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公开(公告)号：US20200279572A1
公开(公告)日：2020-09-03
申请号：US16877389
申请日：2020-05-18
IPC分类号： G10L19/028 , G10L19/26 , G10L21/02 , G10L19/16 , G10L21/0316
摘要： Method and apparatus are provided for reconstructing a noise component of a speech/audio signal. A bitstream is received and decoded to obtain a speech/audio signal. A first speech/audio signal is determined according to the speech/audio signal. A symbol of each sample value in the first speech/audio signal and an amplitude value of each sample value in the first speech/audio signal is determined. An adaptive normalization length and an adjusted amplitude value of each sample value are determined according to the adaptive normalization length and the amplitude value of each sample value. A second speech/audio signal is determined according to the symbol of each sample value and the adjusted amplitude value of each sample value.
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公开(公告)号：US20200273475A1
公开(公告)日：2020-08-27
申请号：US16869000
申请日：2020-05-07
发明人： Emmanuel RAVELLI , Martin DIETZ , Michael SCHNABEL , Arthur TRITTHART , Alexander TSCHEKALINSKIJ
摘要： In apparatus, methods, and programs for selecting pitch lag, an encoder obtains a first and a second estimates of a pitch lag for a current frame. A selected value is chosen by selection between the first and the second estimates, based on a first and a second correlation measurements. The second estimate is conditioned by the pitch lag selected at the previous frame. The selection is based on a comparison between: a downscaled version of a first correlation measurement associated to the current frame and obtained at a lag corresponding to the first estimate; and a second correlation measurement associated to the current frame and obtained at a lag corresponding to the second estimate.
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公开(公告)号：US20200265845A1
公开(公告)日：2020-08-20
申请号：US16867730
申请日：2020-05-06
申请人： Sony Corporation
发明人： Yuki Yamamoto , Toru Chinen , Hiroyuki Honma , Runyu Shi
摘要： The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality.A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.
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公开(公告)号：US20200258532A1
公开(公告)日：2020-08-13
申请号：US16797841
申请日：2020-02-21
发明人： Lars VILLEMOES
IPC分类号： G10L19/02 , G06F30/327 , G06F30/30 , G10L19/26 , G10L19/06 , G10L19/032 , G10L19/005 , G10L19/093
摘要： The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively. The method comprises determining a model parameter (613) of a signal model; determining a prediction coefficient to be applied to a previous sample (614) of a first decoded subband signals derived from the first subband signal, based on the signal model, based on the model parameter (613) and based on the analysis filterbank (612); wherein a time slot of the previous sample (614) is prior to a time slot of the first sample (615); and determining an estimate of the first sample (615) by applying the prediction coefficient to the previous sample (614).
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公开(公告)号：US10741196B2
公开(公告)日：2020-08-11
申请号：US16087006
申请日：2016-03-24
IPC分类号： G10L21/038 , G10L25/69 , G10L19/26 , G10L21/02 , G10L19/24
摘要： A sampler module divides an audio signal into a series of sequential samples. A signal quality detector module identifies, over a plurality of samples at an outset of the audio signal, a spectral variance of a first range of frequencies of the audio signal below a predetermined threshold frequency as being consistently greater than a spectral variance of a second range of frequencies of the audio signal above the predetermined threshold frequency and determines a signal treatment indication responsive to the identification. A signal enhancer module sequentially receives and analyzes one or more sample components of the audio signal to identify lost parts of the audio signal in the one or more sample components of respective sequential samples, and generates, in accordance with the signal treatment indication, a corresponding signal treatment for each of the one or more sample components of respective sequential samples having a corresponding identified lost part.
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公开(公告)号：US10720170B2
公开(公告)日：2020-07-21
申请号：US15884190
申请日：2018-01-30
发明人： Florin Ghido , Sascha Disch , Jürgen Herre , Alexander Adami , Franz Reutelhuber
IPC分类号： G10L19/26 , G10L19/032 , H03G5/16 , H03G5/00 , G10L19/008
摘要： An audio post-processor for post-processing an audio signal having a time-variable high frequency gain information as side information includes: a band extractor for extracting a high frequency band of the audio signal and a low frequency band of the audio signal; a high band processor for performing a time-variable modification of the high frequency band in accordance with the time-variable high frequency gain information to obtain a processed high frequency band; and a combiner for combining the processed high frequency band and the low frequency band. Furthermore, a pre-processor is illustrated.
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公开(公告)号：US20200227061A1
公开(公告)日：2020-07-16
申请号：US16834930
申请日：2020-03-30
发明人： Mi-Suk LEE
IPC分类号： G10L19/26
摘要： The present invention relates to a codec device and method for encoding/decoding voice and audio signals in a communication system, wherein: a fixed codebook excited signal is generated by using a pulse index for a voice signal; a first adaptive codebook excited signal is generated by using a pitch index for the voice signal; a fixed codebook signal is generated by multiplying the fixed codebook excited signal by a fixed codebook gain; a first adaptive codebook signal is generated by multiplying the first adaptive codebook excited signal by a first adaptive codebook gain; and a synthesized filter excited signal is generated by adding the fixed codebook signal and the first adaptive codebook signal.
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公开(公告)号：US10692513B2
公开(公告)日：2020-06-23
申请号：US15956591
申请日：2018-04-18
摘要： The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.
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79.发明授权公开(公告)号：US10692510B2
公开(公告)日：2020-06-23
申请号：US15920907
申请日：2018-03-14
发明人： Johannes Fischer , Tom Bäckström , Emma Jokinen
IPC分类号： G10L19/012 , G10L21/0208 , G10L19/06 , G10L21/0216 , G10L21/0224 , G10L19/08 , G10L21/0308 , G10L21/02 , G10L19/005 , G10L25/12 , G10L19/26 , G10L19/16 , G10L21/0232 , G10L19/125
摘要： It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder includes a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder includes an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.
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公开(公告)号：US10657937B2
公开(公告)日：2020-05-19
申请号：US16376433
申请日：2019-04-05
发明人： Per Ekstrand , Lars Villemoes , Per Hedelin
IPC分类号： G10H1/00 , G10L21/038 , G10L19/26 , G10H1/12 , G10L21/0388
摘要： The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δf. The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf; with F being a resolution factor, with F≥1; wherein the transposition order P is different from the resolution factor F.
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