摘要:
A method and apparatus reliably encode and decode information over a communication system. The method includes transforming two coefficients into two pairs of random variables, one random variable in each pair having substantially equal energy as one random variable in the other pair. The method further includes quantizing each of the pairs of random variables and entropy coding each quantized random variable separately creating an encoded bitstreams. The encoded bitstreams are received by a decoder which first determines which channels of the communication system are working. The encoded bitstream is entropy decoded, inversed quantized and inversed transformed. An inverse transform performs three different transformations depending upon which channels are working, i.e., whether the first, second or both channels are working.
摘要:
A transcoder-codec circuit arrangement that supports voice-switched hands-free radio operation. A first register is arranged to store a value indicative of a peak signal in a receive signal path, a first attenuator is coupled to the receive path, a second register is arranged to store a value indicative of a peak signal in a transmit signal path, a second attenuator is coupled to the transmit signal path, and a digital signal processor is coupled to the first and second registers and configured and arranged to update the value in the first register with a present peak receive signal level if the value in the first register is less than the present peak receive signal level. The digital signal processor is further arranged to update the value in the second register to a present peak transmit signal level if the value in the second register is less than the present peak transmit signal level. A microcontroller is coupled to the first and second registers and to the first and second attenuators and is configured to read values from the first and second register and adjust the first and second attenuators in response to the values.
摘要:
A coding method for a pulse transmission system specifies temporal and/or non-temporal pulse characteristics according to temporal and/or non-temporal characteristic value layouts having one or more allowable and non-allowable regions. The method generates codes having predefined properties. The method generates a pulse train by mapping codes to the characteristic value layouts, where the codes satisfy predefined criteria. In addition, the predefined criteria can limit the number of pulse characteristic values within a non-allowable region. The predefined criteria can be based on relative pulse characteristic values. The predefined criteria can also pertain to spectral properties and to correlation properties. The predefined criteria may pertain to code length and to the number of members of a code family. The pulse train characteristics may pertain to a subset of the pulse train.
摘要:
Voice recording and playback mode using the G.726 half-rate within the personal handy phone system (PHS). When a portable station within the PHS operates as a voice recorder (e.g., functioning as an answering machine), a cost effective system in accordance with the present invention is adapted to compress and store received voice/sound signals in order to increase the usage of limited memory resources provided within the portable station. The present invention also enables previously compressed and stored voice/sound signals to be decompressed and played back in various portable station playback modes. Specifically, the portable station receives a voice/sound signal in a full rate (e.g., 32 kilobits-per-second) 4-bit adaptive differential pulse code modulation (ADPCM) data format in compliance with the International Telecommunication Union (ITU) recommendation G.726. The present invention compresses this received voice/sound signal to a half rate (16 kilobit-per-second) 2-bit ADPCM data format in compliance with the ITU recommendation G.726 in order to increase the usage of the limited memory resources provided within the portable station. During a playback mode of the portable station, the present invention decompresses the previously compressed and stored voice/sound signal to facilitate its playback.
摘要:
In a PCM modem system in which equivalence classes are used to communicate information from a transmitter to a receiver, a method is provided to solve the problem of 180° phase reversals in the communications channel which result in a garbled transmission. This is accomplished by remapping the equivalence classes into a form that can be differentially encoded and decoded such that equivalence class identity is not lost during a phase reversal of the channel.
摘要:
Glitch filters, methods, and computer program products that utilize the generally monotonically increasing characteristics of the expected levels of code points to detect and remove noise spikes by replacing values in the code point sequence with new values based on the code points around a suspect value are provided. Measured values associated with two code points in the sequence of code points which are immediately higher in the sequence of code points than a code point of interest are evaluated so as to select a larger value of the two code points in the sequence as a first reference value. The first reference value is compared with a measured value associated with a code point in the sequence of code points immediately lower than the code point of interest to determine if the first reference value is smaller than the measured value associated with the code point in the sequence of code points immediately lower than the code point of interest. The smaller of the first reference value and the measured value associated with a code point in the sequence of code points immediately lower than the code point of interest is then selected so as to provide a first replacement value. The measured value associated with the code point of interest is then replaced with the first replacement value if the first reference value is smaller than the measured value associated with the code point of interest.
摘要:
A predictive decoder removes granular noise from the decoded signal by one of the following methods: discarding a predetermined number of least significant bits of the decoded signal; taking the sum of each two successive values of the decoded signal and dividing the sum by two to obtain an output signal; or resetting the decoded signal to a fixed value during inactive periods. Inactive periods are detected according to criteria involving an adaptively adjusted step size and a quantized difference signal, both of which are obtained in the decoding process.
摘要:
A digital system for filtering a single bit input signal according to the transfer function H(z), wherein H(z) has a gain G, a pole at location b0, and a zero at location a0. The digital system filters the single bit input signal without using computationally expensive multibit multiplication. The digital system achieves these advantages with a digital circuit having a first gain stage generating a gain corrected signal, a delay element generating a delayed gain corrected signal, a feed-forward stage generating a feed-forward signal, and a summer for generating an output signal based upon the sum of the gain corrected signal, the delayed gain corrected signal and the feed-forward signal.
摘要:
A One Bit Digital Quadrature Vector Modulator (DQVM) and a method of generating single sideband output signals are useful for a wide range of radio frequency, signal processing and wireless applications. The DQVM simplifies the necessary digital multiplication by using noise shaped one bit versions of both the baseband IB and QB signals to be modulated and the ILO and QLO modulating signals. The one bit DQVM enables a much faster digital implementation of the digital quadrature vector modulation function than can be achieved with conventional multi-bit digital techniques. In addition the single sideband upconversion of the DQVM achieves high suppression of the unwanted sideband by applying an offset to one of the low speed input samples. Digital vector modulators are an improvement over conventional analog vector modulators as they are not subject to the amplitude and phase matching problems inherent in analog vector modulators.
摘要:
DC transients are removed from a digital filter such as a sigma delta filter (in particular from a sigma delta high pass filter) from the outset by presetting an input summing node to a sigma delta modulator. While the input summing node may be preset using any appropriate input, in a disclosed embodiment, a sigma delta high pass filter is preset by switching a partial feedback term between an input containing the non-zero preset value and the normal input comprising the output from the input summing node. The preset value is chosen based on the value of the zero of the transfer function of the sigma delta high pass filter, e.g., with the complement of the gain factor.