543 results about "Signal coding" patented technology

Exploiting the substantive reciprocity of internode channel responses through dynamic, adaptive modification of receive and transmit weights, enables locally enabled global optimization of a multipoint, wireless electromagnetic communications network of communication nodes. Each diversity-channel-capable node uses computationally efficient exploitation of pilot tone data and diversity-adaptive signal processing of the weightings and the signal to further convey optimization and channel information which promote local and thereby network-global efficiency. The preferred embodiment performs complex digital signal manipulation that includes a linear combining and linear distribution of the transmit and receive weights, the generation of piloting signals containing origination and destination node information, as well as interference-avoiding pseudorandom delay timing, and both symbol and multitione encoding, to gain the benefit of substantive orthogonality at the physical level without requiring actual substantive orthogonality at the physical level.

Apparatus is described for digitally encoding an input audio signal for storage or transmission. A distinguishing parameter is measure from the input signal. It is determined from the measured distinguishing parameter whether the input signal contains an audio signal of a first type or a second type. First and second coders are provided for digitally encoding the input signal using first and second coding methods respectively and a switching arrangement directs, at any particular time, the generation of an output signal by encoding the input signal using either the first or second coders according to whether the input signal contains an audio signal of the first type or the second type at that time. A method for adaptively switching between transform audio coder and CELP coder, is presented. In a preferred embodiment, the method makes use of the superior performance of CELP coders for speech signal coding, while enjoying the benefits of transform coder for other audio signals. The combined coder is designed to handle both speech and music and achieve an improved quality.

A system and method are disclosed for the management of WLANs in cases where unmanaged access points are present as well as with the addition or removal of access points. The disclosed system and method use signal data and network traffic statistics collected by mobile units to determine optimal configuration settings for the access points. The access point settings so managed can include the operating channel or center frequency, orthogonal signal coding used (optionally including the data rate), if any, and the transmission power. The solutions computed can account for the inherent trade-offs between wireless network coverage area and mutual interference that may arises when two or more access points use the same or overlapping frequency bands or channels and the same or similar signal coding.

A compression system and process include adaptive quantization for selection of one quantizer, for each video frame or frame portion, from a group of quantizers, where each quantizer is a set of predefined quantizer values, and coding of the quantized values using a common entropy coding table for all quantizers, or a separate entropy coding table for each quantizer. The entropy coding tables are obtained during a training procedure. The selection of the quantizer is based on a formula which takes into account the distortion and bit rate characteristics of each quantizer. A similar formula, based on the distortion and bit rate characteristics, is used to select the particular quantizer value within the chosen quantizer for each video signal value being coded.

A method of transmitting picture information of a video signal from an encoder and a method of using the picture information in a decoder are provided. When a video signal is encoded, the video signal is coded according to a specified scheme while being divided into key and non-key pictures, and a value indicating whether or not coded picture data carried in each NAL unit is key picture data is recorded in a ‘nal_ref_idc’ field in a header of the NAL unit or, alternatively, a value (adaptive_ref_pic_marking_mode_flag=1) indicating that a Memory Management Control Operation (MMCO) is present and a control operation value indicating a key picture are recorded in a header of a picture coded into a key picture.

Provided are a depth image obtaining section that obtains a depth image from the same viewpoint as in the input image, a layer section that separates the input image into a foreground image and a background image as layered images using depth information of the depth image, a coding section that encodes the foreground image, a background sprite generating section that generates a background sprite image from the background image, and a sprite coding section that encodes the background sprite image. The depth image from the same viewpoint as in the input image is obtained in the depth image obtaining section, the input image is separated into a foreground image and a background image as layered images using the depth information, and based on the separated background image, a background sprite image is generated.

A 3-D wireframe model expressed in terms of nodes and vertices receives a 3-D video or like signal representative of a scene expressed in terms of a reference model, I frames and P frames. A DPCM coder takes advantage of the temporal correlation of the displacement of each vertex along every axis in the 3-D space. The 3-D signal is a set of non-zero displacements of all vertices and all nodes (s_l[n, v]) at time t_l. The decoded set (animation frame) of the previous instance is used as the predicted value (s_i-l[n, v]). The prediction error e_l[n, v], i.e. the difference between the current displacement set and the predicted one, is computed and quantised (e_l[n, v]). Finally, the quantised samples are entropy coded (c_i[n, v]) using an adaptive arithmetic coding algorithm to handle the unknown data statistics. The predictive scheme described above prevents quantization error accumulation. A DPCM decoder first decodes arithmetically the received samples (‘e’ [n, v]) and computers the decoded samples (s_i′ [n, v]).

A digital acoustic signal coding apparatus, a method of coding the digital acoustic signal, and a recording medium for recording a program of coding the digital acoustic signal are respectively realized. It is possible to provide the digital acoustic signal coding method and apparatus, in which, corresponding to the difference between the sampling frequencies of the input acoustic signal, short blocks can be suitably classified into groups without deteriorating sound quality and the suitability of using either long/short blocks can be judged. The coding apparatus is composed of a calculation medium for calculating the sensation entropy of an input acoustic signal per each of the respective short sensation blocks; a sensation entropy sum total calculation medium for obtaining a total sum in a frame of the sensation entropy; a comparison medium for comparing an absolute value of the difference between the respective total sums of the sensation entropy of successive two frames with a previously determined threshold value; and a long/short block judgment medium for judging whether a long block or short blocks should be used to convert a block of the input acoustic signal on the basis of the comparison result.

The invention discloses a process, a method and a mobile phone utilizing a mobile phone sharing television screen to control a television. The method comprises the following steps: the mobile phone with the screen sharing program is connected with the television through a wireless network; the current screen picture of the television is sent to the mobile phone end and scaled down for being displayed on the mobile phone screen through the wireless network; when the mobile phone end triggers a keystroke, ID coding information and keystroke coding information are generated through the signal coding module at the mobile phone end; and after the information is sent to the television for decoding, an corresponding keystroke event is triggered at the television end. The invention adopts the mobile phone sharing television screen and utilizes the mobile phone touch screen and the keystrokes to control the television, a corresponding position on the mobile phone screen is touched, the corresponding position touch event on the television screen can be correspondingly triggered, and the mobile phone keystrokes can also be used to trigger corresponding television keystroke event; and the mobile phone can also be used for performing menu selection and character input on the television so as to realize the purpose of utilizing the mobile phone to operate the television. Therefore, the television remote control function is added to the mobile phone.

The time required to transcode an encoded image signal to an image signal encoded by a different encoding method is reduced.

Disclosed herein is a transcoder comprising: a picture selector 4 which extracts specific types of pictures (I- and P- pictures) from a first image signal (MPEG2 stream) and generates a subset of the first image signal; a first decoder 5 which decodes the subset image signal; and a first encoder 7 which encodes the decoded image signal to a second image signal (MPEG4 stream). The picture selector 4 uses the extracted pictures to generate the subset image signal with a reduced effective length.

The present invention relates to a method that uses a motion vector of a predictive video frame of a sub-layer to encode a video signal and decode encoded video data. The method encodes a video signal using a preset method to a bit stream of a base layer while encoding the video signal using a scalable MCTF method to a bit stream of an enhanced layer. When an arbitrary frame of the video signal is encoded, information, enabling at least one vector, derived from a first motion vector of a first block included in the bit stream of the base layer in the same direction as the first motion vector, to be used as a motion vector of an image block in the arbitrary frame, is recorded in the bit stream of the enhanced layer. The first motion vector is directed in the same direction as a temporal direction from the arbitrary frame to the first block.

A method and system for encoding and decoding an input signal in relation to the most perceptually relevant aspects of the input signal. A two-dimensional (2D) transform is applied to the input signal to produce a magnitude matrix and a phase matrix that can be inverse quantized by a decoder. A first column of coefficients of the magnitude matrix represents a mean spectral density (MSD) function of the input signal. Relevant aspects of the MSD function are encoded at a beginning of a data packet. The MSD function is also processed through a core perception model to determine bit allocation. The matrices are then quantized and priority ordered into a data packet, with the least perceptually relevant information at the end of the packet so that it may be ignored or truncated for scalability to the channel data rate capacity.

Video signal coding apparatus includes: a quantization unit which quantizes, using coefficient values in a first quantization matrix, a luminance signal of each of a first picture and a second picture coded after the first picture, and quantizes, using coefficient values in a second quantization matrix, a chrominance signal of each of the first picture and the second picture; a code amount obtainment unit which obtains a luminance code amount and a chrominance code amount of each of the first picture and the second picture; a rate calculation unit which calculates a rate of the chrominance code amount with respect to the luminance code amount for each of the first picture and the second picture, obtained by the code amount obtainment unit; and a rate control unit which changes a coefficient value of at least one of the first matrix and the second matrix used in quantizing the second picture, before the quantization unit quantizes the second picture, so that the rate for the second picture becomes smaller than the rate for the first picture.

The invention provides an all-digital satellite signal simulated source. The all-digital satellite signal simulated source comprises a double-port storage, the double-port storage is a FPGA which provides interfaces for various rate conversions between coding and modulation, is connected with a digital to analog converter through a high-speed serial interface to form a core hardware framework of the satellite signal simulated source, and is connected with a broadband filter directly through a simulation output end to produce multimode high and moderate frequency broadband satellite signal simulated signals with code rates which can vary continuously, a digital signal processor serves as a main control device of the simulated source and performs analysis of various parameters issued by a monitor and parameter configuration of the FPGA, a 32-channel parallel modulation system is arranged inside the FPGA, for example, a 32-channel direct digital synthesizer (DDS) performs parallel output and composition of the central frequency, the FPGA determines the overturn of the carrier phase according to a symbol rate and a main frequency of parallel calculation, IQ two-channel signals, signal coding, code conversion and quadrature modulation are produced through a logical algorithm, a Q channel uniformization coefficient is calculated according to an IQ unbalanced parameter, Q channel uniformization coefficient and the Q channel signal amplitude are subjected to multiply-add arithmetic to calculate the Q channel amplitude, IQ is added, and a unbalanced quadrature phase shift keying (UQPSK) modulation method is simulated.

The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (410) that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit (404) that codes the first downmix signal; a first t-f converting unit (401) that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit (409) that generates spatial information for generating a multi-channel audio signal from a downmix signal.

The invention relates to a synchronizer and a synchronization method for quantum key distribution. The synchronizer is characterized in that the signal output end of a synchronous signal coding module is connected with the signal input end of a synchronous laser transmitting module the signal output end of which is connected with the signal input end of a comparator circuit module through optical fibers; and the signal output ends of an optical detector and the comparator circuit module are respectively connected with the signal input ends of an adjustable time delay circuit which is used to generate door control signals and has adjustable time delay and a synchronous detection and coding module which counts by using one frame counter and codes the synchronous signals. The invention further provides a high-performance quantum key distribution system which uses single optical fiber to transmit signal light and synchronous light in a system with frequency of dozens of megahertz, realizes accurate synchronization of both systems, can realize synchronization again in the case of loss of the synchronization and improves the fault tolerance of the systems.

In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.

The invention relates to one mobile digital multi-media broadcast signal channel coding and knitting method and device, wherein, the device comprises the following parts: LDPC coding device and inner knitting device by use of bit and outer coding and knitting device for data flow input on RS(240, M) for short codes and outer coding; then sending the outer codes and knitting data into LDPC coder, wherein, the M is for information bit length, then the outer coder and knit device for LDPC knitting to fulfill signal coding and knit process.