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    Disambiguation of a spoken query term
    31.
    发明授权
    Disambiguation of a spoken query term 有权

    公开(公告)号：US10210267B1

    公开(公告)日：2019-02-19

    申请号：US15233141

    申请日：2016-08-10

    Applicant: Google LLC

    Inventor： Matthew I. Lloyd ,  Johan Schalkwyk ,  Pankaj Risbood

    IPC: G10L15/22 ,  G06F17/30 ,  G10L15/183 ,  G10L15/01

    Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for processing spoken query terms. In one aspect, a method includes performing speech recognition on an audio signal to select two or more textual, candidate transcriptions that match a spoken query term, and to establish a speech recognition confidence value for each candidate transcription, obtaining a search history for a user who spoke the spoken query term, where the search history references one or more past search queries that have been submitted by the user, generating one or more n-grams from each candidate transcription, where each n-gram is a subsequence of n phonemes, syllables, letters, characters, words or terms from a respective candidate transcription, and determining, for each n-gram, a frequency with which the n-gram occurs in the past search queries, and a weighting value that is based on the respective frequency.

    Structured Video Documents
    32.
    发明申请
    Structured Video Documents 有权

    公开(公告)号：US20250094491A1

    公开(公告)日：2025-03-20

    申请号：US18961038

    申请日：2024-11-26

    Applicant: Google LLC

    Inventor： Johan Schalkwyk ,  Francoise Beaufays

    IPC: G06F16/783 ,  G06F16/738 ,  G06F40/169 ,  G06F40/30

    Abstract: A method includes receiving a content feed that includes audio data corresponding to speech utterances and processing the content feed to generate a semantically-rich, structured document. The structured document includes a transcription of the speech utterances and includes a plurality of words each aligned with a corresponding audio segment of the audio data that indicates a time when the word was recognized in the audio data. During playback of the content feed, the method also includes receiving a query from a user requesting information contained in the content feed and processing, by a large language model, the query and the structured document to generate a response to the query. The response conveys the requested information contained in the content feed. The method also includes providing, for output from a user device associated with the user, the response to the query.

    Using corrections, of predicted textual segments of spoken utterances, for training of on-device speech recognition model
    33.
    发明授权
    Using corrections, of predicted textual segments of spoken utterances, for training of on-device speech recognition model 有权

    公开(公告)号：US12183321B2

    公开(公告)日：2024-12-31

    申请号：US18377122

    申请日：2023-10-05

    Applicant: GOOGLE LLC

    Inventor： Françoise Beaufays ,  Johan Schalkwyk ,  Giovanni Motta

    IPC: G10L15/00 ,  G06F3/04842 ,  G06F3/04883 ,  G10L25/51

    Abstract: Processor(s) of a client device can: receive audio data that captures a spoken utterance of a user of the client device; process, using an on-device speech recognition model, the audio data to generate a predicted textual segment that is a prediction of the spoken utterance; cause at least part of the predicted textual segment to be rendered (e.g., visually and/or audibly); receive further user interface input that is a correction of the predicted textual segment to an alternate textual segment; and generate a gradient based on comparing at least part of the predicted output to ground truth output that corresponds to the alternate textual segment. The gradient is used, by processor(s) of the client device, to update weights of the on-device speech recognition model and/or is transmitted to a remote system for use in remote updating of global weights of a global speech recognition model.

    Unsupervised federated learning of machine learning model layers
    35.
    发明授权
    Unsupervised federated learning of machine learning model layers 有权

    公开(公告)号：US12014724B2

    公开(公告)日：2024-06-18

    申请号：US16973605

    申请日：2020-07-20

    Applicant: Google LLC

    Inventor： Françoise Beaufays ,  Khe Chai Sim ,  Johan Schalkwyk

    IPC: G10L15/06 ,  G10L15/187 ,  G10L15/22 ,  G10L15/30

    CPC classification number: G10L15/063 ,  G10L15/187 ,  G10L15/22 ,  G10L15/30 ,  G10L2015/0635

    Abstract: Implementations disclosed herein are directed to unsupervised federated training of global machine learning (“ML”) model layers that, after the federated training, can be combined with additional layer(s), thereby resulting in a combined ML model. Processor(s) can: detect audio data that captures a spoken utterance of a user of a client device; process, using a local ML model, the audio data to generate predicted output(s); generate, using unsupervised learning locally at the client device, a gradient based on the predicted output(s); transmit the gradient to a remote system; update weight(s) of the global ML model layers based on the gradient; subsequent to updating the weight(s), train, using supervised learning remotely at the remote system, a combined ML model that includes the updated global ML model layers and additional layer(s); transmit the combined ML model to the client device; and use the combined ML model to make prediction(s) at the client device.

    EXPORTING MODULAR ENCODER FEATURES FOR STREAMING AND DELIBERATION ASR
    36.
    发明公开
    EXPORTING MODULAR ENCODER FEATURES FOR STREAMING AND DELIBERATION ASR 审中-公开

    公开(公告)号：US20240144917A1

    公开(公告)日：2024-05-02

    申请号：US18494763

    申请日：2023-10-25

    Applicant: Google LLC

    Inventor： Rami Magdi Fahmi Botros ,  Rohit Prakash Prabhavalkar ,  Johan Schalkwyk ,  Tara N. Sainath ,  Ciprian Ioan Chelba ,  Francoise Beaufays

    IPC: G10L15/16

    CPC classification number: G10L15/16

    Abstract: A method includes obtaining a base encoder from a pre-trained model, and receiving training data comprising a sequence of acoustic frames characterizing an utterance paired with a ground-truth transcription of the utterance. At each of a plurality of output steps, the method includes: generating, by the base encoder, a first encoded representation for a corresponding acoustic frame; generating, by an exporter network configured to receive a continuous sequence of first encoded representations generated by the base encoder, a second encoded representation for a corresponding acoustic frame; generating, by an exporter decoder, a probability distribution over possible logits; and determining an exporter decoder loss based on the probability distribution over possible logits generated by the exporter decoder at the corresponding output step and the ground-truth transcription. The method also includes training the exporter network based on the exporter decoder losses while parameters of the base encoder are frozen.

    Methods and systems for attending to a presenting user
    39.
    发明授权
    Methods and systems for attending to a presenting user 有权

    公开(公告)号：US11789697B2

    公开(公告)日：2023-10-17

    申请号：US17370656

    申请日：2021-07-08

    Applicant: GOOGLE LLC

    Inventor： Yuan Yuan ,  Johan Schalkwyk ,  Kenneth Mixter

    IPC: G10L15/22 ,  G06F3/16 ,  H04N7/14 ,  H04N7/15 ,  G06V40/16

    CPC classification number: G06F3/167 ,  G06F3/165 ,  G06V40/174 ,  G10L15/22 ,  H04N7/142 ,  H04N7/15

    Abstract: The various implementations described herein include methods, devices, and systems for attending to a presenting user. In one aspect, a method is performed at an electronic device that includes an image sensor, microphones, a display, processor(s), and memory. The device (1) obtains audio signals by concurrently receiving audio data at each microphone; (2) determines based on the obtained audio signals that a person is speaking in a vicinity of the device; (3) obtains video data from the image sensor; (4) determines via the video data that the person is not within a field of view of the image sensor; (5) reorients the electronic device based on differences in the received audio data; (6) after reorienting the electronic device, obtains second video data from the image sensor and determines that the person is within the field of view; and (7) attends to the person by directing the display toward the person.

    ENABLING NATURAL CONVERSATIONS WITH SOFT ENDPOINTING FOR AN AUTOMATED ASSISTANT
    40.
    发明申请
    ENABLING NATURAL CONVERSATIONS WITH SOFT ENDPOINTING FOR AN AUTOMATED ASSISTANT 有权

    公开(公告)号：US20230053341A1

    公开(公告)日：2023-02-23

    申请号：US17532819

    申请日：2021-11-22

    Applicant: GOOGLE LLC

    Inventor： Jaclyn Konzelmann ,  Trevor Strohman ,  Jonathan Bloom ,  Johan Schalkwyk ,  Joseph Smarr

    IPC: G10L15/22 ,  G10L15/18 ,  G08B5/36 ,  G06N20/00

    Abstract: As part of a dialog session between a user and an automated assistant, implementations can process, using a streaming ASR model, a stream of audio data that captures a portion of a spoken utterance to generate ASR output, process, using an NLU model, the ASR output to generate NLU output, and cause, based on the NLU output, a stream of fulfillment data to be generated. Further, implementations can further determine, based on processing the stream of audio data, audio-based characteristics associated with the portion of the spoken utterance captured in the stream of audio data. Based on the audio-based characteristics and/the stream of NLU output, implementations can determine whether the user has paused in providing the spoken utterance or has completed providing of the spoken utterance. If the user has paused, implementations can cause natural conversation output to be provided for presentation to the user.

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