Abstract: A system controller (106) is for transferring a low bit rate digital voice message. The system controller generates from an analog voice signal representing the voice message a set of speech model parameters, and generates a first derived set of speech model parameters from a first subset of the set of speech model parameters, the first derived set encoding the voice signal at a second voice quality and second vocoder rate that are less, respectively, than a first voice quality and vocoder rate. The system controller transmits (3610) the low bit rate-digital voice message comprising the first derived set of speech model parameters to a communication receiver (114). The communication receiver requests (3640) an incremental message when the quality of the voice message is unsatisfactory. The system controller generates and transmits (3555, 3650) an incremental message-and the communication receiver uses (3660) the incremental message to generate a higher quality voice message.

Abstract: A bandwidth efficient method of wireless communication among simultaneous multiple users includes the steps of monitoring (102) for voice activity from a plurality of sources (302 or 304) on a channel and detecting voice activity (107) among the plurality of sources during a predetermined time period. If voice activity is detected from only a first source, then code the voice activity from the first source and transmit a full-rate packet of data (109) to a wireless subscriber (50). If voice activity is detected from at least a first source and a second source amounting to N sources, then code such N sources using a 1/N-rate vocoder to obtain N different 1/N rate data packets (112) and combining (114) the N 1/N-rate data packets from at least the first source and the second source before transmitting (116) a full packet of data to the wireless subscriber.

Abstract: A system controller (106) includes a speech encoder (107) that encodes a low bit rate digital voice message. The speech encoder sets values of words of a header of the encoded message. The values of the words define a quantity of frames in the voice message, N, and define a vocoder rate used for the encoded message. The speech encoder sets a state of each indicator in each frame status field of N frame status fields that are transmitted after the header of the encoded message. The speech encoder assembles N frame data fields, wherein each of the frame data fields comprises a set of data words. The N frame data fields follow the N frame status fields. Each set of data words conforms to at least one of the vocoder rate and the states of the indicators. A decoder (3310) decodes the encoded low bit rate digital message.

Abstract: A system controller (106) includes a speech encoder (107) that dynamically segments frames of a low bit rate digital voice message. Speech model parameters have been generated in a sequence of frames. The speech model parameters include quantized speech spectral parameter vectors. The speech encoder selects (1820) a first quantized speech spectral parameter vector as a current anchor vector, selects (1820, 1830) a second quantized speech spectral parameter vector located a predetermined number of frames (LMAX) from the current anchor vector as a target speech parameter vector, and perturbs (1840) the target speech parameter vector to derive a plurality (K) of perturbed speech parameter vectors.

Abstract: A pitch determiner (931) of a system controller (106) that generates a smoothed pitch value for a current frame of a low bit rate voice message includes a pitch function generator (955) that generates a pitch detection function (PDF) for each frame of digital samples of a voice signal, a pitch candidate selector (960) that selects a future frame pitch candidate from a pitch detection function (PDF), and a pitch adjuster (978) that generates the smoothed pitch value. The pitch adjuster includes a subharmonic pitch corrector (965) that determines a future frame pitch value by performing pitch subharmonic correction of a future frame pitch candidate using a roughness factor of the frequency transformed window.

Abstract: A technique is used in a speech encoder (107) that reduces non-speech activity of a low bit rate digital voice message. Speech model parameters that include quantized speech spectral parameter vectors are generated in a sequence of frames. A determination is made as to which frames of the sequence of frames are voiced frames and which frames are unvoiced frames. A consecutive sequence of frames of unvoiced frames is identified (2330) as an unvoiced burst when a length, NUV, of the consecutive sequence of frames exceeds a predetermined length, Ns. A non-speech activity portion of the unvoiced burst is identified (2335-2365) and removed.

Abstract: A method (100) of compressing a digital signal that is parametrically modeled and encoded includes the steps of storing (102) the digital signal in a memory in a plurality of frames having a plurality of parameters in each frame of the plurality of frames, wherein the digital signal was encoded at a higher rate and converting the digital signal to a lower rate by selecting (106) from each frame of the plurality of frames a subset of the plurality of parameters and discarding (108) the subset of the plurality of parameters within each frame of the plurality of frames.

Abstract: A processing system (150) that generates a voice output signal having high precision controlled gain from a voice input signal by generating (225) a slow gain signal from the voice message by controlling a signal level gain of the voice signal using a setup gain and a first slow time constant, and then generating (230) a high precision controlled gain voice signal from the slow gain signal by controlling a signal level gain of the slow gain signal using at least one of a fast attack time constant and a slow release time constant.

Abstract: A method and apparatus for time-scale modification of speech using a modified version of the Waveform Similarity based Overlap-Add technique (WSOLA) comprises the steps of storing a portion of an input speech signal in a memory, analyzing the portion of the input speech signal to determined at least one filtered pitch value, calculating an estimated pitch value (12) from the at least one filtered pitch value, determining a segment size (14) in response to the estimated pitch value (12), the segment size (14) having a value greater than the estimated pitch value (12), and time-scale compressing (18) the input speech signal in response to the segment size determined.

Abstract: A method and apparatus for providing intelligible fast forward and reverse playback of messages which include time-scale compressed speech using a time scale modification technique. A receiver (2604) receives (2902) a message including speech compressed at a predetermined compression rate and a message rate identifier which identifies the predetermined compression rate which are then stored in a memory by a processor (2610). The processor (2610) processes (2912) the stored message to time-scale expand the compressed speech at an expansion rate that is equal to the predetermined compression defined by the message rate identifier so as to produce a normal speech playback speed. The processor (2610) also processes (3000, 3100) the stored message to time-scale expand the compressed speech at an expansion rate that is lower than the predetermined compression rate defined by the message rate identifier, so as to produce a perceptibly increased speech playback speed.

Abstract: A method used by a controller (112) and a communication receiver (122) for performing a time-scale modification of a signal comprised of an input stream of signal representations to form an output stream of signal representations. An output block of S.sub.o signal representations is selected from the output stream for use in overlapping signal representations from the output block with S.sub.o contiguous signal representations of the input stream, the S.sub.o contiguous signal representations beginning with a predetermined signal representation. The overlapping process includes determining (406) a maximum similarity measure between selected portions of the input and the output streams, applying (410) a weighting function to each, and summing the results of each. The overlapping process further includes replacing (412) a selected portion of the output stream with the latter summation, and selecting (414) a new set of samples from the input stream to append to the output stream.

Abstract: A method and apparatus is for use in a radio communication system (100) for delivering a message from a system controller (102) to a selective call device. In the system controller (102) the method includes the steps of generating a first message intended for the selective call device (106), changing a delivery state of the first message to undelivered after the first message has been transmitted at least once from the system controller (102), and storing the first message in the system controller (102) when the delivery state is changed to undelivered. In the selective call device the method includes the steps of determining that the first message is stored in the system controller (102) and presenting user with information which indicates that an undeliverable message is being stored by the system controller (102) in response to determining that the first message (102) is stored in the system controller (102).

Abstract: A communication system using voice compression includes a transmitter base station (113) and a selective call receiver (112). The transmitter base station (113) includes an input device (204) to receive an audio voice message which is stored in a memory (209). A processing device (208) digitizes the audio voice message to provide an input signal which is divided into a sequence of equivalent time frames, and differences in a short-term frequency spectrum between adjacent time frames is determined to derive distance measurements. A speed factor is computed as an average of the distance measurements, and the input signal is time-scales in accordance with the speed factor and then transmitted by a transmitter (102).

Abstract: The present invention comprises a method for compressing a plurality of voice signals within a voice communication resource (see FIG. 6) having a given bandwidth within a voice communication system (100). The method comprises the steps of subchanneling the voice communication resource into a plurality of subchannels (441, 442, 443), placing a pair of the plurality of voice signals (401, 402) on a subchannel (441); modulating the pair of the plurality of voice signals (401, 402) about a pilot signal (581) within the subchannel (441) using single sideband modulation; and compressing the time of each of the voice signals (401, 402) within the plurality of subchannels (441, 442, 443), wherein these step provide a compressed voice signal.

Abstract: A modulation scheme (600) useful in a voice paging system in which both of two orthogonal modulation components (500 and 510) are used to carry two halves of a single voice message destined for a receiver, or two separate voice messages for a receiver. The single voice message is transmitted in half the time.

Abstract: A method and apparatus for mitigating effects of multipath fading, interference, and noise in a single sideband (SSB) signal transmitted by a radio communication system and demodulated by a portable subscriber unit (122) utilizing a demodulator (304) compensated by a pilot signal power (406). The apparatus performs the mitigation process by calculating (708) a pilot fading threshold in addition to a muting threshold. When samples of the mean pilot signal power fall below the fading threshold, the compensation of the demodulated signal is adjusted (714). In addition, when the mean analog message signal power falls below the muting threshold, the samples included in the predetermined portion of the message are muted (718).