Abstract: To address the need for reducing pilot overhead, a method and apparatus for transmitting and receiving data is provided herein. A communication system having a transmitter (100) that utilizes adaptive modulation and coding (AMC) and utilizes lower-order modulated (e.g., QPSK) data as a substitute for pilot symbols during transmission. Data intended for a user utilizing a lower order modulation can be reliably detected, and if properly processed, used as pilots for a user that is trying to receive a higher order modulation (e.g., 64-QAM). Because lower order modulated data is utilized for pilots, fewer overall pilots are used. This greatly reduces pilot overhead, increasing data throughput.

Abstract: Channel assignment is performed by utilizing a base station controller (BSC) or a base station depending upon a mobile unit's RF environment and/or the type of call being made. By performing call setup in this manner, mobiles that access with no soft handoff requirement (approximately 50% of mobile in IS-2000 systems) will have faster setup times.

Abstract: A communication system is designed that reduces co-channel interference when heterogeneous users are sharing spectrum. The system design enables a plurality of pairs of users to communicate their channel utilization time periods by using the durations and start or stop times of certain transmissions to signal the start times and durations of later transmissions. Other system users using possibly different communications protocols and different modulation technologies can measure the durations of received transmissions and to infer channel usage based on these durations.

Abstract: The present invention provides a method and apparatus for improving the audio quality of a signal by reducing the effect of mis-determining the frame rate of a frame. The method includes the steps of determining that the frame rate of the current frame of information is eighth rate (324/340), determining that the previous frame was a full rate frame (334) and resetting the filter states of a speech decoder (336). The method further comprises the steps of utilizing alternative symbol error thresholds based on the number of consecutive frames with the same frame rate (308/328).

Abstract: A method and apparatus for data transmission is provided herein. In accordance with the preferred embodiment of the present invention a loss-ratio estimator (105) estimates a current loss (L) for a communication channel (108). Once the actual loss for the channel is known, a generator (104) compares the actual loss (L) to a target loss (T). A retransmission control parameter (R) is then adjusted by the generator (104) and output to a transmitter 103 where it is used to control the retransmission behavior and to determine when to abort a bad frame. When a bad frame is aborted, transmitter 103 indicates the abortion to a receiving device (102). A receiver then utilizes the indication to stop reporting the bad frame in all subsequent ACK/NAKs.

Abstract: A long code scrambler has been provided to scramble the data bits prior to QAM mapping. Different data rates are accommodated by using different long code m-tuples sampled at the modulation symbol rate, where m equals 2, 4 or 6 for 4, 16 or 64 QAM, respectively. In order to support 1XEV-DV users employing multiple Walsh code channels, a long code 6-tuple sample is permutated, to further randomize the QAM symbols. As appropriate for the modulation order, 2, 4 or 6 bits of the permutated 6-tuple are applied to the data sequence. That is, in addition to randomizing the modulation symbols for a given code channel, each code channel will likely have unique QAM symbols at any given symbol time.

Abstract: A method in a communication system (100) includes transmitting from a source user (101) a first data packet (111) over a first time frame (121) having a finite time period (131), transmitting from source user (101) a second data packet (112) over a second time frame (122) immediately subsequent to first time frame (121), detecting an acknowledgment of acceptable reception of data packet associated with either first or said second data packets (111 and 112), repeating transmission of first and second data packets (111 and 112) in a sequence of first and second time frames (121 and 122) in a time frame sequence (190) until the detection.

Abstract: The present invention provides a method for routing data from a service request in a communication system. A context is transferred from a packet data gateway, such as a Serving GPRS (GSM Packet Radio Service) Support Node (SGSN), to a Base Station System (BSS) coupled thereto. The context specifies where data for this user should be sent. The service request is received at the BSS and includes data. The data from the service request is routed to an appropriate location, based at least in part upon the context.

Abstract: Communication systems employing subchannel communication such as OFDM or multicode CDMA have a disadvantage of power variation of the transmitted signal. The present invention provides reduced power variation without compromising performance or bandwidth. Information symbols to be transmitted on the individual subchannel are fed to encoders which map the information symbols into channel symbols by increasing their order. The encoding in response to a forward error correction scheme and includes selection between redundant symbol values to reduce power variation. Preferably the encoder includes a trellis coder jointly encoding the symbols and a compensation data set to reduce power variation.

Abstract: A method and apparatus for speech reconstruction within a distributed speech recognition system is provided herein. Missing MFCCs are reconstructed and utilized to generate speech. Particularly, partial recovery of the missing MFCCs is achieved by exploiting the dependence of the missing MFCCs on the transmitted pitch period P as well as on the transmitted MFCCs. Harmonic magnitudes are then obtained from the transmitted and reconstructed MFCCs, and the speech is reconstructed utilizing these harmonic magnitudes.

Abstract: A method for detecting a presence of an excess delay spread within a received signal is provided. First, a quantitative estimation of a similarity of two segments of the received signal is performed. Thereafter, the presence of the excess delay spread within the received signal based upon the quantitative estimation is determined. In determining the presence of an excess delay spread, the quantitative estimation can be compared to a detection threshold.

Abstract: Remote units within a first communication system (121) utilizes frequencies (channels) from a second communication system (120) when placing a call. When a first remote unit (106) wants to call a second remote unit (107) using the channels, the first remote unit sends a base station (103) a list of top N traffic channel blocks from the second communication system along with the identity of the second remote unit. The base station then pages the second remote unit using a signaling channel reserved by second system. The second remote unit responds to the page with its list of best traffic channel blocks/chunks. The base station then picks the best overlap of idle channels from the list received by the remote units, and sends the list to the remote units instructing them use those traffic channels. Once the remote units receive the available channels, they can begin their call.

Abstract: A downlink frame (401) is divided in to similar sized resource blocks (403, 405, 407) with each co-channel sector scheduled to transmit from the beginning of its respective assigned resource block. Transmissions to remote units within the particular sector will occur only within the particular resource block, up to a point where all N resource units have been utilized. Beyond that point, additional transmissions are scheduled to be transmitted at the end of the resource blocks assigned to the other sectors.

Abstract: During a group call, each remote unit (113–118) receives a downlink supplemental transmission (109–110) that contains high-speed data (e.g., video). A single supplemental channel is utilized for each base station (101–102). On the uplink side, the supplemental channel is shared among all remote units, however only one remote unit (113–118) is allowed to transmit at a time utilizing the uplink supplemental channel. A decision is made as to which remote unit is providing a highest-voice-energy fundamental channel uplink, and that remote unit is assigned the uplink supplemental channel. The remote unit, that is currently transmitting utilizing the uplink supplemental channel, has its transmission broadcast (via downlink supplemental channels (109–110)) to all other remote units participating in the multicast.

Abstract: A convolutional encoder (112) comprises a controller (201), having a transmission rate (e.g. frame rate) as an input. The controller (201) initializes the encoder (112) to an initial state based on a transmission rate currently being utilized. When decoding, a decoder (300) utilizes a Trellis diagram having an initial and final state based upon the frame rate.

Abstract: An apparatus (300) for and method of synchronizing OFDM signals utilizes a single baud to provide synchronization in time, frequency, and per-subcarrier rotation (201). Timing and fractional subcarrier frequency synchronization may be obtained from either a known or unknown (e.g., data symbol) baud having known symmetry properties. Because all three synchronization tasks may be accomplished utilizing a single sync baud, the present invention is spectrally efficient. A differential correlation metric is utilized to efficiently provide integer subcarrier frequency synchronization and per-subcarrier rotation synchronization.

Abstract: A method of processing an information sequence with a decoder is provided. A window within the information sequence is selected. A training period is calculated for the window. At least one recursion of the window is initialized based on the calculated training period. Systems and programs for processing the information sequence are also provided.

Abstract: When polled, a remote unit (113) provides a radio access network (RAN 101) with an acknowledgment message. The acknowledgment message contains information such as the current window size, the number and identification of frames received in error, . . . etc. The polling frequency is based on a channel condition metric. More particularly, active polling timers that control the transmission of status information over the air interface are adjusted based on a bit error rate (BER) of radio channel such that as the BER of the radio channel decreases less control information is transmitted. Similarly as BER increases the timer values will change such that more control information is transmitted.

Abstract: A method of selecting one or more active slave wireless devices from a plurality of active/parked slave wireless devices is implemented by a master wireless device. The master wireless device initially assigns sequential weights to each active/parked slave wireless device, and subsequently designates one or more of the active/parked slave wireless devices as active slave wireless device(s) based partially or entirely on the assigned sequential weights. The assignment of sequential weights can be accomplished by assigning each active/parked slave wireless device to one of a plurality of ranked groups collectively representative of various traffic types between a base site and the active/parked slave wireless devices. Each active/parked slave wireless device is then sorted within its respective group.

Abstract: In a wireless code-division multiple access (CDMA) system (100), a talkgroup (101) of subscriber units is provided. A sub-talkgroup (102) of subscriber units, forming a part of the talkgroup is assigned at least one inbound channel (416-417). The talkgroup is assigned outbound channels (415). Members of the sub-talkgroup may simultaneously transmit voice information (410-411) using the at least one inbound channel, which voice information is summed (412) and re-transmitted to the talkgroup using the outbound channels. Voice information is summed so that an individual talker receives summed voice information without the individual subscriber's voice content. Subscriber units in the sub-listengroup are allowed to transmit voice information only after requesting, and receiving, an additional inbound channel.