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公开(公告)号:US11640827B2
公开(公告)日:2023-05-02
申请号:US17367009
申请日:2021-07-02
发明人: Tom Baeckstroem , Christian Fischer Pedersen , Johannes Fischer , Matthias Huettenberger , Alfonso Pino
摘要: An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P ( z ) = A ( z ) + z - m - l A ( z - 1 ) and Q ( z ) = A ( z ) - z - m - l A ( z - 1 ) , wherein m is an order of the predictive polynomial A(z) and I is greater or equal to zero, wherein the converter is configured to obtain the frequency values by establishing a strictly real spectrum derived from P(z) and a strictly imaginary spectrum from Q(z) and by identifying zeros of the strictly real spectrum derived from P(z) and the strictly imaginary spectrum derived from Q(z).
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公开(公告)号:US11581003B2
公开(公告)日:2023-02-14
申请号:US16885109
申请日:2020-05-27
IPC分类号: G10L19/00 , G10L19/26 , G10L19/025 , G10L19/028 , G10L19/12 , G10L19/22 , G10L25/21 , G10L25/90
摘要: The coding efficiency of an audio codec using a controllable—switchable or even adjustable—harmonic filter tool is improved by performing the harmonicity-dependent controlling of this tool using a temporal structure measure in addition to a measure of harmonicity in order to control the harmonic filter tool. In particular, the temporal structure of the audio signal is evaluated in a manner which depends on the pitch. This enables to achieve a situation-adapted control of the harmonic filter tool so that in situations where a control made solely based on the measure of harmonicity would decide against or reduce the usage of this tool, although using the harmonic filter tool would, in that situation, increase the coding efficiency, the harmonic filter tool is applied, while in other situations where the harmonic filter tool may be inefficient or even destructive, the control reduces the appliance of the harmonic filter tool appropriately.
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公开(公告)号:US11430456B2
公开(公告)日:2022-08-30
申请号:US16999448
申请日:2020-08-21
摘要: An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system, where the encoding method includes dividing a to-be-encoded time-domain signal into a low band signal and a high band signal, performing encoding on the low band signal to obtain a low frequency encoding parameter, performing encoding on the high band signal to obtain a high frequency encoding parameter, obtaining a synthesized high band signal; performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal, and calculating a high frequency gain based on the high band signal and the short-time filtering 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:US20220157328A1
公开(公告)日:2022-05-19
申请号:US17592423
申请日:2022-02-03
发明人: Emmanuel RAVELLI , Manuel JANDER , Grzegorz PIETRZYK , Martin DIETZ , Marc GAYER
IPC分类号: G10L19/26 , G10L19/022 , G10L19/20 , G10L19/12
摘要: A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.
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公开(公告)号:US11276411B2
公开(公告)日:2022-03-15
申请号:US16647801
申请日:2018-09-20
申请人: VOICEAGE CORPORATION
发明人: Vaclav Eksler
IPC分类号: G10L19/12 , G10L19/24 , G10L19/002 , G10L19/038
摘要: A method and device for allocating a bit-budget to a plurality of first parts and to a second part of a CELP core module of (a) an encoder for encoding a sound signal or (b) a decoder for decoding the sound signal. In a frame of the sound signal comprising sub-frames, respective bit-budgets are allocated to the first CELP core module parts and a bit-budget remaining after allocating to the first CELP core module parts their respective bit-budgets is allocated to the second CELP core module part. According to an alternative, the second CELP core module part bit-budget is distributed between the sub-frames of the frame and a larger bit-budget is allocated to at least one of the sub-frames of the frame. The at least one sub-frame may be the first sub-frame of the frame, at least one sub-frame following the first sub-frame, or the sub-frame using a glottal-impulse-shape codebook.
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公开(公告)号:US20220076685A1
公开(公告)日:2022-03-10
申请号:US17479151
申请日:2021-09-20
发明人: Emmanuel Ravelli , Guillaume Fuchs , Sascha Disch , Markus Multrus , Grzegorz Pietrzyk , Benjamin Schubert
摘要: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a linear-prediction-domain decoder configured to provide a first decoded audio information on the basis of an audio frame encoded in a linear prediction domain, a frequency domain decoder configured to provide a second decoded audio information on the basis of an audio frame encoded in a frequency domain, and a transition processor. The transition processor is configured to obtain a zero-input-response of a linear predictive filtering, wherein an initial state of the linear predictive filtering is defined depending on the first decoded audio information and the second decoded audio information, and modify the second decoded audio information depending on the zero-input-response, to obtain a smooth transition between the first and the modified second decoded audio information.
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公开(公告)号:US11270714B2
公开(公告)日:2022-03-08
申请号:US16737543
申请日:2020-01-08
发明人: Thomas Clark
摘要: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into frames including N subframes (where N is an integer greater than 1); computing model parameters for the subframes, the model parameters including spectral parameters; and generating a representation of the frame. The representation includes information representing the spectral parameters of P subframes (where P is an integer and P
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公开(公告)号:US20210390968A1
公开(公告)日:2021-12-16
申请号:US17458879
申请日:2021-08-27
IPC分类号: G10L19/12 , G10L19/06 , G10L19/025
摘要: A method comprises determining a first modification weight according to linear spectral frequency (LSF) differences of the current frame and LSF differences of a previous frame of the current frame when a signal characteristic of the current frame meets a preset modification condition, modifying the linear predictive parameter of the current frame according to the determined first modification weight, and coding the current frame according to the modified linear predictive parameter.
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公开(公告)号:US20210375296A1
公开(公告)日:2021-12-02
申请号:US17444799
申请日:2021-08-10
申请人: VOICEAGE EVS LLC
发明人: Redwan SALAMI , Vaclav EKSLER
摘要: Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.
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