PROTOCOL OVERHEAD REDUCTION FOR PACKET DATA CONVERGENCE PROTOCOL

    公开(公告)号:US20250023962A1

    公开(公告)日:2025-01-16

    申请号:US18898579

    申请日:2024-09-26

    Abstract: A wireless device may output a first message or a compression context message. The first message may include (a) a first payload associated with a first RTP-SN, (b) the first RTP SN, and (c) a first PDCP SN that is incrementally synchronized with the first RTP SN. The compression context message may include an indicator of compression context between the first RTP SN and the first PDCP SN. The wireless device may obtain a confirmation that a compression context between the first RTP SN and the first PDCP SN is established at a receiver. In response, the wireless device may output a second message. The second message may have a second payload associated with a second RTP SN and a second PDCP SN that is incrementally synchronized with the second RTP SN. The second message does not comprise a ROHC header having an RTP SN field.

    AUDIO SIGNAL RECONSTRUCTION
    12.
    发明公开

    公开(公告)号:US20240355344A1

    公开(公告)日:2024-10-24

    申请号:US18684074

    申请日:2022-09-09

    CPC classification number: G10L19/0204 G10L19/032 G10L25/18 G10L25/30

    Abstract: A method includes receiving audio data that includes magnitude spectrum data descriptive of an audio signal. The method also includes providing the audio data as input to a neural network to generate an initial phase estimate for one or more samples of the audio signal. The method further includes determining, using a phase estimation algorithm, target phase data for the one or more samples of the audio signal based on the initial phase estimate and a magnitude spectrum of the one or more samples of the audio signal indicated by the magnitude spectrum data. The method also includes reconstructing the audio signal based on a target phase of the one or more samples of the audio signal indicated by the target phase data and based on the magnitude spectrum.

    SPLIT-DOMAIN SPEECH SIGNAL ENHANCEMENT
    14.
    发明申请

    公开(公告)号:US20190341067A1

    公开(公告)日:2019-11-07

    申请号:US15973214

    申请日:2018-05-07

    Abstract: A method and an apparatus for estimating speech signal in split-domain is disclosed. The method includes performing LP analysis on a noisy speech signal to generate a first plurality of LPC and a first residual signal. The method also includes estimating speech LPC spectrum to generate cleaned LPC. The method further includes estimating speech residual spectrum to generate cleaned residual signal. The method also includes synthesizing output signals based on the cleaned LPC and the cleaned residual signal.

    EFFICIENT PACKET-LOSS PROTECTED DATA ENCODING AND/OR DECODING

    公开(公告)号:US20250090963A1

    公开(公告)日:2025-03-20

    申请号:US18294490

    申请日:2022-09-08

    Abstract: A device includes a memory and one or more processors coupled to the memory and configured to execute instructions from the memory. Execution of the instructions causes the one or more processors to combine two or more data portions to generate input data for a decoder network. A first data portion of the two or more data portions is based on a first encoding of a data sample by a multiple description coding network and content of a second data portion of the two or more data portions depends on whether data based on a second encoding of the data sample by the multiple description coding network is available. Execution of the instructions also causes the one or more processors to obtain, from the decoder network, output data based on the input data and to generate a representation of the data sample based on the output data.

    SAMPLE GENERATION USING PIPELINED PROCESSING UNITS

    公开(公告)号:US20250046295A1

    公开(公告)日:2025-02-06

    申请号:US18718939

    申请日:2022-11-28

    Abstract: A device includes a memory configured to store instructions and a processor coupled to the memory. The processor includes a first processing unit configured to perform a first stage of a sample synthesis operation. The processor includes a second processing unit configured to perform a second stage of the sample synthesis operation based on an output of the first processing unit. The processor also includes a sample synthesizer configured to process input data, using the first processing unit and the second processing unit, to generate output data. The first processing unit and the second processing unit are configured to operate in a pipelined configuration that includes performance of the second stage at the second processing unit in parallel with performance of the first stage at the first processing unit.

    SELECTIVE PROCESSING OF SEGMENTS OF TIME-SERIES DATA BASED ON SEGMENT CLASSIFICATION

    公开(公告)号:US20250037734A1

    公开(公告)日:2025-01-30

    申请号:US18360981

    申请日:2023-07-28

    Abstract: A device includes a memory configured to store one or more segments of time-series data. The device also includes one or more processors configured to generate, using a feature extractor, a latent-space representation of a segment of the time-series data. The one or more processors are also configured to provide one or more inputs to a classifier, the one or more inputs including at least one input based on the latent-space representation. The one or more processors are also configured to generate, based on output of the classifier, a processing control signal for the segment.

    SYSTEMS AND METHODS FOR MULTI-BAND AUDIO CODING

    公开(公告)号:US20240371384A1

    公开(公告)日:2024-11-07

    申请号:US18689052

    申请日:2022-10-10

    Abstract: Systems and techniques are described for audio coding. An audio system receives feature(s) corresponding an audio signal, for example from an encoder and/or a speech synthesis engine. The audio system generates an excitation signal, such as a harmonic signal and/or a noise signal, based on the feature(s). The audio system uses a filterbank to generate band-specific signals from the excitation signal. The band-specific signals correspond to frequency bands. The audio system inputs the feature(s) into a machine learning (ML) filter estimator to generate parameter(s) associated with linear filter(s). The audio system inputs the feature(s) into a voicing estimator to generate gain value(s). The audio system generates an output audio signal based on modification of the band-specific signals, application of the linear filter(s) according to the parameter(s), and amplification using the gain amplifier(s) according to the gain value(s).

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