Inventor
Huan-Yu Su
Inventor
San Clemente, CA, US
Overview
Patents· 74 as inventor
Public. Assignee & technology derived from the patent record. Each patent links to its LexDana page.
G10Lprimary
G10LSpeech analysis techniques or speech synthesis61%
H03GControl of amplification12%
H04LTransmission of digital information, e.g. telegraphic communication5%
H04MTelephonic communication4%
H04RLoudspeakers, microphones, gramophone pick-ups or like acoustic electromechanical transducers4%
G10KSound-producing devices3%
H04BTransmission2%
H04WWireless communication networks2%
G06FElectric digital data processing1%
G06TImage data processing or generation, in general1%
PatentTitleYear
10,366,701Adaptive multi-microphone beamforming20199,978,394Noise suppressor20189,934,791Noise supressor20189,907,113'Efficiency of wireless wide area networks utilizing local wireless connections'20189,843,883'Source independent sound field rotation for virtual and augmented reality applications'20179,484,043'Noise suppressor'20169,437,211'Adaptive delay for enhanced speech processing'20169,437,203'Error concealment for speech decoder'20169,401,156'Adaptive tilt compensation for synthesized speech'20169,299,333'System for adaptive audio signal shaping for improved playback in a noisy environment'20169,269,365'Adaptive gain reduction for encoding a speech signal'20169,208,766'Computer program product for adaptive audio signal shaping for improved playback in a noisy environment'20159,208,767'Method for adaptive audio signal shaping for improved playback in a noisy environment'20158,650,028'Multi-mode speech encoding system for encoding a speech signal used for selection of one of the speech encoding modes including multiple speech encoding rates'20147,876,892'Multiple echo cancellation using a fixed filter delay'20117,796,623'Detecting and reporting a loss of connection by a telephone'20107,711,108'Fast echo canceller reconvergence after TDM slips and echo level changes'20107,593,852'Speech compression system and method'20097,558,295'Voice access model using modem and speech compression technologies'20097,532,581'Voice quality monitoring and reporting'20097,266,493'Pitch determination based on weighting of pitch lag candidates'20077,191,122'Speech compression system and method'20077,146,309'Deriving seed values to generate excitation values in a speech coder'20067,127,390'Rate determination coding'20067,120,578'Silence description coding for multi-rate speech codecs'20067,080,010'Complexity resource manager for multi-channel speech processing'20067,076,421'Method and system for supporting increased channel density'20067,072,832'System for speech encoding having an adaptive encoding arrangement'20067,010,480'Controlling a weighting filter based on the spectral content of a speech signal'20066,961,698'Multi-mode bitstream transmission protocol of encoded voice signals with embeded characteristics'20056,959,274'Fixed rate speech compression system and method'20056,937,979'Coding based on spectral content of a speech signal'20056,898,566'Using signal to noise ratio of a speech signal to adjust thresholds for extracting speech parameters for coding the speech signal'20056,873,956'Multi-channel speech processor with increased channel density'20056,856,954'Flexible variable rate vocoder for wireless communication systems'20056,842,733'Signal processing system for filtering spectral content of a signal for speech coding'20056,823,303'Speech encoder using voice activity detection in coding noise'20046,804,203'Double talk detector for echo cancellation in a speech communication system'20046,789,058'Complexity resource manager for multi-channel speech processing'20046,757,649'Codebook tables for multi-rate encoding and decoding with pre-gain and delayed-gain quantization tables'20046,735,567'Encoding and decoding speech signals variably based on signal classification'20046,721,712'Conversion scheme for use between DTX and non-DTX speech coding systems'20046,697,776'Dynamic signal detector system and method'20046,694,293'Speech coding system with a music classifier'20046,691,089'User configurable levels of security for a speaker verification system'20046,636,829'Speech communication system and method for handling lost frames'20036,618,700'Speech coder output transformation method for reducing audible noise'20036,606,591'Speech coding employing hybrid linear prediction coding'20036,604,070'System of encoding and decoding speech signals'20036,581,032'Bitstream protocol for transmission of encoded voice signals'20036,581,030'Target signal reference shifting employed in code-excited linear prediction speech coding'20036,574,593'Codebook tables for encoding and decoding'20036,523,002'Speech coding having continuous long term preprocessing without any delay'20036,510,409'Intelligent discontinuous transmission and comfort noise generation scheme for pulse code modulation speech coders'20036,493,665'Speech classification and parameter weighting used in codebook search'20026,466,904'Method and apparatus using harmonic modeling in an improved speech decoder'20026,463,414'Conference bridge processing of speech in a packet network environment'20026,397,176'Fixed codebook structure including sub-codebooks'20026,385,573'Adaptive tilt compensation for synthesized speech residual'20026,345,248'Low bit-rate speech coder using adaptive open-loop subframe pitch lag estimation and vector quantization'20026,330,531'Comb codebook structure'20016,330,533'Speech encoder adaptively applying pitch preprocessing with warping of target signal'20016,256,606'Silence description coding for multi-rate speech codecs'20016,240,386'Speech codec employing noise classification for noise compensation'20016,205,423'Method for coding speech containing noise-like speech periods and/or having background noise'20016,122,611'Adding noise during LPC coded voice activity periods to improve the quality of coded speech coexisting with background noise'20006,104,992'Adaptive gain reduction to produce fixed codebook target signal'20006,104,994'Method for speech coding under background noise conditions'20006,014,622'Low bit rate speech coder using adaptive open-loop subframe pitch lag estimation and vector quantization'20005,920,853'Signal compression using index mapping technique for the sharing of quantization tables'19995,781,880'Pitch lag estimation using frequency-domain lowpass filtering of the linear predictive coding (LPC) residual'19985,689,615'Usage of voice activity detection for efficient coding of speech'19975,664,054'Spike code-excited linear prediction'19975,255,343'Method for detecting and masking bad frames in coded speech signals'1993Career & activity timeline
Assembled from patent assignee/location history and declaration credentials.
1993Based in St-Bruno, CA· patent location history
1997–1999Patents assigned to ROCKWELL INTERNATIONAL CORP (4)· patent assignee history
1997–2019Based in San Clemente, CA, US· patent location history
2000–2004Patents assigned to CONEXANT SYSTEMS INC (22)· patent assignee history
2002Based in San Bernardino, CA, US· patent location history
2003–2016Patents assigned to MINDSPEED TECH INC (33)· patent assignee history
2004Based in Clements, CA, US· patent location history
2010–2017Patents assigned to SU HUAN-YU (8)· patent assignee history
2015–2019Patents assigned to QOSOUND INC (10)· patent assignee history
2015–2018Based in Irvine, CA, US· patent location history
Inventor activity
Assignee and location history derived from the patent record.
Assignees
| Assignee | Patents | Span |
|---|---|---|
| MINDSPEED TECH INC | 33 | 2003–2016 |
| CONEXANT SYSTEMS INC | 22 | 2000–2004 |
| QOSOUND INC | 10 | 2015–2019 |
| SU HUAN-YUself | 8 | 2010–2017 |
| ROCKWELL INTERNATIONAL CORP | 4 | 1997–1999 |
Locations
| Location | Patents | Span |
|---|---|---|
| San Clemente, California, US | 63 | 1997–2019 |
| Irvine, California, US | 8 | 2015–2018 |
| Clements, California, US | 1 | 2004 |
| San Bernardino, California, US | 1 | 2002 |
| St-Bruno, CA | 1 | 1993 |