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    Wireless Communication System with Mobile Self-Healing
    1.
    发明公开
    Wireless Communication System with Mobile Self-Healing 审中-公开

    公开(公告)号：US20240244402A1

    公开(公告)日：2024-07-18

    申请号：US18289479

    申请日：2022-05-04

    Applicant: 3M INNOVATIVE PROPERTIES COMPANY ,  RTX A/S Company

    Inventor： Darin K. Thompson ,  Graham Wilson ,  Gareth D. Evans ,  Klaus A. Pedersen

    IPC: H04W4/08 ,  H04W4/90 ,  H04W84/20

    CPC classification number: H04W4/08 ,  H04W4/90 ,  H04W84/20

    Abstract: A method implemented in a first wireless unit of a plurality of wireless units of a first group. The method includes determining whether a second wireless unit of the first group is configured as the master of the first group. If the second wireless unit is configured as the master of the first group, the first wireless unit is configured as a slave and the first wireless unit is joined to the first group to communicate with the other wireless units. In addition, if the second wireless unit is not configured as the master of the first group, the first wireless unit is configured as the master of the first group and one of the first group is formed and the first group joined to communicate with the other wireless units. An apparatus is also provided.

    SYNCHRONIZATION OF MULTIPLE SIGNALS WITH AN UNPRECISE REFERENCE CLOCK
    2.
    发明公开
    SYNCHRONIZATION OF MULTIPLE SIGNALS WITH AN UNPRECISE REFERENCE CLOCK 审中-公开

    公开(公告)号：US20240187118A1

    公开(公告)日：2024-06-06

    申请号：US18550478

    申请日：2022-03-09

    Applicant: RTX A/S

    Inventor： Mads Berre ERIKSEN

    IPC: H04J3/06 ,  H04W56/00

    CPC classification number: H04J3/0644 ,  H04J3/0617 ,  H04W56/0015

    Abstract: The invention provides a method for synchronizing timing clocks in wireless communication, e.g. DECT based, between a timing master (P_l) and a timing slave (P_f). The timing master measures (MPD1) a phase offset between its internal clock and a reference clock, e.g. a network clock, e.g. IEEE 1588 over an Ethernet connection. Next, the timing master transmits (TPD1) data indicative of said measured phase offset to the timing slave or timing slaves, which also measure their phase offset between their internal clocks and the reference clock. Next, the timing slave(s) adjust (APD2) their phase offset between their respective internal clocks and the reference clock in response to the received data indicative of measured phase offset from the timing master. Thus, the timing slaves can ensure that they provides the same phase offset between their internal clocks and the reference clock as the timing master. Thus, DECT synchronization can be obtained in spite of a reference clock with a limited precision.

    Headset
    3.
    外观设计
    Headset 有权

    公开(公告)号：USD1008998S1

    公开(公告)日：2023-12-26

    申请号：US29866788

    申请日：2022-09-27

    Applicant: RTX A/S

    Designer： Jens Christian Lindof

    Abstract: FIG. 1 is a front perspective view of my design;
    FIG. 2 is a front view;
    FIG. 3 is a first side view;
    FIG. 4 is a second side view;
    FIG. 5 is a rear view;
    FIG. 6 is a top view; and,
    FIG. 7 is a bottom view.
    The broken lines in the Figures depict portions of the headset that form no part of the claimed design.

    AUDIO DATA BUFFERING FOR LOW LATENCY WIRELESS COMMUNICATION
    4.
    发明申请
    AUDIO DATA BUFFERING FOR LOW LATENCY WIRELESS COMMUNICATION 有权

    公开(公告)号：US20210266113A1

    公开(公告)日：2021-08-26

    申请号：US17253485

    申请日：2019-07-03

    Applicant: RTX A/S

    Inventor： Jens Toftgaard PETERSEN ,  Flemming MORTENSEN

    IPC: H04L1/18 ,  H04L1/16 ,  H04L5/14 ,  H04B1/06

    Abstract: A method for wireless RF transmission of short audio data blocks, e.g. 0.5 ms to 2 ms blocks, with low latency. The method involves a fixed part (FP) serving as synchronization master, and one or more portable parts (PP) being synchronization slaves in a time division scheme with fixed transmission intervals, and with a fixed and limited payload capacity of the RF transmission channel, such as 1.5-3 times the capacity required to transmit the audio data blocks in real- time. Short length transmission and receiving queues (TQ, RQ), e.g. having each 2-8 spaces for audio data blocks, for the audio data blocks are used to allow retransmission of blocks in—response to a missing acknowledge response from the portable part (PP). The queuing is operated so as to result in a fixed latency C determined e.g. by the transmission and receiving queue (TQ, RQ) lengths. A two-way audio scheme can be implemented following the same principle and utilizing the same RF transmission principles. The method provides a robust and low latency wireless audio interface suitable for dedicated audio devices and/or combined audio and Human Interface Devices (HIDs), e.g. for gaming equipment.

    AUDIO SIGNAL ENCODING AND DECODING
    5.
    发明申请
    AUDIO SIGNAL ENCODING AND DECODING 审中-公开

    公开(公告)号：US20200090672A1

    公开(公告)日：2020-03-19

    申请号：US16619039

    申请日：2018-06-15

    Applicant: RTX A/S

    Inventor： Jens Toftgaard PETERSEN

    IPC: G10L19/035 ,  G10L19/005 ,  G10L19/08

    Abstract: An audio codec suitable for robust wireless transmission of high quality audio with low latency, still at a moderate bit rate. The encoding and decoding methods are based on ADPCM and in addition to the encoded output bits APM, additional data QB are included in output data blocks, namely data QB representing an internal value of the adaptive quantization ADQ of the ADPCM encoding algorithm, especially a scaling factor encoded and truncated to such as 8 bits. Further, output data blocks preferably include data CFB representing an internal value of the predictor PR of the ADPCM encoding algorithm, especially data CFB representing coefficients of a lattice prediction FIR filter which, truncated to such as 8 bits, can be sequentially included in output data blocks. These additional data QB, CFB regarding internal values of the ADPCM encoding algorithm can be utilized at the encoder side to increase robustness against loss of data blocks in wireless transmission. Especially, the decoding algorithm may comprise comparing its current internal ADPCM decoding values corresponding to the received internal values QB, CFB from the encoder, and in case there is a difference, the decoder can adapt or overwrite its internal values to the ones received QB, CFB. This helps to ensure fast recovery after lost data blocks, thereby ensuring robustness against artefacts in the reconstructed signal, e.g. clicks in case of audio.

    JOINT FAR-END AND NEAR-END SPEECH INTELLIGIBILITY ENHANCEMENT
    8.
    发明申请
    JOINT FAR-END AND NEAR-END SPEECH INTELLIGIBILITY ENHANCEMENT 有权

    公开(公告)号：US20240404543A1

    公开(公告)日：2024-12-05

    申请号：US18698158

    申请日：2022-10-04

    Applicant: RTX A/S

    Inventor： Andreas Jonas FUGLSIG

    IPC: G10L21/0232 ,  G10L21/0216 ,  G10L21/034 ,  G10L21/0364 ,  H04M9/08

    Abstract: The invention relates to a computer implemented method for generation of a speech intelligibility enhancement algorithm for a wireless two-way communication system to enhance speech intelligibility in noise at both a near-end and a far-end taking into account joint near-end and far-end noise and audio inputs at the far-end from multiple microphones to capture speech and noise. First, determining (D_SI_OT) a speech intelligibility optimization target, taking into account noise at the near-end and noise at the far-end. Next, determining (D_MVDR) a Minimum Variance Distortionless Response (MVDR) beamformer with a plurality of inputs by optimizing a cost function according to the speech intelligibility optimization target to determine a global optimum. Next, determining (D_FB_G) a set of frequency band dependent gains by optimizing a cost function according to the speech intelligibility optimization target to determine a global optimum of a concave optimization formulation. Finally, generating (G_SIE_A) the speech enhancement processing algorithm as a linear processor with the determined MVDR beamformer followed by the determined set of frequency band dependent gains. In this way, a simple technical-mathematical formulation has been achieved, and the resulting speech intelligibility enhancement is similar to related but complex prior art solutions. The resulting algorithm is suited for wireless two-way communication devices, such as intercom devices to be used in noisy environments. e.g. for firefighters, rescue personnel etc.

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