：2017-06-02
Inventor： Bi Lei Zhu , Ke Li , Yong Jian Wu , Fei Yue Huang
IPC: G10H1/36 , G10L21/0272
Abstract: An audio data processing method and apparatus are provided. The method includes obtaining audio data. An overall spectrum of the audio data is obtained and separated into a singing voice spectrum and an accompaniment spectrum. An accompaniment binary mask of the audio data is calculated according to the audio data. The singing voice spectrum and the accompaniment spectrum are processed using the accompaniment binary mask, to obtain accompaniment data and singing voice data.
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公开(公告)号：US11288309B2
公开(公告)日：2022-03-29
申请号：US15951219
申请日：2018-04-12
Inventor： Bilei Zhu , Fangmai Zheng , Xingming Jin , Ke Li , Yongjian Wu , Feiyue Huang
IPC: G06F16/632 , G06F16/61 , G06F16/683 , G06F16/00 , G06F3/16 , G10L25/90
Abstract: A melody information processing method is described. A piece of Musical Instrument Digital Interface (MIDI) data corresponding to a song is received, a song identifier of the song is obtained, first melody information is generated according to the MIDI data, and the first melody information is stored in association with the song identifier in a melody database. Moreover, a user unaccompanied-singing audio data set that is uploaded from a user terminal is received, second melody information corresponding to the song identifier is extracted according to the user unaccompanied-singing audio data set, and the second melody information is stored in association with the song identifier in the melody database.
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公开(公告)号：US10699699B2
公开(公告)日：2020-06-30
申请号：US15993332
申请日：2018-05-30
Inventor： Fuzhang Wu , Binghua Qian , Wei Li , Ke Li , Yongjian Wu , Feiyue Huang
Abstract: The embodiments of the present disclosure disclose a method for constructing a speech decoding network in digital speech recognition. The method comprises acquiring training data obtained by digital speech recording, the training data comprising a plurality of speech segments, and each speech segment comprising a plurality of digital speeches; performing acoustic feature extraction on the training data to obtain a feature sequence corresponding to each speech segment; performing progressive training starting from a mono-phoneme acoustic model to obtain an acoustic model; acquiring a language model, and constructing a speech decoding network by the language model and the acoustic model obtained by training.
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Patent Agency Ranking
公开(公告)号：US10650830B2
公开(公告)日：2020-05-12
申请号：US15954416
申请日：2018-04-16
Inventor： Wei Li , Binghua Qian , Xingming Jin , Ke Li , Fuzhang Wu , Yongjian Wu , Feiyue Huang
Abstract: Processing circuitry of an information processing apparatus obtains a set of identity vectors that are calculated according to voice samples from speakers. The identity vectors are classified into speaker classes respectively corresponding to the speakers. The processing circuitry selects, from the identity vectors, first subsets of interclass neighboring identity vectors respectively corresponding to the identity vectors and second subsets of intraclass neighboring identity vectors respectively corresponding to the identity vectors. The processing circuitry determines an interclass difference based on the first subsets of interclass neighboring identity vectors and the corresponding identity vectors; and determines an intraclass difference based on the second subsets of intraclass neighboring identify vectors and the corresponding identity vectors. Further, the processing circuitry determines a set of basis vectors to maximize a projection of the interclass difference on the basis vectors and to minimize a projection of the intraclass difference on the basis vectors.
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7.发明申请公开(公告)号：US20190221202A1
公开(公告)日：2019-07-18
申请号：US16365458
申请日：2019-03-26
Inventor： Wei Li , Hangyu Yan , Ke Li , Yongjian Wu , Feiyue Huang
IPC: G10L13/10 , G10L13/047 , G06F17/16 , G06N7/00
Abstract: A statistical parameter modeling method is performed by a server. After obtaining model training data, the model training data including a text feature sequence and a corresponding original speech sample sequence, the server inputs an original vector matrix formed by matching a text feature sample point in the text feature sample sequence with a speech sample point in the original speech sample sequence into a statistical parameter model for training and then performs non-linear mapping calculation on the original vector matrix in a hidden layer, to output a corresponding prediction speech sample point. The server then obtains a model parameter of the statistical parameter model according to the prediction speech sample point and a corresponding original speech sample point by using a smallest difference principle, to obtain a corresponding target statistical parameter model.
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8.发明申请公开(公告)号：US20190115031A1
公开(公告)日：2019-04-18
申请号：US16213421
申请日：2018-12-07
Inventor： Wei Li , Binghua Qian , Xingming Jin , Ke Li , Fuzhang Wu , Yongjian Wu , Feiyue Huang
Abstract: An identity vector generation method is provided. The method includes obtaining to-be-processed speech data. Corresponding acoustic features are extracted from the to-be-processed speech data. A posterior probability that each of the acoustic features belongs to each Gaussian distribution component in a speaker background model is calculated to obtain a statistic. The statistic is mapped to a statistic space to obtain a reference statistic, the statistic space built according to a statistic corresponding to a speech sample exceeding a threshold speech duration. A corrected statistic is determined according to the calculated statistic and the reference statistic; and an identity vector is generated according to the corrected statistic.
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公开(公告)号：US20180233151A1
公开(公告)日：2018-08-16
申请号：US15954416
申请日：2018-04-16
Inventor： Wei Li , Binghua Qian , Xingming Jin , Ke Li , Fuzhang Wu , Yongjian Wu , Feiyue Huang
Abstract: Processing circuitry of an information processing apparatus obtains a set of identity vectors that are calculated according to voice samples from speakers. The identity vectors are classified into speaker classes respectively corresponding to the speakers. The processing circuitry selects, from the identity vectors, first subsets of interclass neighboring identity vectors respectively corresponding to the identity vectors and second subsets of intraclass neighboring identity vectors respectively corresponding to the identity vectors. The processing circuitry determines an interclass difference based on the first subsets of interclass neighboring identity vectors and the corresponding identity vectors; and determines an intraclass difference based on the second subsets of intraclass neighboring identify vectors and the corresponding identity vectors. Further, the processing circuitry determines a set of basis vectors to maximize a projection of the interclass difference on the basis vectors and to minimize a projection of the intraclass difference on the basis vectors.
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10.发明授权公开(公告)号：US11289069B2
公开(公告)日：2022-03-29
申请号：US16365458
申请日：2019-03-26
Inventor： Wei Li , Hangyu Yan , Ke Li , Yongjian Wu , Feiyue Huang
IPC: G10L13/10 , G10L13/08 , G06N7/00 , G06F17/16 , G10L13/047
Abstract: A statistical parameter modeling method is performed by a server. After obtaining model training data, the model training data including a text feature sequence and a corresponding original speech sample sequence, the server inputs an original vector matrix formed by matching a text feature sample point in the text feature sample sequence with a speech sample point in the original speech sample sequence into a statistical parameter model for training and then performs non-linear mapping calculation on the original vector matrix in a hidden layer, to output a corresponding prediction speech sample point. The server then obtains a model parameter of the statistical parameter model according to the prediction speech sample point and a corresponding original speech sample point by using a smallest difference principle, to obtain a corresponding target statistical parameter model.
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