Qualcomm Incorporated

Other

Andrew P Dejaco

Arasanipalai K Ananthapadmanabhan

Paul Eric Jacobs

Khaled Helmi El-Maleh

Sharath Manjunath

Ning Bi

William Robert Gardner

Pengjun Huang

Eddie-Lun Tik Choy

Chienchung Chang

S Katherine Lam

Roberto Padovani

Harinath Garudadri

Noam A Ziv

Ananthapadmanabhan Arasanipalai Kandhadai

Gilbert Christopher Sih

Samir Kumar Gupta

Anthony P Mauro

Charles Seunghwan Han

Eddie Lun Tik Choy

Serge D Willenegger

Lindsay A Weaver, Jr

Yingyong Qi

Franklin Peter Antonio

Jason Borthwick Kenagy

Suhail Jalil

John A Miller

Scott David Beith

Narendranath Malayath

David Puig Oses

William Yee-Ming Huang

Sean S English

Karthick Chinnaswami

Richard Paul Walters

George Gaozhi Pan

Robert Opalsky

Sarath Manjunath

Ananth Ananthpadmanabhan

Scott M Woodward

Arasanipali K Ananthapadmanabhan

A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.

Method and apparatus for predictively quantizing voiced speech

A method and apparatus are described for controlling the data rates for communications to and from a base station and a plurality of remote users. The usage of the communications resource whether the forward link resource (from base station to remote users) or reverse link resource (from remote users to base station) is measured. The measured usage value is compared against at least one predetermined threshold value and the data rates of communications or a subset of communications on said communications resource is modified in accordance with said comparisons.

Method and apparatus for determining the transmission data rate in a multi-user communication system

A speech signal is decoded by a vocoder and the reconstructed speech samples are provided to a decoded frame check unit. The decoded frame check unit examines the energy of the reconstructed speech and compares the energy of the reconstructed speech to a range of acceptable energy values. If the energy is not within the range of energy values, a frame erasure is declared and the decoded frame is prevented from being to the speaker in the telephone. In the exemplary implementation, the speech is reconstructed by a vocoder which includes a postfilter which in turn includes automatic gain control. The automatic gain control element of a post filter includes a means for measuring the energy of the decoded speech data. This measured energy is used by the decoded frame check unit to decide whether to provide the decoded data to the user or to declare a frame erasure. This implementation reduces the amount of additional hardware necessary to implement the present invention.

Method and apparatus for detecting bad data packets received by a mobile telephone using decoded speech parameters

A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

Method and apparatus for improved detection of rate errors in variable rate receivers

Wideband speech signals must be converted to narrowband speech signals if the transmission medium or the destination terminal is constructed with narrowband constraints. A typical wideband-to-narrowband conversion method is the elimination of frequencies above 3400 Hz using a low pass filter and a down sampler. However, this method produces a muffled speech sound since the resulting narrowband signal has a flat frequency response. Methods and apparatus are presented herein to enhance the acoustic quality of a wideband-to-narrowband converted signal. A bandwidth switching filter is used to emphasize a mid-range frequency portion of the wideband signal so that the resulting narrowband signal has a non-flat frequency spectrum.

Enhanced conversion of wideband signals to narrowband signals

According to a disclosed embodiment, an incoming call from a first telephone number is received (). A variable indicating the number of calls received from the first telephone number is incremented (). The variable is compared to a threshold variable (). Subsequently, the user is prompted to save the first telephone number in an address book () when the variable indicating the number of times an incoming call is received from the first telephone number is equal to, or greater than, the threshold variable. The first telephone number is then stored in the address book. Moreover, an apparatus can be assembled () to provide cellular telephone users with, for example, instructions and information via a speaker () and an LCD display (). The apparatus is configured to provide instructions and information to the cellular telephone user at appropriate times, thereby increasing the cellular telephone's usefulness.

Method and apparatus for prompting a cellular telephone user with instructions

Method and apparatus for quantizing pitch, amplitude, phase and linear spectrum of voiced speech

A call setup procedure is presented to permit vocoder bypass, which will allow the transmission of wideband speech packets between wideband terminals over narrowband transmission constraints. In addition, methods and apparatus are presented that allow the conversion between a wideband tandem-free operation, a narrowband tandem-free operation, and a standard tandem operation.

Communications using wideband terminals

The disclosed embodiments provide a method and apparatus for interoperability between CTX and DTX communications systems during transmissions of silence or background noise [FIG.]. Continuous eighth rate encoded noise frames are translated to discontinuous SID frames for transmission to DTX systems (). Discontinuous SID frames are translated to continuous eighth rate encoded noise frames for decoding by a CTX system (). Applications of CTX to DTX interoperability comprise CDMA and GSM interoperability (narrowband voice transmission systems), CDMA next generation vocoder (The Selectable Mode Vocoder) interoperability with the new ITU-T 4 kbps vocoder operating in DTX-mode for Voice Over IP applications, future voice transmission systems that have a common speech encoder/decoder but operate in differing CTX or DTX modes during speech non-activity, and CDMA wideband voice transmission system interoperability with other wideband voice transmission systems with common wideband vocoders but with different modes of operation (DTX or CTX) during voice non-activity.

Method and apparatus for interoperability between voice transmission systems during speech inactivity

A speech processing system modifies various aspects of input speech according to a user-selected one of various preprogrammed voice fonts. Initially, the speech converter receives a formants signal representing an input speech signal and a pitch signal representing the input signal's fundamental frequency. One or both of the following may also be received: a voicing signal comprising an indication of whether the input speech signal is voiced, unvoiced, or mixed, and/or a gain signal representing the input speech signal's energy. The speech converter also receives user selection of one of multiple preprogrammed voice fonts, each specifying a manner of modifying one or more of the received signals (i.e., formants, voicing, pitch, gain). The speech converter modifies at least one of the formants, voicing, pitch, and/or gain signals as specified by the selected voice font.

Speech converter utilizing preprogrammed voice profiles

A method and system for speech recognition combines different types of engines in order to recognize user-defined digits and control words, predefined digits and control words, and nametags. Speaker-independent engines are combined with speaker-dependent engines. A Hidden Markov Model (HMM) engine is combined with Dynamic Time Warping (DTW) engines.

System and method for hybrid voice recognition

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San Diego, CA, United States of America

Andrew P Dejaco