Patents by Inventor Evelyn Kurniawati
Evelyn Kurniawati has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).
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Patent number: 11227601Abstract: A computer-implement voice command authentication method is provided. The method includes obtaining a sound signal stream; calculating a Signal-to-Noise Ratio (SNR) value of the sound signal stream; converting the sound signal stream into a Mel-Frequency Cepstral Coefficients (MFCC) stream; calculating a Dynamic Time Warping (DTW) distance corresponding to the MFCC stream according to the MFCC stream and one of a plurality of sample streams generated by the Gaussian Mixture Model with Universal Background Model (GMM-UBM); calculating, according to the MFCC stream and the sample streams, a Log-likelihood ratio value corresponding to the MFCC stream as a GMM-UBM score; determining whether the sound signal stream passes a voice command authentication according to the GMM-UBM score, the DTW distance and the SNR value; in response to determining that the sound signal stream passes the voice command authentication, determining that the sound signal stream is a voice stream spoken from a legal user.Type: GrantFiled: September 21, 2019Date of Patent: January 18, 2022Assignee: Merry Electronics(Shenzhen) Co., Ltd.Inventors: Evelyn Kurniawati, Sasiraj Somarajan
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Publication number: 20210090577Abstract: A computer-implement voice command authentication method is provided. The method includes obtaining a sound signal stream; calculating a Signal-to-Noise Ratio (SNR) value of the sound signal stream; converting the sound signal stream into a Mel-Frequency Cepstral Coefficients (MFCC) stream; calculating a Dynamic Time Warping (DTW) distance corresponding to the MFCC stream according to the MFCC stream and one of a plurality of sample streams generated by the Gaussian Mixture Model with Universal Background Model (GMM-UBM); calculating, according to the MFCC stream and the sample streams, a Log-likelihood ratio value corresponding to the MFCC stream as a GMM-UBM score; determining whether the sound signal stream passes a voice command authentication according to the GMM-UBM score, the DTW distance and the SNR value; in response to determining that the sound signal stream passes the voice command authentication, determining that the sound signal stream is a voice stream spoken from a legal user.Type: ApplicationFiled: September 21, 2019Publication date: March 25, 2021Applicant: Merry Electronics(Shenzhen) Co., Ltd.Inventors: Evelyn Kurniawati, Sasiraj Somarajan
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Patent number: 9633652Abstract: Embodiments reduce the complexity of speaker dependent speech recognition systems and methods by representing the code phrase (i.e., the word or words to be recognized) using a single Gaussian Mixture Model (GMM) which is adapted from a Universal Background Model (UBM). Only the parameters of the GMM need to be stored. Further reduction in computation is achieved by only checking the GMM component that is relevant to the keyword template. In this scheme, keyword template is represented by a sequence of the index of best performing component of the GMM of the keyword model. Only one template is saved by combining the registration template using Longest Common Sequence algorithm. The quality of the word model is continuously updated by performing expectation maximization iteration using the test word which is accepted as keyword model.Type: GrantFiled: March 31, 2013Date of Patent: April 25, 2017Assignee: STMicroelectronics Asia Pacific Pte Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Patent number: 9530417Abstract: Methods and systems of text independent speaker recognition provide a complexity comparable to text dependent speaker recognition system. These methods and systems exploit the fact that speech is a quasi-stationary signal and simplify the recognition process based on this theory. The speaker modeling allows a speaker profile to be updated progressively with new speech samples that are acquired during usage over time by the speaker.Type: GrantFiled: April 1, 2013Date of Patent: December 27, 2016Assignee: STMicroelectronics Asia Pacific Pte Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Publication number: 20140200890Abstract: Embodiments reduce the complexity of speaker dependent speech recognition systems and methods by representing the code word (i.e., the word to be recognized) using a single Gaussian Mixture Model (GMM) which is adapted from a Universal Background Model (UBM). Only the parameters of the GMM need to be stored. Further reduction in computation is achieved by only checking the GMM component that is relevant to the keyword template. In this scheme, keyword template is represented by a sequence of the index of best performing component of the GMM of the keyword model. Only one template is saved by combining the registration template using Longest Common Sequence algorithm. The quality of the word model is continuously updated by performing expectation maximization iteration using the test word which is accepted as keyword model.Type: ApplicationFiled: March 31, 2013Publication date: July 17, 2014Applicant: STMicroelectronics Asia Pacific Pte Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Patent number: 8489391Abstract: A system method of reusing information in a low power scalable hybrid audio encoder are disclosed. The includes determining a state of an advanced audio coding (AAC) transient flag, performing spectral band replication (SBR) transient detection on at least two possible locations upon a determination that the AAC transient flag is equal to a first value, performing SBR transient detection on a high frequency upon a determination that the AAC transient flag is equal to a second value, and determining whether a transient exists. The system includes a spectral band replication (SBR) coding module configured to determine a state of an advanced audio coding (AAC) transient flag and perform SBR transient detection on at least one location based upon an energy in a signal upon a determination that the AAC transient flag is equal to a first value.Type: GrantFiled: August 5, 2010Date of Patent: July 16, 2013Assignee: STMicroelectronics Asia Pacific Pte., Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Patent number: 8374857Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.Type: GrantFiled: August 3, 2007Date of Patent: February 12, 2013Assignee: STMicroelectronics Asia Pacific Pte, Ltd.Inventors: Evelyn Kurniawati, Kim Hann Kuah, Sapna George
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Patent number: 8332216Abstract: A method for stereo audio perceptual encoding of an input signal includes masking threshold estimation and bit allocation. The masking threshold estimation and bit allocation are performed once every two encoding processes. Another method for stereo audio perceptual encoding of an input signal includes performing a time-to-frequency transformation, performing a quantization, performing a bitstream formatting to produce an output stream, and performing a psychoacoustics analysis. The psychoacoustics analysis includes masking threshold estimation on a first of every two successive frames of the input signal.Type: GrantFiled: August 22, 2006Date of Patent: December 11, 2012Assignee: STMicroelectronics Asia Pacific PTE., Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Publication number: 20120163622Abstract: Methods and apparatuses for detection and reduction of wind noise in audio devices are disclosed. In an embodiment, a method includes acquiring and transforming the audio signals. Correlations from the transformed audio signals are computed. A cross correlation index is compared to a predetermined value to determine if a wind noise spectral content is present. In another embodiment, an apparatus includes an audio processing unit to receive non-decomposed audio signals, and an audio decomposition unit to receive the non-decomposed audio signals and to generate decomposed audio signals. A wind noise spectrum estimation unit receives non-decomposed audio signals and decomposed audio signals and identifies wind noise spectral components in at least one of the non-decomposed and decomposed audio signals. A wind noise spectrum reduction unit receives the wind noise spectral components and removes the wind noise spectral components from at least one of the non-decomposed and the decomposed audio signals.Type: ApplicationFiled: December 28, 2010Publication date: June 28, 2012Applicant: STMICROELECTRONICS ASIA PACIFIC PTE LTDInventors: Muralidhar KARTHIK, Samsudin, Evelyn KURNIAWATI, Sapna GEORGE
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Patent number: 8200351Abstract: A method and audio device are presented that preserve mono energy during downmixing of a hybrid coding process of an audio signal. The method includes calculating a stereo scaling factor in a group level that is definable within a stereo band. The method may also include updating the stereo scaling factor using an update rate and synchronizing the update rate of a spatial parameter during a fast changing transient portion of the signal. A number of groups in a first stereo band may be greater than a number of groups in a second stereo band, and the first stereo band may be a lower frequency band than the second band or may be perceptually more important than the second band.Type: GrantFiled: December 28, 2007Date of Patent: June 12, 2012Assignee: STMicroelectronics Asia PTE., Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Publication number: 20120035936Abstract: A system and method of reusing information in low power scalable hybrid audio encoders. The system and method provides a transform coder and parameterization of high frequency spectrum (SBR).Type: ApplicationFiled: August 5, 2010Publication date: February 9, 2012Applicant: STMICROELECTRONICS ASIA PACIFIC PTE LTDInventors: Evelyn Kurniawati, Sapna George
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Patent number: 7873510Abstract: A system and method for adaptive rate control in audio processing is provided. The process could include receiving uncompressed audio data from an input and generating MDCT spectrum for each frame of the uncompressed audio data using a filterbank. The process could also include estimating masking thresholds for current frame to be encoded based on the MDCT spectrum. The masking thresholds reflect a bit budget for the current frame. The process could also include performing quantization of the current frame based on the masking thresholds. After the quantization of the current frame, the bit budget for next frame is updated for estimating the masking thresholds of the next frame. The process could also include encoding the quantized audio data.Type: GrantFiled: April 26, 2007Date of Patent: January 18, 2011Assignee: STMicroelectronics Asia Pacific Pte. Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Publication number: 20080199014Abstract: A method and audio device are presented that preserve mono energy during downmixing of a hybrid coding process of an audio signal. The method includes calculating a stereo scaling factor in a group level that is definable within a stereo band. The method may also include updating the stereo scaling factor using an update rate and synchronizing the update rate of a spatial parameter during a fast changing transient portion of the signal. A number of groups in a first stereo band may be greater than a number of groups in a second stereo band, and the first stereo band may be a lower frequency band than the second band or may be perceptually more important than the second band.Type: ApplicationFiled: December 28, 2007Publication date: August 21, 2008Applicant: STMicroelectronics Asia Pacific PTE LtdInventors: Evelyn Kurniawati, Sapna George
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Publication number: 20080040120Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.Type: ApplicationFiled: August 3, 2007Publication date: February 14, 2008Applicant: STMicroelectronics Asia Pacific Pte., Ltd.Inventors: Evelyn Kurniawati, Kim Kuah, Sapna George
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Publication number: 20070255562Abstract: A system and method for adaptive rate control in audio processing is provided. The process could include receiving uncompressed audio data from an input and generating MDCT spectrum for each frame of the uncompressed audio data using a filterbank. The process could also include estimating masking thresholds for current frame to be encoded based on the MDCT spectrum. The masking thresholds reflect a bit budget for the current frame. The process could also include performing quantization of the current frame based on the masking thresholds. After the quantization of the current frame, the bit budget for next frame is updated for estimating the masking thresholds of the next frame. The process could also include encoding the quantized audio data.Type: ApplicationFiled: April 26, 2007Publication date: November 1, 2007Applicant: STMicroelectronics Asia Pacific Pte., Ltd.Inventors: Evelyn Kurniawati, Sapna George
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Publication number: 20070162277Abstract: A method for stereo audio perceptual encoding of an input signal includes masking threshold estimation and bit allocation. The masking threshold estimation and bit allocation are performed once every two encoding processes. Another method for stereo audio perceptual encoding of an input signal includes performing a time-to-frequency transformation, performing a quantization, performing a bitstream formatting to produce an output stream, and performing a psychoacoustics analysis. The psychoacoustics analysis includes masking threshold estimation on a first of every two successive frames of the input signal.Type: ApplicationFiled: August 22, 2006Publication date: July 12, 2007Applicant: STMicroelectronics Asia Pacific Pte., Ltd.Inventors: Evelyn Kurniawati, Sapna George