Song Wang

US Patent:

7508327, Mar 24, 2009

Filed:

Nov 9, 2006

Appl. No.:

11/558313

Inventors:

Song Wang - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US
Samir Kumar Gupta - San Diego CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

H03M 7/00

US Classification:

341 61, 314122

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.

US Patent:

7528745, May 5, 2009

Filed:

Jun 13, 2006

Appl. No.:

11/452836

Inventors:

Song Wang - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US
Prajakt V. Kulkarni - San Diego CA, US
Samir Kumar Gupta - San Diego CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

H03M 7/00

US Classification:

341 61, 708270, 708290, 708300, 708313

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. A sampling rate converter up-samples the digital signal at an input sampling frequency to the selected intermediate sampling frequency, filters the digital signal with the derived anti-aliasing filter, and down-samples the digital signal by the selected down-sampling factor to the desired output sampling frequency.

US Patent:

7742746, Jun 22, 2010

Filed:

Apr 30, 2007

Appl. No.:

11/742476

Inventors:

Pei Xiang - San Diego CA, US
Song Wang - San Diego CA, US
Prajakt V. Kulkarni - San Diego CA, US
Samir Kumar Gupta - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

H04B 1/00

US Classification:

455 631, 455501, 455 6313, 455570, 4551142, 381 717, 381 943, 704203

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.

US Patent:

7970564, Jun 28, 2011

Filed:

Oct 20, 2006

Appl. No.:

11/551509

Inventors:

Song Wang - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US
Samir Kumar Gupta - San Diego CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

G01R 13/00

US Classification:

702 66, 702 70, 702 74, 702189, 702190, 702191, 702192, 702193, 702194, 702195, 704200, 704226, 704228, 704233

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.

US Patent:

8175871, May 8, 2012

Filed:

Sep 28, 2007

Appl. No.:

11/864906

Inventors:

Song Wang - San Diego CA, US
Samir Kumar Gupta - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

G10L 21/02
G10L 19/00
G10L 17/00
H04B 15/00
H04M 9/08

US Classification:

704227, 704201, 704226, 704246, 702191, 702195, 702197, 37940601, 37940603

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.

US Patent:

8184816, May 22, 2012

Filed:

Nov 25, 2008

Appl. No.:

12/323186

Inventors:

Dinesh Ramakrishnan - San Diego CA, US
Song Wang - San Diego CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

H04R 3/00

US Classification:

381 56, 381 92

Abstract:

A method for detecting wind noise is described. At least two audio signals are received. The at least two audio signals are filtered to reduce higher frequencies and to reduce lower frequencies to provide at least two filtered audio signals. The cross correlation of the at least two filtered audio signals is computed for multiple delays. A maximum cross correlation is determined from the cross correlations computed for the multiple delays. Wind noise is detected by comparing the maximum cross correlation with a threshold.

US Patent:

8217812, Jul 10, 2012

Filed:

May 5, 2010

Appl. No.:

12/774398

Inventors:

Song Wang - San Diego CA, US
Aris Balatsos - San Diego CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

H03M 7/00

US Classification:

341 61, 341123, 341155, 375316, 375353, 375371, 375372, 370313, 370543

Abstract:

Techniques of this disclosure provide for adjustment of a conversion rate of a sampling rate converter (SRC) in real-time. The SRC determines relative timing of generated output samples based on non-approximated integer components that are recursively updated. The SRC may further base relative timing of output samples on a value of one or more step size components associated with the integer components. Also according to techniques of this disclosure, a conversion rate of an SRC may be adjusted in real-time based on a detected mismatch between a source clock of a digital input signal and a local clock.

US Patent:

8223988, Jul 17, 2012

Filed:

Jan 29, 2008

Appl. No.:

12/022037

Inventors:

Song Wang - San Diego CA, US
Dinesh Ramakrishnan - San Diego CA, US
Samir Gupta - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

H04R 3/00
H04R 1/02
H04R 9/06

US Classification:

381 92, 381334, 381111, 381122

Abstract:

An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages.