Patents by Inventor Eddie L. T. Choy

Eddie L. T. Choy 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).

  • Publication number: 20090136044
    Abstract: In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.
    Type: Application
    Filed: November 28, 2007
    Publication date: May 28, 2009
    Applicant: QUALCOMM INCORPORATED
    Inventors: Pei Xiang, Samir Kumar Gupta, Eddie L. T. Choy, Prajakt V. Kulkarni
  • Patent number: 7528745
    Abstract: Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter.
    Type: Grant
    Filed: June 13, 2006
    Date of Patent: May 5, 2009
    Assignee: QUALCOMM Incorporated
    Inventors: Song Wang, Eddie L. T. Choy, Prajakt V. Kulkarni, Samir Kumar Gupta
  • Publication number: 20090089054
    Abstract: Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.
    Type: Application
    Filed: September 28, 2007
    Publication date: April 2, 2009
    Applicant: QUALCOMM INCORPORATED
    Inventors: Song Wang, Samir Kumar Gupta, Eddie L. T. Choy
  • Publication number: 20090089053
    Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.
    Type: Application
    Filed: September 28, 2007
    Publication date: April 2, 2009
    Applicant: QUALCOMM INCORPORATED
    Inventors: Song Wang, Samir Kumar Gupta, Eddie L. T. Choy
  • Patent number: 7508327
    Abstract: In general, this disclosure describes techniques for changing a sampling frequency of a digital signal. In particular, the techniques provide a more accurate way to determining a relative timing between a desired output sample and a corresponding input sample using a non-approximated integer representation of the relative timing. The relative timing between the desired output sample and corresponding input sample may be represented using a first component that identifies a latest input sample of the digital signal used to generate intermediate samples, a second component that identifies an intermediate sample, and a third component that identifies a timing difference between the desired output sample and the intermediate sample. Each of the components may be recursively updated using non-approximated integer values.
    Type: Grant
    Filed: November 9, 2006
    Date of Patent: March 24, 2009
    Assignee: QUALCOMM Incorporated
    Inventors: Song Wang, Eddie L. T. Choy, Samir Kumar Gupta
  • Publication number: 20090070119
    Abstract: Power savings in a mobile device is accomplished by generating audio samples by decoding a bitstream with a decoding system within the mobile device. The generated audio samples are transferred into at least one memory bank in a set of memory banks in a power saver block within the mobile device. Parts of the decoding system not involved in the storing of the generated audio samples are switched off after batch decoding a bitstream associated with multiple audio frames. The bitstream includes bits less than that found in one audio file. At least one of the memory banks in the set of memory banks is power collapsible. The fetching of the decoded by the decoding system can be synchronized with a paging channel of a modem in the mobile device. The transferred audio samples is a lossless compression and may occur after a re-encoding.
    Type: Application
    Filed: September 4, 2008
    Publication date: March 12, 2009
    Applicant: QUALCOMM Incorporated
    Inventors: Heejong Yoo, Nidish R. Kamath, Eddie L.T. Choy, Johnny Kallacheril John, Samir K. Gupta
  • Publication number: 20090024397
    Abstract: A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules.
    Type: Application
    Filed: July 16, 2008
    Publication date: January 22, 2009
    Applicant: QUALCOMM Incorporated
    Inventors: Sang-Uk Ryu, Eddie L.T. Choy, Nidish Ramachandra Kamath, Samir Kumar Gupta, Suresh Devalapalli
  • Publication number: 20080269926
    Abstract: A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.
    Type: Application
    Filed: April 30, 2007
    Publication date: October 30, 2008
    Inventors: Pei Xiang, Song Wang, Prajakt V. Kulkarni, Samir Kumar Gupta, Eddie L.T. Choy
  • Publication number: 20080229918
    Abstract: This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, various tasks associated with MIDI file processing are delegated between software operating on a general purpose processor, firmware associated with a digital signal processor (DSP), and dedicated hardware that is specifically designed for MIDI file processing. Alternatively, a multi-threaded DSP may be used instead of a general purpose processor and the DSP. In one aspect, this disclosure provides a method comprising parsing MIDI files and scheduling MIDI events associated with the MIDI files using a first process, processing the MIDI events using a second process to generate MIDI synthesis parameters, and generating audio samples using a hardware unit based on the synthesis parameters.
    Type: Application
    Filed: March 4, 2008
    Publication date: September 25, 2008
    Applicant: QUALCOMM Incorporated
    Inventors: Prajakt Kulkarni, Eddie L. T. Choy, Nidish Ramachandra Kamath, Samir Kumar Gupta, Stephen Molloy, Suresh Devalapalli
  • Publication number: 20080229919
    Abstract: This disclosure describes techniques that make use of a plurality of hardware elements that operate simultaneously to service synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises storing audio synthesis parameters generated for one or more audio files of an audio frame, processing a first audio synthesis parameter using a first audio processing element of a hardware unit to generate first audio information, processing a second audio synthesis parameter using a second audio processing element of the hardware unit to generate second audio information, and generating audio samples for the audio frame based at least in part on a combination of the first and second audio information.
    Type: Application
    Filed: March 4, 2008
    Publication date: September 25, 2008
    Applicant: QUALCOMM Incorporated
    Inventors: Nidish Ramachandra Kamath, Eddie L.T. Choy, Prajakt Kulkarni, Samir Kumar Gupta, Stephen Molloy, Suresh Devalapalli
  • Publication number: 20080074542
    Abstract: A method and system for resynchronizing an embedded multimedia system using bytes consumed in an audio decoder. The bytes consumed provides a mechanism to compensate for bit error handling and correction in a system that does not require re-transmission. The audio decoder keeps track of the bytes consumed and periodically reports the bytes consumed. A host microprocessor indexes the actual bytes consumed since bit errors may have been handled or corrected to a predetermined byte count to determine whether resynchronization is necessary.
    Type: Application
    Filed: September 26, 2006
    Publication date: March 27, 2008
    Inventors: Mingxia Cheng, Anthony Patrick Mauro, Eddie L.T. Choy, Yujie Gao, Kuntal Dilipsinh Sampat, Matthew Blaine Zivney, Satish Goverdhan, Samir Kumar Gupta, Harinath Garudadri
  • Publication number: 20070290900
    Abstract: In general, this disclosure describes techniques for changing a sampling frequency of a digital signal. In particular, the techniques provide a more accurate way to determining a relative timing between a desired output sample and a corresponding input sample using a non-approximated integer representation of the relative timing. The relative timing between the desired output sample and corresponding input sample may be represented using a first component that identifies a latest input sample of the digital signal used to generate intermediate samples, a second component that identifies an intermediate sample, and a third component that identifies a timing difference between the desired output sample and the intermediate sample. Each of the components may be recursively updated using non-approximated integer values.
    Type: Application
    Filed: November 9, 2006
    Publication date: December 20, 2007
    Inventors: Song Wang, Eddie L.T. Choy, Samir Kumar Gupta
  • Publication number: 20070286426
    Abstract: This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.
    Type: Application
    Filed: June 7, 2006
    Publication date: December 13, 2007
    Inventors: Pei Xiang, Eddie L.T. Choy, Prajakt V. Kulkarni, Samir Kumar Gupta
  • Publication number: 20070257840
    Abstract: This disclosure describes signal processing techniques that can improve the performance of blind source separation (BSS) techniques. In particular, the described techniques propose pre-processing steps that can help to de-correlate the different signals from one another prior to execution of the BSS techniques. In addition, the described techniques also propose optional post-processing steps that can further de-correlate the different signals following execution of the BSS techniques. The techniques may be particularly useful for improving BSS performance with highly correlated audio signals, e.g., from two microphones that are in close spatial proximity to one another.
    Type: Application
    Filed: October 20, 2006
    Publication date: November 8, 2007
    Inventors: Song Wang, Eddie L.T. Choy, Samir Kumar Gupta
  • Publication number: 20070192390
    Abstract: Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter.
    Type: Application
    Filed: June 13, 2006
    Publication date: August 16, 2007
    Inventors: Song Wang, Eddie L.T. Choy, Prajakt V. Kulkarni, Samir Kumar Gupta
  • Patent number: 6754630
    Abstract: In a method of synthesizing voiced speech from pitch prototype waveforms by time-synchronous waveform interpolation (TSWI), one or more pitch prototypes is extracted from a speech signal or a residue signal. The extraction process is performed in such a way that the prototype has minimum energy at the boundary. Each prototype is circularly shifted so as to be time-synchronous with the original signal. A linear phase shift is applied to each extracted prototype relative to the previously extracted prototype so as to maximize the cross-correlation between successive extracted prototypes. A two-dimensional prototype-evolving surface is constructed by unsampling the prototypes to every sample point. The two-dimensional prototype-evolving surface is re-sampled to generate a one-dimensional, synthesized signal frame with sample points defined by piecewise continuous cubic phase contour functions computed from the pitch lags and the phase shifts added to the extracted prototypes.
    Type: Grant
    Filed: November 13, 1998
    Date of Patent: June 22, 2004
    Assignee: Qualcomm, Inc.
    Inventors: Amitava Das, Eddie L. T. Choy
  • Publication number: 20010051873
    Abstract: In a method of synthesizing voiced speech from pitch prototype waveforms by time-synchronous waveform interpolation (TSWI), one or more pitch prototypes is extracted from a speech signal or a residue signal. The extraction process is performed in such a way that the prototype has minimum energy at the boundary. Each prototype is circularly shifted so as to be time-synchronous with the original signal. A linear phase shift is applied to each extracted prototype relative to the previously extracted prototype so as to maximize the cross-correlation between successive extracted prototypes. A two-dimensional prototype-evolving surface is constructed by unsampling the prototypes to every sample point. The two-dimensional prototype-evolving surface is re-sampled to generate a one-dimensional, synthesized signal frame with sample points defined by piecewise continuous cubic phase contour functions computed from the pitch lags and the phase shifts added to the extracted prototypes.
    Type: Application
    Filed: November 13, 1998
    Publication date: December 13, 2001
    Inventors: AMITAVA DAS, EDDIE L. T. CHOY