Samir N Gupta

US Patent:

6606382, Aug 12, 2003

Filed:

Jan 27, 2000

Appl. No.:

09/491933

Inventors:

Samir K. Gupta - San Diego CA

Assignee:

Qualcomm Incorporated - San Diego CA

International Classification:

H04M 100

US Classification:

37940605, 37940607

Abstract:

A system and method for canceling an echo signal. An input waveform is provided to an acoustic processor, and a determination is made whether the input waveform includes information representative of an echo signal. If the input waveform includes information representative of an echo signal, an output waveform is formed by attenuating a residual waveform with the acoustic processor. The residual waveform is attenuated by an attenuation factor that gradually changes from an initial attenuation value to a final attenuation value during the attenuation step. A system and method for adjusting an acoustic signal from a muted state to an unmuted state by varying an attenuation factor applied to an acoustic signal by an acoustic processor. The acoustic signal is provided to an acoustic processor. An output waveform is formed from the acoustic processor by adjusting the attenuation factor from a muted state to a first attenuation value associated with the non-muted state.

US Patent:

6744882, Jun 1, 2004

Filed:

Mar 30, 1999

Appl. No.:

09/281564

Inventors:

Samir Gupta - San Diego CA
Anthony P. Mauro - San Diego CA
Andrew P. Dejaco - San Diego CA

Assignee:

Qualcomm Inc. - San Diego CA

International Classification:

H04M 100

US Classification:

37938701, 37938803, 37939003, 381104, 381 57, 4552321, 4552341

Abstract:

The mobile telephone is provided with the capability for automatically adjusting the gain of a microphone of the telephone based upon the detected noise level in which the cellular telephone is operated. As the noise level increases, the gain of the microphone is automatically decreased, thereby compensating for the natural tendency of telephone users to speak more loudly in noisy environments. Also, by decreasing the microphone gain, any clipping that might otherwise occur as a result of the user speaking more loudly is avoided and the signal-to-noise ratio is not thereby decreased. Furthermore, because the microphone gain decreases, the volume level of the voice of the user as it is output from the other partys telephone is not unduly loud. Hence, the other party need not manually decrease the speaker gain of his or her telephone. In the exemplary embodiment, the cellular telephone includes a digital signal processor configured or programmed to apply the detected noise level to look-up tables relating various noise levels to appropriate speaker and microphone gain levels.

US Patent:

6766176, Jul 20, 2004

Filed:

Mar 30, 1999

Appl. No.:

09/281568

Inventors:

Samir Gupta - San Diego CA
Anthony P. Mauro - San Diego CA
Andrew P. Dejaco - San Diego CA

Assignee:

Qualcomm Incorporated - San Diego CA

International Classification:

H04M 100

US Classification:

4555501, 4554141, 455 6711, 455219

Abstract:

The mobile telephone is provided with the capability for automatically adjusting the gain of a microphone of the telephone based upon the detected noise level in which the cellular telephone is operated. As the noise level increases, the gain of the microphone is automatically decreased, thereby compensating for the natural tendency of telephone users to speak more loudly in noisy environments. Also, by decreasing the microphone gain, any clipping that might otherwise occur as a result of the user speaking more loudly is avoided and the signal-to-noise ratio is not thereby decreased. Furthermore, because the microphone gain decreases, the volume level of the voice of the user as it is output from the other partys telephone is not unduly loud. Hence, the other party need not manually decrease the speaker gain of his or her telephone. In the exemplary embodiment, the cellular telephone includes a digital signal processor configured or programmed to apply the detected noise level to look-up tables relating various noise levels to appropriate speaker and microphone gain levels.

US Patent:

6889066, May 3, 2005

Filed:

Mar 27, 2001

Appl. No.:

09/819109

Inventors:

Samir Gupta - San Diego CA, US
Ivan K. Oei - San Diego CA, US

Assignee:

Qualcomm Incorporated - San Diego CA

International Classification:

H04M001/00

US Classification:

455570, 3794061, 37940607

Abstract:

In one embodiment, a mobile station such as a dual-mode, AMPS and CDMA, handset communicates with an AMPS base station in a wireless communication system. The mobile station includes an acoustic echo canceller, which can detect acoustic echo, for example, feedback from the handset speaker to the handset microphone, during communication between the mobile station and the base station, and cancel or eliminate the acoustic echo. The mobile station also includes a network echo suppressor coupled to the acoustic echo canceller. For example, the network echo suppressor can read a parameter set by the acoustic echo canceller. The network echo suppressor attenuates or mutes a network echo caused by portions of the communication returning to the mobile station from the base station after a network propagation delay. Coupling the network echo suppressor to the acoustic echo canceller ensures proper interaction between the network echo suppressor and acoustic echo canceller.

US Patent:

6892177, May 10, 2005

Filed:

Feb 26, 2001

Appl. No.:

09/793337

Inventors:

Louis Dominic Oliveira - San Diego CA, US
Samir Kamar Gupta - San Diego CA, US
Seyfollah Bazarjani - San Diego CA, US
Wenjun Su - San Diego CA, US

Assignee:

Qualcomm Incorporated - San Diego CA

International Classification:

G10L021/00

US Classification:

704500, 341139, 341141, 341144

Abstract:

A wireless communication device () may include a receiver (), a memory (), a digital-to-analog converter (), an audio playback system () and other features. A dynamic range controller () selectively generates control signals to adjust, at least in part, the operational dynamic range of the digital-to-analog converter () for digital signals received by the receiver () or stored in the memory (). The selection of dynamic range is based on identifying a characteristic. In one embodiment, the control signals are used to selectively operate the digital-to-analog converter () at a particular dynamic range based on a sampling rate of a received digital audio signal.

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:

7663046, Feb 16, 2010

Filed:

Mar 4, 2008

Appl. No.:

12/042170

Inventors:

Prajakt Kulkarni - San Diego CA, US
Eddie L. T. Choy - Carlsbad CA, US
Nidish Ramachandra Kamath - Placentia CA, US
Samir K Gupta - San Diego CA, US
Stephen Molloy - San Diego CA, US
Suresh Devalapalli - San Diego CA, US

Assignee:

QUALCOMM Incorporated - San Diego CA

International Classification:

G10H 1/00

US Classification:

84600, 84626, 84645

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.