公开(公告)号：US20240331709A1

公开(公告)日：2024-10-03

申请号：US18355928

申请日：2023-07-20

Applicant: GOOGLE LLC

Inventor： Sze Chie Lim ,  Shawn Singh ,  Anjali Wheeler ,  Jani Huoponen ,  Jan Skoglund

IPC: G10L19/02

CPC classification number: G10L19/02

Abstract: A method including receiving first audio data, receiving second audio data, compressing the first audio data as first compressed audio data, compressing the second audio data as second compressed audio data, generating a codec dependent container including a parameter associated with compressing the first audio data, compressing the second audio data, a reference to the first compressed audio data, and a reference to the second compressed audio data, generating a codec agnostic container including at least one parameter representing time-varying data associated with playback of the first audio data and the second audio data, and generating an audio package including the codec dependent container and the codec agnostic container.

公开(公告)号：US11443756B2

公开(公告)日：2022-09-13

申请号：US16934801

申请日：2020-07-21

Applicant: Google LLC

Inventor： Simon J. Godsill ,  Herbert Buchner ,  Jan Skoglund

IPC: G10L15/20 ,  H04M9/08 ,  G10L21/0208 ,  H04R3/00 ,  G10L21/0216 ,  G10L21/0272

Abstract: Provided are methods and systems for enhancing speech when corrupted by transient noise (e.g., keyboard typing noise). The methods and systems utilize a reference microphone input signal for the transient noise in a signal restoration process used for the voice part of the signal. A robust Bayesian statistical model is used to regress the voice microphone on the reference microphone, which allows for direct inference about the desired voice signal while marginalizing the unwanted power spectral values of the voice and transient noise. Also provided is a straightforward and efficient Expectation-maximization (EM) procedure for fast enhancement of the corrupted signal. The methods and systems are designed to operate easily in real-time on standard hardware, and have very low latency so that there is no irritating delay in speaker response.

公开(公告)号：US20210287038A1

公开(公告)日：2021-09-16

申请号：US17250506

申请日：2019-05-16

Applicant: Google LLC

Inventor： Willem Bastiaan Kleijn ,  Sze Chie Lim ,  Michael Chinen ,  Jan Skoglund

IPC: G06K9/62 ,  G06N3/08 ,  G06N3/04

Abstract: Implementations identify a small set of independent, salient features from an input signal. The salient features may be used for conditioning a generative network, making the generative network robust to noise. The salient features may facilitate compression and data transmission. An example method includes receiving an input signal and extracting salient features for the input signal by providing the input signal to an encoder trained to extract salient features. The salient features may be independent and have a sparse distribution. The encoder may be configured to generate almost identical features from two input signals a system designer deems equivalent. The method also includes conditioning a generative network using the salient features. In some implementations, the method may also include extracting a plurality of time sequences from the input signal and extracting the salient features for each time sequence.

公开(公告)号：US10672405B2

公开(公告)日：2020-06-02

申请号：US15973287

申请日：2018-05-07

Applicant: GOOGLE LLC

Inventor： Andrew Hines ,  Jan Skoglund ,  Andrew Allen ,  Miroslaw Narbutt

IPC: G10L19/00 ,  G10L19/008 ,  G10L19/022 ,  G10L19/16

Abstract: A computing device includes a processor and a memory. The processor is configured to generate spectrograms, for example, using short-time Fourier transform, for a plurality of channels of reference and test ambisonic signals. In some implementations, the test ambisonic signal may be generated by decoding an encoded version of the reference ambisonic signal. The processor is further configured to compare, for each of the plurality of channels of a reference ambisonic signal, at least a patch associated with a channel of the reference ambisonic signal with at least a corresponding patch of a corresponding channel of the test ambisonic signal and determine a localization accuracy of the test ambisonic signal based on the comparison. In some implementations, the comparing may be based on phaseograms of the reference and test ambisonic signals.

公开(公告)号：US20190096418A1

公开(公告)日：2019-03-28

申请号：US16197645

申请日：2018-11-21

Applicant: GOOGLE LLC

Inventor： Minyue Li ,  Willem Bastiaan Kleijn ,  Jan Skoglund

IPC: G10L19/24 ,  G10L19/008 ,  G10L19/16 ,  G10L19/035 ,  G10L19/02

CPC classification number: G10L19/24 ,  G10L19/008 ,  G10L19/0212 ,  G10L19/035 ,  G10L19/167

Abstract: Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled.

公开(公告)号：US11380342B2

公开(公告)日：2022-07-05

申请号：US16780506

申请日：2020-02-03

Applicant: GOOGLE LLC

Inventor： Minyue Li ,  Willem Bastiaan Kleijn ,  Jan Skoglund

IPC: G10L19/00 ,  G10L19/24 ,  G10L19/008 ,  G10L19/02 ,  G10L19/035 ,  G10L19/16

Abstract: Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled.

公开(公告)号：US11297424B2

公开(公告)日：2022-04-05

申请号：US16624704

申请日：2018-10-10

Applicant: GOOGLE LLC

Inventor： Willem Bastiaan Kleijn ,  Jan Skoglund ,  Christos Tzagkarakis

IPC: H04R3/00 ,  H04R1/40

Abstract: Techniques of source localization and acquisition involve a wideband joint acoustic source localization and acquisition approach in light of sparse optimization framework based on an orthogonal matching pursuit-based grid-shift procedure. Along these lines, a specific grid structure is constructed with the same number of grid points as compared to the on-grid case, but which is “shifted” across the acoustic scene. More specifically, it is expected that each source will be located close to a grid point in at least one of the set of shifted grids. The sparse solutions corresponding to the set of shifted grids are combined to obtain the source location estimates. The estimated source positions are used as side information to obtain the original source signals.

公开(公告)号：US10861479B2

公开(公告)日：2020-12-08

申请号：US16598462

申请日：2019-10-10

Applicant: GOOGLE LLC

Inventor： Turaj Zakizadeh Shabestary ,  Willem Bastiaan Kleijn ,  Jan Skoglund

IPC: G10L21/0208 ,  G10L21/0232 ,  G10L15/08 ,  H04R3/04 ,  H04M9/08

Abstract: Techniques of performing linear acoustic echo cancellation performing a phase correction operation on the estimate of the echo signal based on a clock drift between a capture of an input microphone signal and a playout of a loudspeaker signal. Along these lines, the existence of the clock drift, i.e., a small difference in the sampling rates of the input microphone signal and the loudspeaker signal, can cause processing circuitry in a device configured to perform LAEC operations to generate a filter based on the magnitudes of the short-term Fourier transforms (STFTs) of the input microphone signal and the loudspeaker signal. Such a filter is real-valued and results in a positive estimate of the acoustic echo signal included in the input microphone signal. The phase of this estimate may then be aligned with the phase of the input microphone signal.

公开(公告)号：US20200349964A1

公开(公告)日：2020-11-05

申请号：US16934801

申请日：2020-07-21

Applicant: Google LLC

Inventor： Simon J. Godsill ,  Herbert Buchner ,  Jan Skoglund

IPC: G10L21/0208 ,  H04R3/00 ,  G10L21/0216 ,  G10L21/0272

Abstract: Provided are methods and systems for enhancing speech when corrupted by transient noise (e.g., keyboard typing noise). The methods and systems utilize a reference microphone input signal for the transient noise in a signal restoration process used for the voice part of the signal. A robust Bayesian statistical model is used to regress the voice microphone on the reference microphone, which allows for direct inference about the desired voice signal while marginalizing the unwanted power spectral values of the voice and transient noise. Also provided is a straightforward and efficient Expectation-maximization (EM) procedure far fast enhancement of the corrupted signal. The methods and systems are designed to operate easily in real-time on standard hardware, and have very low latency so that there is no irritating delay in speaker response.

公开(公告)号：US20180174598A1

公开(公告)日：2018-06-21

申请号：US15846049

申请日：2017-12-18

Applicant: GOOGLE LLC

Inventor： Turaj Zakizadeh Shabestary ,  Willem Bastiaan Kleijn ,  Jan Skoglund

IPC: G10L21/0232 ,  G10L15/08 ,  H04R3/04

CPC classification number: G10L21/0232 ,  G10L15/08 ,  G10L21/0208 ,  G10L2015/088 ,  G10L2021/02082 ,  H04M9/082 ,  H04R3/04

Abstract: Techniques of performing linear acoustic echo cancellation performing a phase correction operation on the estimate of the echo signal based on a clock drift between a capture of an input microphone signal and a playout of a loudspeaker signal. Along these lines, the existence of the clock drift, i.e., a small difference in the sampling rates of the input microphone signal and the loudspeaker signal, can cause processing circuitry in a device configured to perform LAEC operations to generate a filter based on the magnitudes of the short-term Fourier transforms (STFTs) of the input microphone signal and the loudspeaker signal. Such a filter is real-valued and results in a positive estimate of the acoustic echo signal included in the input microphone signal. The phase of this estimate may then be aligned with the phase of the input microphone signal.