Abstract: Disclosed herein are methods for analyzing ion mobility data. In one embodiment, a method for analyzing ion mobility spectrometry data is disclosed. The method comprises conducting a number of ion mobility scans on a sample, generating an ion map comprising a first ion line, comparing the first ion line to a standard deviation, wherein the standard deviation is calculated from the system measurement noise, and determining if the sample is a contraband substance.

Abstract: A system for training a linearization compensation model includes a tone generator for providing at least two different RF tones, receiver path components for processing the RF tones, an analog-to-digital converter for converting the processed RF tones into digital signals, and a processor for using the digital signals to generate the linearization error compensation model. The resulting compensation model is particularly useful in a linearization system which includes a receiver for measuring a signal, an electro-optical modulator configured for converting the measured signal to an optical signal, an optical-electrical detector configured for converting the optical signal to an analog electrical signal, an analog-to-digital converter for converting the analog electrical signal into a digital signal with the processor being used for removing linearization errors from the digital signal.

Abstract: A method for decoding (voice)data where the data is encoded by a finite-state data encoder (216) for transmission over a data channel (18), 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 “current” 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A vocoder transmitter (12) sends coded speech to a vocoder receiver (16) over a limited-bandwidth channel (14). The transmitter includes a LPC-residual-based first gain estimator (46), which gain value works well with women's voices, but less so with men's. A second, analysis-by-synthesis, gain estimator (42) therein, uses a unit-gain version (342) of the receiver's synthesizer, whose input (24) and output (310) speech power is estimated, to produce the second gain value from their ratio (312). The second gain value has been found to work well with men's voices, but to produce "explosive" artifacts with women's. A combiner (518) weights these two gain estimates, under control of the estimated pitch, to produce the vocoder gain estimate transmitted to the vocoder receiver (16). In a particular embodiment the first gain estimate predominates at low pitch periods, and the second gain estimate at higher ones, with crossover at about 45 sample lags for an 8000 Hz sampling rate.

Abstract: 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.