General Electric Company

Lockheed Martin Corporation

Ericsson Gmbh

Steven Robert Koch

Richard Louis Zinser

Mark Lewis Grabb

Glen William Brooksby

Richard L Zinser, Jr

William Thomas Hatfield

William C Hughes

Raymond L Toy

Charles R Stein

Neil R Shapiro

John Anderson Fergus Ross

Nick Andrew Van Stralen

Seth D Silverstein

John Bailey Anderson

Philip T Choong

A method for transcoding a bit stream encoded according to a mixed-excitation linear prediction (MELP) standard to a bit stream encoded according to a linear predictive coding (LPC) standard, including: decoding a bit stream into a first set of vocoder parameters compatible with the MELP standard; transforming the first set of vocoder parameters into a second set of vocoder parameters compatible with the LPC standard without converting the first set of vocoder parameters to an analog or digital waveform representation; and encoding the second set of vocoder parameters into a bit stream compatible with the LPC vocoder standard.

MELP-to-LPC transcoder

A method for transcoding a bit stream encoded according to a linear predictive coding (LPC) standard to a bit stream encoded according to a mixed-excitation linear prediction (MELP) standard, including: decoding a bit stream into a first set of vocoder parameters compatible with the LPC standard; transforming the first set of vocoder parameters into a second set of vocoder parameters compatible with the MELP standard without converting the first set of vocoder parameters to an analog or digital waveform representation; and encoding the second set of vocoder parameters into a bit stream compatible with the MELP vocoder standard.

LPC-to-MELP transcoder

A method for enhancing speech prediction coefficient signals, including: a) receiving a set of prediction coefficient signals; b) calculating a set of amplitude spectrum coefficient signals based on the prediction coefficient signals; c) calculating a set of enhancement coefficient signals based on the amplitude spectrum coefficient signals; d) calculating a set of enhanced autocorrelation coefficient signals based on the enhancement coefficient signals; e) converting at least some of the enhanced autocorrelation coefficient signals into reflection coefficient signals; f) testing the range of values of the reflection coefficient signals; and g) converting at least some of the tested reflection coefficient signals into prediction coefficient signals.

Frequency domain format enhancement

The system and method of the present invention comprises a compressed domain voice activity detector that detects the presence or absence of voice activity in a digital input signal. The method includes converting a digital input signal into parametric data. The parametric data is subsequently analyzed, and then compared against a background noise threshold to determine if voice activity is present.

Compressed domain voice activity detector

The system and method of the present invention comprises a compressed domain conference bridge. The method includes maintaining full intelligibility for all users even when there are multiple simultaneous talkers. The bridge also allows a user to hear a plurality of talkers over a single channel.

Compressed domain conference bridge

The system and method of the present invention comprises a compressed domain universal transcoder that transcodes a bit stream representing frames of data encoded according to a first compression standard (TDVC coding standard) to a bit stream representing frames of data according to a second compression standard (MELP coding standard). The method includes decoding a bit stream into a first set of parameters compatible with a first compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard without converting the first set of parameters to an analog or digital waveform representation. Lastly, the second set of parameters are encoded into a bit stream compatible with the second compression standard.

TDVC-to-MELP transcoder

A method for decoding (voice)data where the data is encoded by a finite-state data encoder ( ) for transmission over a data channel ( ), includes the step of procuring a table of joint statistics representing the probability of occurrence, in a frame of source data of each of the bits of the frame. The joint statistics may be determined ahead of time. The method includes the step of calculating intermediate gamma signals in response to the decoded data of the preceding frame and the joint statistics. The joint statistics source distribution signals represent the likelihood that, for a given logic level of the preceding bit, the &ldquo;current&rdquo; bit takes on a particular state; for uncorrelated bits, this value is 0.5. State probability signals and transition probability signals are generated from the gamma signals. The state probability and transition probability signals are processed to produce bit probability signals indicative of the probability that the current bit is in a given state, from which hard bit decisions can be made.

Channel decoder using vocoder joint statistics

A speech coding system and associated method relies on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes an LPC synthesis filter (90), a Gaussian noise generator (80) for generating unvoiced excitation, an epoch-adaptive harmonic generator (70) for generating voiced excitation for pitch harmonics below voicing cutoff frequency, and an excitation summer (72) for summing the voiced and unvoiced excitation generated by the Gaussian noise generator (80) and the harmonic generator (70). The output of the excitation summer (72) is provided to the LPC synthesis filter (90) to generate synthesized speech. The system and method provides natural sounding synthesized speech at a low bit rate.

CELP speech synthesizer with epoch-adaptive harmonic generator for pitch

A speech coding system (10) and associated method rely on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes a Linear Predictive Coding (LPC) filter (90) having an input and an output. The LPC filter (90) provides synthesized speech at its output in response to voiced and unvoiced excitation provided at its input. A harmonic generator (70) for providing voiced excitation to the input of the LPC filter includes a pitch-related randomized adaptive phase off-setter. The adaptive phase off-setter off-sets the harmonic phases of the voiced excitation in accordance with the fundamental pitch frequency of respective frames of speech. The system and method thereby reduce perceived buzziness of synthesized speech provided at the output of the LPC filter.

Speech coding system and method including harmonic generator having an

A speech coding system and associated method rely on a speech encoder and a speech decoder. The speech decoder includes a Linear Predictive Coding (LPC) filter having an input and an output. The LPC filter provides synthesized speech at the output in response to voiced and unvoiced excitation provided at the input. A harmonic generator for providing voiced excitation to the input of the LPC filter includes a spectral formant enhancer for attenuating the amplitude of harmonics generate by the harmonic generator in spectral valleys between format peaks of respective frames of voiced speech. The system and method reduce perceived buzziness while increasing perceived spectral depth of synthesized speech at the output of the LPC filter.

Speech coding system and method including spectral formant enhancer

A speech coding system and associated method relies on a speech encoder and a speech decoder. The encoder includes a spectral quantizer for computing line spectral frequencies (LSFs) for respective frames of speech and for quantizing the LSFs to obtain a minimum bit representation of a spectral envelope of each respective frame of speech. For even numbered frames of speech the LSFs are quantized using a vector quantization technique. For odd numbered frames of speech samples the LSFs are quantized using a dynamic bit allocation (DBA) method. The dynamic bit allocation method determines an interpolation factor for interpolating between the LSFs of the previous and next frames. According to the dynamic bit allocation method the most perceptually important LSFs are represented by relatively more bits, while the least perceptually important LSFs are represented by relatively fewer bits. The system and associated method thereby reduces an overall bit rate required to represent, transmit or store the speech samples.

Speech coding system and method including spectral quantizer

A speech coding system and associated method relies on a speech coder and a speech decoder. The speech coder provides speech spectrum information, including Line Spectral Frequency (LSF) information, for respective frames of speech. The LSF information is provided to the speech decoder. The speech decoder includes an LSF smoothing filter comprising an adaptive finite impulse response filter for averaging consecutive LSFs which are similar. Consecutive LSFs which are dissimilar are not averaged. The system and method causes a reduction in the shaky quality of synthesized speech without causing a corresponding decrease in the clarity of the synthesized speech.

Speech coding system and method including adaptive finite impulse

A speech coding system and associated method relies on a speech encoder (15) and a speech decoder (20). The speech encoder (15) includes a voicing cut off frequency analyzer (60). Voicing cut off frequency analyzer (60) includes voicing cut off frequency estimator (61) and voicing cut off frequency quantizer (62). Voicing cut off frequency estimator (61) estimates a voicing cut off frequency value for respective samples of an input speech waveform (1). To accomplish this, voicing cut off frequency estimator (61) utilizes a bandpass filter to estimate a frequency above which a sample of speech is voiced and below which the sample of speech is unvoiced. Voicing cut off frequency quantizer (62) quantizes the estimated voicing cut off frequency value and provides, for respective samples, a voicing cut off frequency index signal (6) which may be stored or transmitted. Voicing cut off frequency index signal (6) may comprise as few as 1 bit, and in a preferred embodiment, as few as 3 bits.

Speech coding system and method including voicing cut off frequency

A speech coding system (10) and associated method relies on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes a harmonic generator (70) which modulates the phase of each generated harmonic with a low frequency, low bandwidth signal to remove the buzzy quality of the speech and to produce natural sounding speech. The amplitude of the phase modulating signal is adjusted in accordance with the harmonic magnitude. For harmonics residing in a spectral valley the amplitude of the modulating signal is relatively large and for harmonics residing near spectral peaks, the amplitude of the modulation signal is relatively small.

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Niskayuna, NY, United States of America

Steven Robert Koch