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公开(公告)号：US20240177724A1
公开(公告)日：2024-05-30
申请号：US18406351
申请日：2024-01-08
发明人： Goran MARKOVIC
IPC分类号： G10L19/083 , G10L19/032 , G10L21/10 , G10L25/18
CPC分类号： G10L19/083 , G10L19/032 , G10L21/10 , G10L25/18
摘要： An audio encoder for encoding an audio signal comprising an pulse portion and a stationary portion, comprises: a pulse extractor configured for extracting the pulse portion from the audio signal, further comprising a pulse coder for encoding the extracted pulse portion to acquire an encoded pulse portion; wherein the pulse extractor is configured to determine a spectrogram of the audio signal to extract the pulse portion, wherein the spectrogram has higher time resolution than the signal encoder; a signal encoder configured for encoding a residual signal derived from the audio signal to acquire an encoded residual signal, the residual signal being derived from the audio signal so that the pulse portion is reduced or eliminated from the audio signal; and an output interface configured for outputting the encoded pulse portion and the encoded residual signal to provide an encoded signal.
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2.发明授权公开(公告)号：US11776551B2
公开(公告)日：2023-10-03
申请号：US17120526
申请日：2020-12-14
IPC分类号： G10L19/005 , G10L19/06 , G10L19/002 , G10L19/012 , G10L19/083 , G10L19/09 , G10L19/12 , G10L19/07 , G10L19/22 , G10L19/02 , G10L19/00
CPC分类号： G10L19/005 , G10L19/002 , G10L19/012 , G10L19/06 , G10L19/07 , G10L19/083 , G10L19/09 , G10L19/12 , G10L19/22 , G10L19/0212 , G10L2019/0002 , G10L2019/0011 , G10L2019/0016
摘要： An apparatus for decoding an audio signal is provided, having a receiving interface, configured to receive a first frame having a first audio signal portion of the audio signal, and configured to receive a second frame having a second audio signal portion of the audio signal; a noise level tracing unit, wherein the noise level tracing unit is configured to determine noise level information depending on at least one of the first audio signal portion and the second audio signal portion; a first reconstruction unit for reconstructing, in a first reconstruction domain, a third audio signal portion of the audio signal depending on the noise level information; a transform unit for transforming the noise level information to a second reconstruction domain; and a second reconstruction unit for reconstructing, in the second reconstruction domain, a fourth audio signal portion of the audio signal depending on the noise level information.
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公开(公告)号：US11749290B2
公开(公告)日：2023-09-05
申请号：US17373148
申请日：2021-07-12
发明人： Yang Gao
IPC分类号： G10L19/005 , G10L19/09 , G10L25/90 , G10L19/083
CPC分类号： G10L19/09 , G10L19/005 , G10L19/083 , G10L25/90
摘要： Methods, systems, and apparatus, including computer programs encoded on computer storage media, for performing long-term prediction (LTP) are described. One example of the methods includes determining a pitch gain and a pitch lag of an input audio signal for at least a predetermined number of frames. It is determined that the pitch gain of the input audio signal has exceeded a predetermined threshold and that a change of the pitch lag of the input audio signal has been within a predetermined range for at least the predetermined number of frames. In response to determining that the pitch gain of the input audio signal has exceeded the predetermined threshold and that the change of the third pitch lag has been within the predetermined range for at least the predetermined number of frames, a pitch gain is set for a current frame of the input audio signal.
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公开(公告)号：US11423916B2
公开(公告)日：2022-08-23
申请号：US16900950
申请日：2020-06-14
IPC分类号： G10L19/02 , G10L25/18 , G10L25/48 , H03H17/02 , G10L19/00 , G10L19/12 , G10L19/26 , G10L25/21 , G10L19/025 , G10L19/083 , G10L25/12
摘要： The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
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公开(公告)号：US11270688B2
公开(公告)日：2022-03-08
申请号：US16930337
申请日：2020-07-16
申请人： EVOCO LABS CO., LTD.
发明人： Congxi Lu , Linkai Li , Hongcheng Sun , Xinke Liu
IPC分类号： G10L15/06 , G06F17/14 , G06N3/04 , G06N3/08 , G10L15/16 , G10L19/005 , G10L19/083 , G10L25/30 , H04R25/00 , G10L21/0208 , G10L21/0364
摘要： A deep neural network based audio processing method is provided. The method includes: obtaining a deep neural network based speech extraction model; receiving an audio input object having a speech portion and a non-speech portion, wherein the audio input object includes one or more audio data frames each having a set of audio data samples sampled at a predetermined sampling interval and represented in time domain data format; obtaining a user audiogram and a set of user gain compensation coefficients associated with the user audiogram; and inputting the audio input object and the set of user gain compensation coefficients into the trained speech extraction model to obtain an audio output result represented in time domain data format outputted by the trained speech extraction model, wherein the non-speech portion of the audio input object is at least partially attenuated in or removed from the audio output result.
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公开(公告)号：US11056125B2
公开(公告)日：2021-07-06
申请号：US16565920
申请日：2019-09-10
发明人： Erik Norvell , Volodya Grancharov
IPC分类号： G10L19/02 , G10L19/032 , G10L19/038 , G10L21/0232 , G10L19/083
摘要： A gain adjustment apparatus for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter configured to estimate an accuracy measure of the shape representation, and to determine a gain correction based on the estimated accuracy measure. An envelope adjuster further included in the apparatus is configured to adjust the gain representation based on the determined gain correction.
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公开(公告)号：US20210074266A1
公开(公告)日：2021-03-11
申请号：US16930337
申请日：2020-07-16
申请人： EVOCO LABS CO., LTD.
发明人： Congxi LU , Linkai LI , Hongcheng SUN , Xinke LIU
IPC分类号： G10L15/06 , G06N3/04 , G10L15/16 , G10L19/005 , G06F17/14 , G10L19/083 , G06N3/08
摘要： A deep neural network based audio processing method is provided. The method includes: obtaining a deep neural network based speech extraction model; receiving an audio input object having a speech portion and a non-speech portion, wherein the audio input object includes one or more audio data frames each having a set of audio data samples sampled at a predetermined sampling interval and represented in time domain data format; obtaining a user audiogram and a set of user gain compensation coefficients associated with the user audiogram; and inputting the audio input object and the set of user gain compensation coefficients into the trained speech extraction model to obtain an audio output result represented in time domain data format outputted by the trained speech extraction model, wherein the non-speech portion of the audio input object is at least partially attenuated in or removed from the audio output result.
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公开(公告)号：US20200175228A1
公开(公告)日：2020-06-04
申请号：US16784551
申请日：2020-02-07
IPC分类号： G06F40/284 , G06N5/04 , G10L19/083 , G06K9/62 , G06F16/35
摘要： A method performed by a device may include identifying a plurality of samples of textual content; performing tokenization of the plurality of samples to generate a respective plurality of tokenized samples; performing embedding of the plurality of tokenized samples to generate a sample matrix; determining groupings of attributes of the sample matrix using a convolutional neural network; determining context relationships between the groupings of attributes using a bidirectional long short term memory (LSTM) technique; selecting predicted labels for the plurality of samples using a model, wherein the model selects, for a particular sample of the plurality of samples, a predicted label of the predicted labels from a plurality of labels based on respective scores of the particular sample with regard to the plurality of labels and based on a nonparametric paired comparison of the respective scores; and providing information identifying the predicted labels.
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9.发明申请公开(公告)号：US20190362729A1
公开(公告)日：2019-11-28
申请号：US16534685
申请日：2019-08-07
IPC分类号： G10L19/02 , H03H17/02 , G10L25/18 , G10L25/21 , G10L19/083 , G10L19/025 , G10L25/48 , G10L25/12 , G10L19/26 , G10L19/12 , G10L19/00
摘要： The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
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公开(公告)号：US10311885B2
公开(公告)日：2019-06-04
申请号：US15817296
申请日：2017-11-20
IPC分类号： G10L19/02 , G10L19/24 , G10L25/93 , G10L19/005 , G10L19/083 , G10L21/038 , G10L21/0388
摘要： A method for recovering a lost frame in a received audio signal includes: obtaining an initial high-frequency band signal of a current lost frame in the received audio signal; calculating a ratio R, wherein the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame; obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame; and recovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame. The method can be used in an audio signal decoding process for low-loss recovery of lost frames of the audio signal.
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