Patents by Inventor Jes Thyssen

Jes Thyssen 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).

  • Patent number: 8032363
    Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.
    Type: Grant
    Filed: August 9, 2002
    Date of Patent: October 4, 2011
    Assignee: Broadcom Corporation
    Inventors: Juin-Hwey Chen, Jes Thyssen, Chris C Lee
  • Patent number: 8024192
    Abstract: A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain.
    Type: Grant
    Filed: August 15, 2007
    Date of Patent: September 20, 2011
    Assignee: Broadcom Corporation
    Inventors: Robert W. Zopf, Juin-Hwey Chen, Jes Thyssen
  • Patent number: 8015000
    Abstract: An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.
    Type: Grant
    Filed: April 13, 2007
    Date of Patent: September 6, 2011
    Assignee: Broadcom Corporation
    Inventors: Robert W. Zopf, Juin-Hwey Chen, Jes Thyssen
  • Patent number: 8005678
    Abstract: A technique is described herein for updating a state of a decoder configured to decode a series of frames representing an encoded audio signal. In accordance with the technique, an output audio signal associated with a lost frame in the series of frames is synthesized. The decoder state is set to align with the synthesized output audio signal at a frame boundary. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with a first received frame after the lost frame in the series of frames, wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoder state is then reset based on the time lag.
    Type: Grant
    Filed: August 15, 2007
    Date of Patent: August 23, 2011
    Assignee: Broadcom Corporation
    Inventors: Robert W. Zopf, Jes Thyssen, Juin-Hwey Chen
  • Patent number: 8000960
    Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.
    Type: Grant
    Filed: August 15, 2007
    Date of Patent: August 16, 2011
    Assignee: Broadcom Corporation
    Inventors: Juin-Hwey Chen, Robert W. Zopf, Jes Thyssen
  • Patent number: 7987095
    Abstract: Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal.
    Type: Grant
    Filed: February 7, 2007
    Date of Patent: July 26, 2011
    Assignee: Broadcom Corporation
    Inventors: Wilfrid LeBlanc, Jes Thyssen
  • Publication number: 20110096942
    Abstract: Systems and methods are described for applying noise suppression to one or more audio signals to generate a noise-suppressed audio signal therefrom. In a single-channel implementation, an input signal is received that comprises a desired audio signal and an additive noise signal. Noise suppression is then applied to the input signal to generate a noise-suppressed signal in a manner that is controlled by at least a parameter that specifies a degree of balance between distortion of the desired audio signal and unnaturalness of a residual noise signal included in the noise-suppressed signal. In an alternative single-channel implementation, a plurality of sub-band signals obtained by applying a frequency conversion process to a time domain representation of an input signal is received. Noise suppression is then applied to each of the sub-band signals by passing each of the sub-band signals through a time direction filter. Multi-channel noise suppression variants are also described.
    Type: Application
    Filed: October 4, 2010
    Publication date: April 28, 2011
    Applicant: BROADCOM CORPORATION
    Inventor: Jes Thyssen
  • Publication number: 20110095875
    Abstract: Systems and methods are described that automatically adjust a value of a parameter relating to the delivery of media content, such as audio content or image content, based on both environmental conditions and on automatically-learned user preference data. For example, a first embodiment adjusts a volume setting used to control the delivery of an audio signal based both on environmental noise conditions and upon automatically-learned user preference information, wherein the user preference information is derived by monitoring user-implemented adjustments to the volume setting after application of an automatic adjustment thereto.
    Type: Application
    Filed: March 31, 2010
    Publication date: April 28, 2011
    Applicant: BROADCOM CORPORATION
    Inventors: Jes Thyssen, Wilfrid LeBlanc
  • Publication number: 20110029317
    Abstract: Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.
    Type: Application
    Filed: July 30, 2010
    Publication date: February 3, 2011
    Applicant: BROADCOM CORPORATION
    Inventors: Juin-Hwey Chen, Hong-goo Kang, Robert W. Zopf, Jes Thyssen
  • Publication number: 20100223054
    Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.
    Type: Application
    Filed: May 14, 2010
    Publication date: September 2, 2010
    Applicant: BROADCOM CORPORATION
    Inventors: Elias Nemer, Wilfrid LeBlanc, Syavosh Zad-Issa, Jes Thyssen
  • Publication number: 20100217590
    Abstract: A system and method for performing speaker localization is described. The system and method utilizes speaker recognition to provide an estimate of the direction of arrival (DOA) of speech sound waves emanating from a desired speaker with respect to a microphone array included in the system. Candidate DOA estimates may be preselected or generated by one or more other DOA estimation techniques. The system and method is suited to support steerable beamforming as well as other applications that utilize or benefit from DOA estimation. The system and method provides robust performance even in systems and devices having small microphone arrays and thus may advantageously be implemented to steer a beamformer in a cellular telephone or other mobile telephony terminal featuring a speakerphone mode.
    Type: Application
    Filed: February 24, 2009
    Publication date: August 26, 2010
    Applicant: BROADCOM CORPORATION
    Inventors: Elias Nemer, Jes Thyssen
  • Patent number: 7684521
    Abstract: Typical communication systems operate with a single channel decoder, and hence would have to settle for the performance from the single channel decoder regardless of the conditions of the communications channel. The present invention uses a hybrid channel decoder comprising multiple channel decoders, each configured to optimize the quality of the re-constructed signal for different channel conditions. Therefore, the desired decoder can be selected as conditions of the communications channel, or the data signal, change over time, so as to optimize the re-constructed data signal. In embodiments, the data signal is a speech signal.
    Type: Grant
    Filed: February 3, 2005
    Date of Patent: March 23, 2010
    Assignee: Broadcom Corporation
    Inventors: Jes Thyssen, Juin-Hwey Chen, Nambi Seshadri
  • Publication number: 20100020986
    Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.
    Type: Application
    Filed: October 30, 2008
    Publication date: January 28, 2010
    Applicant: BROADCOM CORPORATION
    Inventors: Elias Nemer, Wilfrid LeBlanc, Mohammad Zad-Issa, Jes Thyssen
  • Patent number: 7647223
    Abstract: A sub-quantizer for sub-quantization of a vector includes a sub-codevector generator that generates a set of candidate sub-codevectors, and transformation logic that transforms each candidate sub-codevector into a corresponding codevector. A memory stores an illegal space definition representing illegal vectors. A legal status tester determines legal codevectors among the codevectors based on the illegal space definition. An error calculator generates error terms corresponding to the candidate sub-codevector, and a sub-codevector selector determines a best one of the sub-codevectors corresponding to a legal codevector and a best error term. The vector includes parameters relating to a speech and/or audio signal, such as Line Spectral Frequencies (LSFs).
    Type: Grant
    Filed: June 7, 2002
    Date of Patent: January 12, 2010
    Assignee: Broadcom Corporation
    Inventor: Jes Thyssen
  • Publication number: 20090287496
    Abstract: A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs.
    Type: Application
    Filed: July 28, 2009
    Publication date: November 19, 2009
    Applicant: BROADCOM CORPORATION
    Inventors: Jes Thyssen, Wilfrid LeBlanc, Juin-Hwey Chen
  • Publication number: 20090281803
    Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.
    Type: Application
    Filed: May 12, 2009
    Publication date: November 12, 2009
    Applicant: Broadcom Corporation
    Inventors: Juin-Hwey Chen, Jes Thyssen
  • Publication number: 20090281802
    Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.
    Type: Application
    Filed: May 12, 2009
    Publication date: November 12, 2009
    Applicant: BROADCOM CORPORATION
    Inventors: Jes Thyssen, Juin-Hwey Chen, Wilfrid LeBlanc
  • Publication number: 20090281805
    Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.
    Type: Application
    Filed: May 12, 2009
    Publication date: November 12, 2009
    Applicant: Broadcom Corporation
    Inventors: Wilfrid LeBlanc, Jes Thyssen, Juin-Hwey Chen
  • Publication number: 20090281801
    Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.
    Type: Application
    Filed: May 12, 2009
    Publication date: November 12, 2009
    Applicant: BROADCOM CORPORATION
    Inventors: Jes Thyssen, Wilfrid LeBlanc, Juin-Hwey Chen
  • Publication number: 20090281800
    Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.
    Type: Application
    Filed: May 12, 2009
    Publication date: November 12, 2009
    Applicant: BROADCOM CORPORATION
    Inventors: Wilfrid LeBlanc, Juin-Hwey Chen, Jes Thyssen