Method and device for decoding signals

    公开(公告)号:US12100401B2

    公开(公告)日:2024-09-24

    申请号:US18489875

    申请日:2023-10-19

    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.

    Audio coding method and apparatus

    公开(公告)号:US12057129B2

    公开(公告)日:2024-08-06

    申请号:US17697455

    申请日:2022-03-17

    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.

    AUDIO CODEC WITH ADAPTIVE GAIN CONTROL OF DOWNMIXED SIGNALS

    公开(公告)号:US20240153512A1

    公开(公告)日:2024-05-09

    申请号:US18548817

    申请日:2022-03-08

    IPC分类号: G10L19/008 G10L19/005

    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.

    FRAME ERROR CONCEALMENT
    7.
    发明公开

    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.

    Model based prediction in a critically sampled filterbank

    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

    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).