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公开(公告)号:US12125491B2
公开(公告)日:2024-10-22
申请号:US17100247
申请日:2020-11-20
IPC分类号: G10L19/005 , G10L19/00 , G10L19/002 , G10L19/012 , G10L19/02 , G10L19/06 , G10L19/07 , G10L19/083 , G10L19/09 , G10L19/12 , G10L19/22
CPC分类号: G10L19/005 , G10L19/002 , G10L19/012 , G10L19/06 , G10L19/07 , G10L19/083 , G10L19/09 , G10L19/12 , G10L19/22 , G10L2019/0002 , G10L2019/0011 , G10L2019/0016 , G10L19/0212
摘要: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame.
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公开(公告)号:US12100401B2
公开(公告)日:2024-09-24
申请号:US18489875
申请日:2023-10-19
发明人: Zexin Liu , Fengyan Qi , Lei Miao
IPC分类号: G10L19/002 , G10L19/005 , G10L19/02 , G10L19/028
CPC分类号: G10L19/002 , G10L19/005 , G10L19/0204 , G10L19/028
摘要: In a method to decode signals, a computing device decodes spectral coefficients of a current frame are grouped into a plurality of sub-bands. The computing device classifies a sub-band as a bit allocation unsaturated sub-band based on an average quantity of allocated bits per spectral coefficient of a sub-band of the plurality of sub-bands and a threshold. The computing device obtains a noise filling gain based on an envelope of the sub-band, and obtains a reconstructed spectral coefficient of the sub-band by performing noise filling based on the noise filling gain. The computing device then obtains a frequency domain audio signal based on spectral coefficients in the sub-band obtained by decoding and the reconstructed spectral coefficient.
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公开(公告)号:US12057129B2
公开(公告)日:2024-08-06
申请号:US17697455
申请日:2022-03-17
IPC分类号: G10L19/005 , G06N3/08 , G10L19/008 , G10L19/16 , H04R3/12
CPC分类号: G10L19/005 , G06N3/08 , G10L19/008 , G10L19/167 , H04R3/12 , H04R2420/07
摘要: An audio coding method and apparatus are provided. The audio coding method includes: obtaining first audio data; obtaining a target bit rate and a Bluetooth packet type, where the target bit rate and the Bluetooth packet type correspond to a current status of a Bluetooth channel; obtaining one or more of a bit pool parameter set, a psychoacoustic parameter set, and a spectrum bandwidth parameter set by using a neural network obtained through pre-training based on the first audio data, the target bit rate, and the Bluetooth packet type; and coding the first audio data based on one or more of the bit pool parameter set, the psychoacoustic parameter set, and the spectrum bandwidth parameter set to obtain a to-be-sent bit stream. The status of the Bluetooth channel can be adaptively matched, and continuous audio listening experience is provided when audio quality is maximally ensured.
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公开(公告)号:US12014746B2
公开(公告)日:2024-06-18
申请号:US17580578
申请日:2022-01-20
发明人: Emmanuel Ravelli , Manuel Jander , Grzegorz Pietrzyk , Martin Dietz , Marc Gayer
IPC分类号: G10L19/005 , G10L19/022 , G10L19/03 , G10L19/12 , G10L19/20 , G10L19/26 , H04B1/10 , G10L21/0364 , G11B27/038
CPC分类号: G10L19/26 , G10L19/005 , G10L19/022 , G10L19/03 , G10L19/12 , G10L19/20 , G10L21/0364 , G11B27/038 , H04B1/1027
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:US20240194209A1
公开(公告)日:2024-06-13
申请号:US18545607
申请日:2023-12-19
IPC分类号: G10L19/02 , G10L19/005 , G10L19/26
CPC分类号: G10L19/0204 , G10L19/005 , G10L19/26
摘要: An apparatus for processing an audio input signal to obtain an audio output signal according to an embodiment. The apparatus has a signal analyser configured for determining information on an auditory roughness of one or more spectral bands of the audio input signal. Moreover, the apparatus has a signal processor configured for processing the audio input signal depending on the information on the auditory roughness of the one or more spectral bands.
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公开(公告)号:US20240153512A1
公开(公告)日:2024-05-09
申请号:US18548817
申请日:2022-03-08
发明人: Panji Setiawan , Rishabh Tyagi , Stefan Bruhn
IPC分类号: G10L19/008 , G10L19/005
CPC分类号: G10L19/008 , G10L19/005 , G10L19/002
摘要: A method for performing gain control on audio signals is provided. In some implementations, the method involves determining downmixed signals associated with one or more downmix channels associated with a current frame of an audio signal to be encoded. In some implementations, the method involves determining whether an overload condition exists for an encoder. In some implementation, the method involves determining a gain parameter. In some implementations, the method involves determining at least one gain transition function based on the gain parameter and a gain parameter associated with a preceding frame of the audio signal. In some implementations, the method involves applying the at least one gain transition function to one or more of the downmixed signals. In some implementations, the method involves encoding the downmixed signals in connection with information indicative of gain control applied to the current frame.
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公开(公告)号:US20240144939A1
公开(公告)日:2024-05-02
申请号:US18386020
申请日:2023-11-01
IPC分类号: G10L19/005 , G10L19/025
CPC分类号: G10L19/005 , G10L19/025
摘要: A frame error concealment method based on frames including transform coefficient vectors including the following steps: It tracks sign changes between corresponding transform coefficients of predetermined sub-vectors of consecutive good stationary frames. It accumulates the number of sign changes in corresponding sub-vectors of a predetermined number of consecutive good stationary frames. It reconstructs an erroneous frame with the latest good stationary frame, but with reversed signs of transform coefficients in sub-vectors having an accumulated number of sign changes that exceeds a predetermined threshold.
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公开(公告)号:US20240096334A1
公开(公告)日:2024-03-21
申请号:US17932650
申请日:2022-09-15
发明人: Brandon Sangston
IPC分类号: G10L19/008 , G10L19/005
CPC分类号: G10L19/008 , G10L19/005
摘要: Ambisonics audio such as may be used for computer simulations such as computer games is improved by using multi-order optimizations that frame an optimization problem that minimizes a cost function across a subset of Ambisonics orders for a chosen Ambisonics order “N”. In a simple form, this cost function minimizes error across all orders (0
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公开(公告)号:US20240087577A1
公开(公告)日:2024-03-14
申请号:US18014924
申请日:2021-07-02
发明人: Seung Kwon BEACK , Jongmo SUNG , Mi Suk LEE , Tae Jin LEE , Woo-taek LIM , Inseon JANG
IPC分类号: G10L19/005 , G10L19/16
CPC分类号: G10L19/005 , G10L19/167
摘要: Disclosed is an apparatus and method for audio encoding/decoding that is robust against coding distortion in a transition section. An audio encoding method includes outputting a frequency domain signal by time-to-frequency (T/F) transform of an input signal, outputting a frequency domain residual signal in which a frequency axis envelope is removed from the frequency domain signal by applying frequency domain noise shaping (FDNS) encoding to the frequency domain signal, outputting a time domain residual signal in which a time axis envelope is removed by performing linear prediction coefficient (LPC) analysis based on the frequency domain residual signal, and quantizing and transmitting the time domain residual signal.
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公开(公告)号:US11915713B2
公开(公告)日:2024-02-27
申请号:US18128494
申请日:2023-03-30
发明人: Lars Villemoes
IPC分类号: G10L19/005 , G10L19/02 , G10L19/093 , G06F30/30 , G06F30/327 , G10L19/032 , G10L19/06 , G10L19/26 , G06F111/08
CPC分类号: G10L19/0208 , G06F30/30 , G06F30/327 , G10L19/005 , G10L19/0212 , G10L19/032 , G10L19/06 , G10L19/093 , G10L19/26 , G10L19/265 , G06F2111/08
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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