157 results about "Full Rate" patented technology

Methods and systems are provided for allocating resources including VoIP (voice over Internet Protocol) and Non-VoIP resources. In some embodiments, multiplexing schemes are provided for use with OFDMA (orthogonal frequency division multiplexing access) systems, for example for use in transmitting VoIP traffic, in some embodiments, various HARQ (Hybrid Automatic request) techniques are provided for use with OFDMA systems. In various embodiments, there are provided methods and systems for dealing with issuea such as Handling non-full rate vocoder frames, VoIP packet jitter handling, VoIP capacity increasing schemes, persistent and non-persistent assignment of resources in OFDMA systems.

A forward link device of a multicarrier CDMA communication system with an overlay scheme in which a multicarrier system and an IS-95 system share same frequency bands. In the forward link device, four encoders encode input data of corresponding rates with a 1/3 coding rate, respectively. A first repeater repeats two times full rate symbols output from the first encoder, and second to fourth repeaters repeat symbols output from the second to fourth encoders, respectively, according to a predetermined number of times, to match the number of corresponding output symbols to the number of full rate symbols. First to fourth interleavers interleave the symbols output from the first to fourth repeaters, respectively, to uniformly distribute the symbols of the same data bit to the carriers.

A data transmission apparatus and method using a full-diversity, full-rate STBC are provided. In the data transmission apparatus, a serial-to-parallel converter converts an input bit stream to parallel binary vectors. A bit/symbol mapper generates modulator input symbols by combining the bits of the binary vectors. A modulator modulates the modulator input symbols to complex symbols. A transmit matrix block encoder encodes the complex symbols using a transmit matrix and transmits the coded symbols through corresponding transmit antennas.

A system and method for data communication over a cellular communications network that allows the transmission of digital data over a voice channel using a vocoder that operates in different modes depending upon characteristics of the inputted signal it receives. To prepare the digital data for transmission, one or more carrier signals are encoded with the digital data using one of a number of modulation schemes that utilize differential phase shift keying to give the modulated carrier signal certain periodicity and energy characteristics that allow it to be transmitted by the vocoder at full rate. The modulation schemes include DPSK using either a single or multiple frequency carriers, combined FSK-DPSK modulation, combined ASK-DPSK, as well as PSK with a phase tracker in the demodulator. These modulation schemes permit data communication via a CDMA, GSM, or other type of voice traffic channel at a low bit error rate.

Parallel adaptive filters and filtering methods that enable processing of an input signal in a circuit that has an clock speed many times slower than the input rate of the input signal that is processed. A polyphase decimator structure processes a data stream requiring a low pass filtered bandlimited (low-rate) output that is used for high-rate output structures. The filters and methods break an input data stream into parallel paths that efficiently produce a bandlimited (decimated, low-rate) filtered output. Each of the parallel paths is processed at a decimated rate to provide a filtered output signals corresponding to a filtered version of the input signal.

A 1/n-rate decision feedback equalizer (DFE) and method include a plurality of branches. Each branch includes a summer circuit configured to add a feedback signal to a received input, and a latch configured to receive an output of the summer circuit in accordance with a clock signal. A feedback circuit includes a multiplexer configured to receive as input, an output of each branch, the multiplexer having a clocked select input and configured to multiplex the output of each branch to assemble a full rate bit sequence, and a filter configured to provide cancellation of intersymbol interference (ISI) from the received input to be provided to the summer circuit of each branch.

A space-time code has a codeword matrix. The codeword matrix is a square matrix with dimension equal to the number of the transmit antennas, wherein a row of the codeword matrix represents combined signals transmitted by each transmit antenna and a column of the codeword matrix represents timeslots of the number of the transmit antennas. Signals are transmitted through the rows of the number of spatial multiplexing rate, in each column of the matrix. Further, the symbols of the number of the transmit antennas are combined by way of utilizing complex weights, in each row of the matrix. With this space-time code, a delay is minimized, so that change of channels can be managed efficiently. Also, the space-time code has minimum number of data symbols, and thus its complexity is minimized.

There is disclosed a system and method for sending low rate data on a packet basis in an 8-VSB standard data packet stream. The system of the invention comprises an 8-VSB signal transmitter that is capable of transmitting either standard full rate data packets or low rate data packets. A low rate data packet contains less data than is normally contained in a standard data full rate data packet. Each data byte in a low rate data packet contains information bearing bits and non-information bearing bits. The system is capable of assigning appropriate bit values to the non-information bearing bits in each data byte of a low rate data packet so that each low rate data packet will be properly encoded for transmission within an 8-VSB standard data packet stream. High definition television signals composed of low rate data packets possess increased resistance to noise and multipath channels.

A system and method for data communication over a cellular communications network that allows the transmission of digital data over a voice channel using a vocoder that operates in different modes depending upon characteristics of the inputted signal it receives. To prepare the digital data for transmission, one or more carrier signals are encoded with the digital data using one of a number of modulation schemes that utilize differential phase shift keying to give the modulated carrier signal certain periodicity and energy characteristics that allow it to be transmitted by the vocoder at full rate. The modulation schemes include DPSK using either a single or multiple frequency carriers, combined FSK-DPSK modulation, combined ASK-DPSK, PSK with a phase tracker in the demodulator, as well as continuous signal modulation (ASK or FSK) with inserted discontinuities that can be independent of the digital data.

There is provided rate selection methods and systems for selecting coding rates for coding frames of a speech signal to realize an average bit rate indicated by a mode. For example, a mode 0, mode 1, and a mode 2 may be defined, with each mode requiring a different average bit rate. To achieve the average bit rate of a particular mode, a coding rate is selected for each frame of the speech signal, based on the characteristics of a frame. A frame can be categorized in a class, such as noise or silence, noise-like unvoiced speech, pulse-like unvoiced speech, transition into voiced speech, unstable voiced speech, stable voiced speech. Other parameters may also be used, such as the sharpness, noise-to-signal ratio, pitch correlation, energy, and reflection coefficient. A frame may then be coded at a full-rate, a half-rate, a quarter-rate, or an eighth-rate.

A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codec are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech. The overall quality of the system is strongly related to the excitation. In order to enhance the excitation, the system contains a fixed codebook comprising several subcodebooks. The invention reveals a way to apply a pitch enhancement efficiently and differently for different subcodebooks without using additional bits. The technique is particularly applicable to selectable mode vocoder (SMV) systems.

An Ethernet network may comprise multi-rate link partners coupled via an Ethernet link wherein a pattern, over time, of offered data rates may be determined. Link partners may establish a communication policy for modifying transmission rates to mimic offered data rates and may vary rates accordingly. The communication control policy may be communicated between nodes via one or more of physical layer control characters, an out of band channel and higher layer protocol packets. An established communication policy may improve energy efficiency and may be dynamically modified, for example, based on link utilization. In this regard, data rates may be symmetrical or asymmetrical with respect to direction of transmission. Moreover, data rates may comprise the offered data rate plus additional bandwidth for head room and may comprise one or more of a full rate, an intermediate rate and/or a rate of zero. Rate transitions may be synchronized among network nodes.

The embodiment of the invention provides methods and devices for detecting and repairing an error voice frame, and relates to the field of communication. The error frame can be detected and repaired according to the characteristics of a voice signal and the prior statistic characteristics of a coding parameter. The detection method comprises the following steps of: in a silent mode, receiving a voice frame which is indicated to be correct by the parameter, detecting the parameter of the voice frame according to a preset detection rule, and determining that the voice frame is the error voice frame when the parameter meets the condition specified by the detection rule; and in a voice mode, receiving a voice frame which is indicated to be wrong by the parameter, detecting a silence insertiondescriptor (SID) of the voice frame according to the parameter average value of the voice frame, and determining that the voice frame is an SID frame when the SID meets the detection condition. The embodiment of the invention can be used for the full rate voice coding and decoding of a global system for mobile communications (GSM).

The invention provides a device and a method for querying subway passenger distribution and a full rate in real time. The device comprises a camera arranged at the top of the middle of a carriage, an embedded main control unit, a screen in the carriage and a screen in a subway station, and the camera is connected with the embedded main control unit; the embedded main control unit is directly connected with the screen in the carriage and displays information on the screen in the carriage through a communication network; the passenger distribution condition and the congestion degree of the current subway carriage can be queried in real time; the device provided by the invention can not only query the passenger distribution condition and the congestion degree of the current subway carriage, but also can query the historical passenger distribution condition and the congestion degree; and the device can not only query before on board and can also query in the carriage.

A full-rate distributed space-time (ST) code design is provided for amplify-and-forward cooperative wireless channels. A signal space diversity technique is employed at the source node and a unique signature vector at each relay node. The distributed space-time (ST) codes can achieve full cooperative diversity and full rate. The achievable diversity gain is M+1, where M is the number of relay nodes. Optimal power allocation can be used to maximize the coding gain under a total power constraint.

A mobile communication system using an STBC scheme having an even number of Tx antennas is provided. In a transmitter having an even number of Tx antennas, a pre-coder pre-codes an input symbol sequence using a pre-coding matrix. The pre-coding matrix is a matrix produced by puncturing a unitary matrix. A space-time coder space-time-encodes the pre-coded symbol sequence received from the pre-coder using a coding matrix.

Automatic rate negotiation logic for a high speed serial interface in a programmable logic device determines whether multiple occurrences of a single-bit transition (i.e., a data transition from “0” to “1” to “0” or from “1” to “0” to “1”) occur within a predetermined time interval on a data channel of a high-speed serial interface. The interval preferably is selected such that multiple occurrences of a single-bit transition mean that the data channel is operating in full-rate mode. The rate negotiation logic may share a phase detector with clock data recovery circuitry in the interface. The phase detector may be a bang-bang phase detector specially adapted to detect single-bit transitions.

A method and apparatus are disclosed for increasing the time and frequency diversity of a multi-stream signal in a DAB system. A plurality of audio streams are divided into four (4) digital sub-streams, C₀₀, C₀₁, C₁₀, and C₁₁. Each sub-stream C₀₀, C₀₁, C₁₀, and C₁₁ is assigned a unique frequency band, and time slot. A first core sub-stream C₁₀ is mapped to one frequency partition and a second core sub-stream C₀₀ is mapped to another frequency partition and delayed relative to the first core sub-stream. Similarly, two enhancement sub-streams C₁₁ and C₀₁ are mapped to different frequency partitions and are time delayed relative to each other and the core sub-streams. The two core sub-streams C₀₀ and C₁₀ can have a maximum separation across both the time and frequency axes. Each core sub-stream C₀₀ and C₁₀ is separate from one of the enhancement sub-streams in the frequency domain and separate in the time domain from the other enhancement sub-stream. Each core sub-stream C₀₀ and C₁₀ can be combined with any other available core or enhancement sub-stream to form a 64 kbps PAC. In addition, a 96 kbps PAC can be obtained by combining the two core sub-streams C₀₀ and C₁₀ with one of the enhancement sub-streams C₀₀ or C₁₁. Finally, the combination of all four sub-streams produces a full-rate 128 kbps PAC.