Abstract: An on-site monitor is disclosed for providing automatic measurements of chemical concentrations in a liquid or gaseous environment. The opto-electronic monitor has an extended probe, which can be inserted into the liquid or to be in contact with the ambiance, comprising a indicator chamber for continuously supplying fresh indicator material, a light source, and a detector. The light source and the detector are either embodied outside of liquid or into the extended plastic probe inside the liquid. A delivery unit, which is activated by a timer, delivers a small amount of indicator material into the liquid or a sampling cavity. The indicator material reacts with the ions in the liquid automatically. The light source has a wavelength which matches with the indicator's absorption wavelength and the detector measures the transmission (or reflection) through the indicator material or solution.

Abstract: A circuit and method for Ethernet combination chaining of auto-negotiation sessions for multiple physical layer capability. The present invention provides control circuitry for arbitrating between different physical layer circuit types that heretofore were allowed autonomous auto-negotiation processes because only one type of physical layer circuitry was applied at each end of a communication line. The present invention operates effectively within a hub of a local area network (LAN) communication system where multiple communication standards can be employed within circuitry that communicate at different ends of a communication line, e.g., twisted pair wire. In the environment of the invention, different types of physical layer circuits are integrated within communication devices. The invention provides a mechanism whereby, at each end of the communication line, a first physical layer circuit (e.g., the master) of a first type is allowed to auto-negotiate under the IEEE 802.

Abstract: A signal compression system includes a coder and a decoder. The coder includes an extract unit for extracting an input feature vector from an input signal, a coder memory unit for storing a predesigned vector quantization (VQ) table for the coder such that the coder memory unit uses a set of primary indices to address entries within the pre-designed VQ table, a coder mapping unit for mapping indices from a set of secondary indices to the first set of indices, and a search unit for searching for one index out of the set of secondary indices, wherein the index from the set of secondary indices corresponds to an entry in the coder memory unit, and the entry best represents the input feature vector according to some predetermined criteria.

Abstract: The concept of integer ratio sample rate conversion is extended to fractional values, where the exact ratio is expressed as ?M+N/D!/L with L being a constant. This methodology allows sample rate conversion for all synchronous rates, where all rates are derived from the same crystal oscillator. For the implementation, the value of N/D is accumulated and depending on the overflow condition, either M or M+1 is selected as sample rate ratio for the conversion of each sample.

Abstract: A methodology for a modem control channel is disclosed. The channel allows faster seamless rate change and precoder tap exchange than the baseline procedure for seamless rate change, allowing for more robust transmission of control information. It can also be use to convey side-information in the case of multiple data applications, serving the purpose of mode switching. Thus, with a single control channel both the needs for seamless rate change and transmitting control information for multiple data applications can be met.

Abstract: A cordless direct sequence spread spectrum telephone using a direct conversion radio architecture wherein the code sequence used for spreading and despreading is an even code sequence to prevent addition of low frequency noise in the down-conversion process, the preferred even code being a 12-chip code and, more particularly, being an even 12-chip code that is generated by adding one bit to an inverted, shifted, and/or reversed 11-chip Barker sequence.

Abstract: A system for decimating a source signal with a source sample rate by a non-integer factor to a sink signal with a sink sample rate is disclosed. The system first determines a decimation factor based on a ratio of the source sample rate to the sink sample rate. The system then decimates the source signal by the decimation factor to an intermediate ("IM") signal with an IM sample rate which is larger than the sink sample rate. The IM signal is then applied to a sample-rate converter to down convert to the sink signal with the sink sample rate, using a decimation factor between only 1 and 2.

Abstract: A low-power stop mode "snooze" timer is implemented in the processing unit of a communication system, which runs without the use of the usual crystal-controlled oscillator. The "snooze" timer is implemented by a second oscillator with associated timer circuit. The second oscillator runs continually and feeds a timer which is loaded with a suitable value. The timer, when timed out or expired, will reactivate the main processing functions, thus allowing the successful reception of an external signal from a remote transmission source. To compensate for the inaccuracy of the second oscillator due to its dependence upon voltage, process and temperature, the snooze timer's output is calibrated during the wake-up periods against a timing signal derived from the internal crystal-controlled timing source. This calibration compensates for the inherent frequency instability of the second oscillator and allows a timing signal of sufficient accuracy to be maintained over the stop cycle.

Abstract: A method of providing redundancy in an electrically alterable memory divided into addressable blocks within the memory. The electrically alterable memory should have more available blocks than required by the firmware. The method includes the steps of testing the memory to create block status information about good versus bad blocks, writing the block status information into a table in the electrically alterable memory, reordering the table so that contiguous series of entries in the table point only to good blocks, and then providing a block status array comprising block select registers which are initialized with the corresponding ones of the contiguous series of entries so that a nominal address to a bad block is redirected to a good block.

Abstract: A synthesizer system includes a CPU and host memory operating a software routine. The software routine stores a first part of each waveform signal in a sample pool of host memory and provides remaining portions of selected musical sounds from the hard drive to a stream cell array without an audio perceivable delay. The synthesizer system utilizes a caching system which allows low cost, high storage devices to be utilized in an audio synthesizer system. MIDI control signals are provided to an audio processor for selecting appropriate digital waveform signals.

Abstract: A space vector data path for integrating SIMD scheme into a general-purpose programmable processor. The programmable processor uses a mode field in each instruction to specify, for each instruction, whether an operand is processed in either one of vector or scalar modes. The programmable processor also has a plurality of sub-processing units for receiving the operand and, responsive to an instruction as specified by the mode field in each instruction, for processing the operand in either one of the vector or scalar modes, wherein the vector mode indicates to the plurality of sub-processing units that there are a plurality of elements within the operand and the scalar mode indicates to the plurality of sub-processing units that there is but one element within the operand. For the vector mode, each element is processed by one of the sub-processing units concurrently to generate a vector result.

Abstract: A method is disclosed for generating frame voicing decisions for an incoming speech signal having periods of active voice and non-active voice for a speech encoder in a speech communication system. The method first extracts a predetermined set of parameters from the incoming speech signal for each frame and then makes a frame voicing decision of the incoming speech signal for each frame according to a set of difference measures extracted from the predetermined set of parameters. The predetermined set of extracted parameters comprises a description of the spectrum of the incoming speech signal based on line spectral frequencies ("LSF"). Additional parameters may include full band energy, low band energy and zero crossing rate.

Abstract: An apparatus and method of quantizing a sequence of input data vectors using delayed decision switched prediction and vector quantization. The method has the following steps of operation: (a) predicting a next vector element from said sequence of input data vectors to generate a set of prediction vectors; (b) subtracting the set of prediction vectors from the next vector element to generate a set of prediction error vectors; (c) multi-stage vector quantizing the set of prediction error vectors to generate a set of quantized prediction error vectors with each of the stages having at least one of the tables and local decision means to generate a final quantization error vector according to a predetermined distance measure; (d) selecting one predictor out of the set of predictors from the switched prediction step and selecting, for each of the stages, at least one entry from the set of tables of the vector quantization step according to the predetermined distance measure, generating a quantized data vector.

Abstract: A method of digital data transmission between a first transceiver and a second transceiver is disclosed. The first transceiver has a first transmitter and a first receiver and the second transceiver has a second transmitter and a second receiver. The first transmitter also contains a programmable encoder for encoding its transmission, while the second receiver uses at least one of a plurality of error correcting codes (ECC's) for decoding its reception. Each ECC can be specified by a predetermined set of parameters.

Abstract: A method and apparatus for iterative carrier frequency estimation of a base station by a mobile station in a wireless communication system is disclosed. Upon detecting a frequency correction burst ("FCB") signal from the wireless transmission of the base station, the mobile station first buffers the FCB signal and then filters the buffered signal with a bandpass filter to generate a first set of filtered data. After filtering, the first filtered data is again filtered with an auto-regressive filter to generate a corresponding set of second filtered data. With the second filtered data, iterative pole estimation of the AR filter and filtering by the AR filter is made for a predetermined set of iterations to generate a converged pole estimate, which can then be used to derive the carrier frequency of the base station.

Abstract: A circuit for reducing switching noise caused by harmonics of an output coupled to a band of interest in a mixed signal communication device, with the mixed signal communication device having a predetermined sampling clock.

Abstract: An subsystem processor for converting an input audio signal to complex audio samples for QADM processing by a modem for communication with a remote modem is disclosed. The audio process comprises a decorrelator for processing the input audio to generate a residual signal, an adaptive scaler for adaptively scaling the residual signal to generate a scaled residual signal, a randomizer for phase-randomizing the scaled residual signal to generate a complex signal with randomized phases, and a limiter for limiting the complex signal with randomized phases, such that the complex signal is transmitted by the modem to a remote modem.

Abstract: An adaptive frequency correction burst detector for GSM handset systems in a wireless communication system is disclosed. The detector uses a bandpass filter to filter the complex received samples to generate a first filtered output r(n). The first filtered output is then applied to a notch filter for the notch filter to adaptively generate a second filtered output y(n). An error calculation unit is used for calculating an error signal based on the difference between the second filtered output y(n) and a desired signal d(n), which is zero in the case of a notch filter. A notch filter coefficient updating unit is used for adaptively updating the notch filter using a least-mean-square algorithm. Then an error power is estimated based on the second filtered output y(n), while the signal power is estimated based on the first filtered output r(n).

Abstract: A resolution conversion system for real-time expansion and reduction of a first plurality of input pixels into a second plurality of output pixels. The resolution conversion system comprises a base-n counter for counting input pixels as each pixel is captured. The base-n counter has a plurality of carry out signals. Also, there is a control register for selectively enabling the carry out signals of the base-n counter based on a predetermined resolution ratio between the first plurality of input pixels and the second plurality of output pixels. The control register uses a mode bit to select either one of inserting and removing operations. An inserting logic unit is used for inserting, as indicated by the mode bit, a pixel into the plurality of input pixels upon a transition of one of the carry out signals if the carry out signals have been enabled by the control register.

Abstract: Multiple speech bit-stream frame buffers are used between the controller and the speech decoder. Whenever excessive or missing speech packages are detected, the speech decoder switches to a special corrective mode. If there is too much, the buffered frames are played out fast; if there is too little the buffered frames are played out slowly. For the fast play, some speech information has to be discarded, while for the slow play some speech-like information has to be synthesized. The speech may be handled in sub-frame units, which may be 52 samples at a time. Low energy, silent or unvoiced sub-frames, which also indicate non-periodicity, are detected and manipulated. Moreover, the decoded signal is manipulated at the excitation phase, before the final LPC synthesis filter, resulting in a transparent perceptual effect on the manipulated speech quality. Additionally, the buffers are enlarged such that the problem caused by controller asynchronicity is eliminated.