Abstract: Methods and apparatus for speech coding are disclosed for converting analog speech signals to digital speech signals for transmission. The speech coder, utilizing CELP techniques, includes a first filter for filtering out the spectral information from the speech signal. The spectral information is provided for transmission. A second filter is provided for filtering out the pitch information from the speech signal and such pitch information is also provided for transmission. A codevector generator determines, in one embodiment, the characteristics of a bi-pulse codevector representative of the speech signal. In this embodiment the impulse response of the first filter is truncated for determining the codevector characteristics. In this embodiment it is also preferred to determine the codevector characteristics by conducting a numerator only search in relation to a traditional fraction used for determining codevectors.

Abstract: An improved frame structure for use in mobile communications systems utilizing slotted protocols is provided. According to the present invention, consecutive slots in a frame are assigned to each mobile unit so that the mobile unit transmits to a base station during the first slot and receives a transmission from the base station during the second slot. This frame structure, permits the mobile unit to monitor various frequencies for a longer and more continuous period of time. It is important for the mobile unit to have enough time to monitor frequencies while it is not transmitting or receiving so that base station handoff and interference avoidance handoff techniques can be performed by the mobile communications system. Furthermore, according to the frame structure provided by the present invention, the mobile units can be designed with a standard frequency synthesizer.

Abstract: A programmable digital modulator and methods of modulating digital data for transmission by a communication system according to operating parameters selected for various applications are provided. A two-chip system is utilized by a preferred embodiment of the invention. One chip comprises a PROM for storing impulse response data which would result from filtering the data to be transmitted. The second chip comprises a data interface for accepting input data, an address generator for generating an address of the PROM where the impulse response data is stored which corresponds to the data input to the chip and for causing the PROM to output the impulse response data stored at the address generated, and a data modulator for modulating a carrier signal with the impulse response data provided by the PROM.

Abstract: A phase-locked loop frequency modulation circuit that compensates for the inability of a phase-locked loop to pass low-frequency content of an input modulation signal, and that may be utilized with existing communication apparatus, such as a cellular telephone voice radio, for accurate data transmission without having to modify such apparatus, includes a compensation circuit for processing an input modulation signal to provide a compensation signal that is added to a loop filter output signal by processing the input modulation signal to provide the same effect as adding the input modulation signal to an integrated input modulation signal that is filtered by a filter having the same transfer function as the loop filter to provide the compensation signal.

Abstract: A method for reducing power consumption in a cellular telephone that receives multiple copies of a message comprises the steps of receiving a copy of a message block having a plurality of data bits; generating soft output signals representative of all data bits in the message block; computing a block quality metric indicative of the reliability of the message block; comparing the block quality metric with a first threshold; if the block quality metric bears a predetermined relation to the first threshold, combining the copy of the message block with a previously accepted message block to obtain a combined message block; determining the number of erasures in the combined message block; comparing the number of erasures with a second threshold; if the number of erasures is below the second threshold, decoding the combined message block to determine the number of errors in the combined message block; comparing the number of errors with a third threshold and, if the number of errors is below the third threshold, a

Abstract: A digital demodulator and method for demodulating digital data representing a phase shift keyed (PSK) signal are provided. The demodulator comprises a phase detector, automatic frequency controller, automatic timing recovery controller, data decoder, and unique word detector. According to the method of the present invention, a PSK signal is received and digitized to substantially remove the signal's amplitude characteristics. The phase detector receives an input of the digital data and based upon transitions in the data from a high state to low state and from a low state to a high state, provides phase estimates. The phase estimates are converted by the data decoder into binary data representing the symbols transmitted to form the PSK signal. A number of overlapping windows of digital data are used to determine phase estimates.

Abstract: Apparatus and method for reconstructing non-quantized adaptively transformed voice signals are shown to include noise shaping wherein the spectral envelope is scaled prior to generating bit allocation and energy substitution which is achieved after dequantization by generating the spectral envelope information for each block of transform coefficients based upon side information, generating transform coefficients which correspond to transform coefficients which were not de-quantized and for substituting the generated transform coefficients into said blocks; and transforming said blocks of de-quantized transform coefficients and generated transform coefficients from said transform domain into said time domain.

Abstract: A method and apparatus for removing the periodicity from a speech signal in a transform coder prior to the quantization of the speech signal, which speech signal is a sampled time domain speech signal composed of information samples, the transform coder sequenctially segregating the speech signal into blocks of information samples, is shown to include apparatus and method for determining the pitch in each of the sample blocks, determininig a long term predetermined parameter (LTP) for each of the blocks based on the pitch determined for each block, calculating a periodicity value for each sample in the block wherein the calculation of the periodicity value is based upon the pitch and the long term predictor parameter, generating a revised block of difference samples by subtracting the periodically value from the corresponding sample, and performing adaptive transform coding on each of the difference blocks.

Abstract: A transform coder operates on a sampled speech signal transformed from the time domain to a frequency domain to develop pitch information in relation to a given speech signal. The coder segregates groups of information samples into blocks, transforms each block of samples, and generates an auto-correlation function of the transformed signal for each block. Next, the coder determines the pitch period and pitch gain from the auto-correlation function, and determines the striation magnitude and energy from the pitch period and pitch gain. Then a reference pitch model including a number of data points is retrieved from data memory. A striation scaling factor is generated in response to the striation magnitude and energy, and is multiplied by each of the retrieved data points to adaptively generate a pitch model. Finally, the adaptively determined model is sampled to establish the pitch information.

Abstract: Adaptive transform coding of a speech signal by a single digital signal processing chip is performed at low bit rates with reduced quantization noise and distortion. A windowed speech signal is transformed and quantized. New processes are shown for generating envelope information as well as bit allocation which control quantization. The quantized signal and necessary side information are formatted for transmission and subsequent decoding.