子分类：
| 序号 | 专利名 | 申请号 | 申请日 | 公开（公告）号 | 公开（公告）日 | 发明人 |
|---|---|---|---|---|---|---|
| 161 | Information transmission method, mobile communications system, base station and mobile station | US11311272 | 2005-12-20 | US07593369B2 | 2009-09-22 | Michiaki Takano |
| In an information transmission method, a radio communications system, a base station and a mobile station, a TBS size, a modulation scheme and the number of codes in a multicode are converted into identification data having a relatively smaller data size before being transmitted to a destination of communication. The TBS size is identified by using, in combination, an identification code identifying a channelization code set, an identification code identifying a modulation scheme, and an identification code obtained by converting a combination of the number of codes in a multicode and a modulation pattern identification (TFRC) into a corresponding code. Accordingly, the data size for TBS size identification is reduced. | ||||||
| 162 | Parallelized dynamic Huffman decoder | US11873328 | 2007-10-16 | US07564379B2 | 2009-07-21 | Michael D. Ruehle |
| Several code detectors in parallel simultaneously examine varying overlapping segments of a data stream containing variable length codes, referred to as a data window. The data window segments directly address memory structures within each of the code detectors without any previous logic stages. Each code detector is responsible for a range of code lengths, and ignores data window bits that are not relevant to its code length range. Each code detector outputs a possible result to a layer of logic that selects the possible result of the single code detector which contains result data corresponding to a variable length code in the data window. | ||||||
| 163 | Compression rate control system and method with variable subband processing | US11232165 | 2005-09-20 | US07525463B2 | 2009-04-28 | Steven E. Saunders; William C. Lynch; Krasimir D. Kolarov |
| A system, method and computer program product provide finer rate control in data compression by processing a data stream through a plurality of parallel subbands, wherein a first subband processes the data differently than a second subband. Separate shift quantization parameters for each separate run-of-zeros compressed storage area or pile can be provided, instead of a single common shift parameter for every coefficient as in the prior art. The parameter value for each such area or pile can be recorded in the compressed output file. The separate shift quantization parameters can also be adjusted dynamically as data is being compressed. | ||||||
| 164 | Method and system of compressing and decompressing data | US11654827 | 2007-01-18 | US20080175312A1 | 2008-07-24 | Qin Zhang |
| The present invention relates to a data compression and decompression system and method for lossless compressing digital data. In one preferred embodiment, the method for handling a data stream having a number of data objects comprising a step of performing a compressing process on a data object based on a compression based value to obtain at least one compressed data result, wherein the data object is considered as one numerical value. The device for handling the data stream comprising a compression means for compressing the data objects according to a compression base value. In another preferred embodiment, the method for handling data having a number of data objects comprising a step of performing a compressing process on a data object by obtaining a compression code for the data object according to a compression coding table, wherein the data object is considered as one numerical value. The device for handling a data stream comprising a compression means for compressing the data objects according to a compression coding table. | ||||||
| 165 | Coding device and method, and recording medium | US10145831 | 2002-05-14 | US07330555B2 | 2008-02-12 | Shiro Suzuki |
| An L-channel signal and an R-channel signal are input into a subtractor in which the difference between the L-channel signal and the R-channel signal is determined. The resulting signal is input into first and second multipliers. In the first multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a previous frame. In the second multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a current frame. The mixed output signals from the first and second multipliers respectively undergo spectrum transform in first and second spectrum transformers, and quantized in first and second quantizers. A comparator compares a quantization error in the first quantizer with that in the second quantizer. If the quantization error in the second quantizer is greater than that in the first quantizer by a predetermined factor, the comparator controls a switch to select the output of the first quantizer. | ||||||
| 166 | Preparing electronic data for transmission | US10629886 | 2003-07-29 | US07305490B2 | 2007-12-04 | Kristofer Erik Metz; Nolan W. Letellier |
| A method for preparing electronic data for transmission includes calculating a duration for compressing the electronic data, calculating a duration for transmitting the electronic data if not compressed, and compressing the electronic data only if the duration for compressing does not exceed the duration for transmitting. | ||||||
| 167 | System and method for minimising bandwidth utilisation in a wireless interactive voice response system | US10475673 | 2002-04-17 | US07277696B2 | 2007-10-02 | Bruno Richard Preiss; Daniel Wilk |
| The present invention provides a novel system, method and apparatus of delivering interactive voice response services in a more efficient manner over a network. The system provides for the placement of a subscriber station local 28 to the subscriber and a gateway protocol converter at the base station 40. The gateway protocol converter converts bandwidth-intensive audio messages into compact data messages, which upon transmission to the subscriber station 28 are converted back into audio messages. By using data messages instead of audio messages over the network, bandwidth is saved on the network for other traffic. Another embodiment of the invention provides a method for transmitting subscriber responses utilizing the system. In another embodiment of the invention, there is provided an IVR system that utilizes the VoiceXML standard, or the like, whereby the Document Server, VoiceXML Interpreter Context, and the Implementation Platform are distributed at various locations along the communication link between the IVR service provider and the subscriber station—such components being distributed along the link according to a desired utilization of network resources, such as bandwidth. | ||||||
| 168 | EFFICIENT SYSTEMS AND METHODS FOR TRANSMITTING COMPRESSED VIDEO DATA HAVING DIFFERENT RESOLUTIONS | US11683914 | 2007-03-08 | US20070147804A1 | 2007-06-28 | Ji Zhang; Humphrey Liu |
| The present invention relates to systems and methods for combining compressed video data encoded or received with different resolutions. The present invention combines the compressed video data from the two separate bitstreams without decoding and re-encoding each bitstream. To do so, the present invention determines which compressed video bitstream has video data with a lower resolution, and applies a tiling process that alters the low resolution compressed video data such that it may be displayed at a high resolution. | ||||||
| 169 | Audio signal processing apparatus | US10721167 | 2003-11-26 | US07236933B2 | 2007-06-26 | Norihiko Fuchigami; Shoji Ueno; Yoshiaki Tanaka |
| In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals. A plurality of second predictors have different prediction characteristics respectively, and are responsive to the second difference signal for generating second different prediction signals for the second difference signal, respectively. A plurality of second subtracters operate for generating second prediction-error signals representing differences between the second difference signal and the second different prediction signals, respectively. A second minimum prediction-error signal representative of a smallest difference is selected from among the second prediction-error signals. | ||||||
| 170 | Method for replacing corrupted audio data | US10497187 | 2001-11-30 | US07206986B2 | 2007-04-17 | Jan Stemerdink; Arjan Meijerink |
| A decoding method for coded data representing original data. Corrupted data is detected and replaced with buffered data. The buffered data is stored in the buffer a time interval corresponding to an estimated periodicity or an integer multiple thereof before the corrupted data was received. The estimated periodicity is determined by estimating the periodicity of the original data represented by the corrupted data. | ||||||
| 171 | Audio enhancement communication techniques | US10075145 | 2002-02-14 | US07158572B2 | 2007-01-02 | Bruce E. Dunne; Ravi Chandran; Daniel J. Marchok |
| A communication system (10) receives a communication signal comprising first and second data with different compression levels, such as highly compressed and weakly compressed levels. A mode detector (15) detects the level of compression. One or more signal decoders (20, 22) decode the highly compressed data. An analyzer (30) determines the type of enhancement required. One or more processors (48, 50, 80) enhance the data as required. An encoder (60) reencodes the enhanced decoded data. Metrics (90) may aid the operation of the analyzer (30).The communication system may include telephones (120, 122, 124, 126). Processors (103, 104) enhance signals in opposite first and second directions between pairs of the telephones. A path (106) connects the processors in tandem. One or more switches (101, 102) disable signal enhancement for one of the processors depending on the compression level of the signals to avoid degrading call quality. | ||||||
| 172 | Methods of digital filtering and multi-dimensional data compression using the farey quadrature and arithmetic, fan, and modular wavelets | US09869640 | 1999-12-21 | US07158569B1 | 2007-01-02 | Robert C. Penner |
| Methods are presented for calculating the wavelet filters and their inverses which rely on a new method of sampling (UA, TA, A) either digital or analog data. These methods combine and extend to give novel procedures for non-reversible multi-dimensional data compression. For selected applications, this procedure improves achievable compression factors by an estimated one to three orders of magnitude and is well-suited to picture build-up or other iterative refinement. Combining these wavelet filters and their inverse with previous theoretical work furthermore provides novel methods for calculating Fourier and other transforms. In a preferred embodiment used to calculate the Fourier transform (1) and its inverse (34) applied to digital input date, the method replaces the fast Fourier transform and its inverse and provides improvements in achievable accuracy. The new sampling method is inherently multiscale, and the invention thereby obviates the usual Nyquist constraint on the meaningful bandwidth in terms of the number of samples. Finally, the invention provides novel and efficient analog-to-digital and digital-to-analog interface. | ||||||
| 173 | Distortion-based method and apparatus for buffer control in a communication system | US11403530 | 2006-04-13 | US20060184358A1 | 2006-08-17 | Christof Faller |
| Methods and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. The distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value. | ||||||
| 174 | Audio signal processing apparatus | US10721218 | 2003-11-26 | US07092889B2 | 2006-08-15 | Norihiko Fuchigami; Shoji Ueno; Yoshiaki Tanaka |
| In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals. A plurality of second predictors have different prediction characteristics respectively, and are responsive to the second difference signal for generating second different prediction signals for the second difference signal, respectively. A plurality of second subtracters operate for generating second prediction-error signals representing differences between the second difference signal and the second different prediction signals, respectively. A second minimum prediction-error signal representative of a smallest difference is selected from among the second prediction-error signals. | ||||||
| 175 | Method for reducing bit rate requirements for encoding multimedia data | US11053362 | 2005-02-08 | US20060176961A1 | 2006-08-10 | Yan Huo; Oscar Au; Zhibin Lei |
| Some representative embodiments are directed to systems and methods for compressing a data set. In one embodiment, a method comprises receiving a frame of data to be encoded, generating a residual frame that represents a difference between the received frame and one or several reference frames, performing a respective sum of absolute differences (SAD) calculation for each block within the residual frame, and applying a transform function to each data value within the residual frame, wherein the transform function is at least a function of a SAD value calculated for the block containing the respective data value. | ||||||
| 176 | Programmable quantization dead zone and threshold for standard-based H.264 and/or VC1 video encoding | US11010029 | 2004-12-10 | US20060126724A1 | 2006-06-15 | Guy Cote; Elliot Linzer; Lowell Winger |
| A video encoder comprising an encoder circuit, a quantizer circuit and a control circuit. The encoder circuit may be configured to generate a number of coefficient values in response to a video stream and a number of quantized values. The quantizer circuit may be configured to generate the number of quantized values in response to the coefficient values, two or more quantization dead zones and two or more offsets. The control circuit may be configured to set the two or more quantization dead zones and the two or more offsets to different values. The two or more quantization dead zones and the two or more offsets are independently programmable. | ||||||
| 177 | Information transmission method, mobile communications system, base station and mobile station in which data size of identification data is reduced | US11124270 | 2005-05-09 | US07050413B2 | 2006-05-23 | Michiaki Takano |
| In an information transmission method, a radio communications system, a base station and a mobile station, a TBS size, a modulation scheme and the number of codes in a multicode are converted into identification data having a relatively smaller data size before being transmitted to a destination of communication. The TBS size is identified by using, in combination, an identification code identifying a channelization code set, an identification code identifying a modulation scheme, and an identification code obtained by converting a combination of the number of codes in a multicode and a modulation pattern identification (TFRC) into a corresponding code. Accordingly, the data size for TBS size identification is reduced. | ||||||
| 178 | METHOD AND APPARATUS FOR SYNCHRONIZED RECORDING OF AUDIO AND VIDEO STREAMS | US10906457 | 2005-02-21 | US20060104344A1 | 2006-05-18 | Cheng-Che Chen |
| A method for synchronized recording of audio and video signals is provided. The video signals are grouped into a plurality of video frames. The method comprises the steps of incrementing a counter value of an audio counter when an audio signal is received; and recording the current counter value of the audio counter when a video frame is received. | ||||||
| 179 | Information transmission method, mobile communication system, base station, and mobile station | US11311160 | 2005-12-20 | US20060098601A1 | 2006-05-11 | Michiaki Takano |
| In an information transmission method, a radio communications system, a base station and a mobile station, a TBS size, a modulation scheme and the number of codes in a multicode are converted into identification data having a relatively smaller data size before being transmitted to a destination of communication. The TBS size is identified by using, in combination, an identification code identifying a channelization code set, an identification code identifying a modulation scheme, and an identification code obtained by converting a combination of the number of codes in a multicode and a modulation pattern identification (TFRC) into a corresponding code. Accordingly, the data size for TBS size identification is reduced. | ||||||
| 180 | Low power scheduling for multimedia systems | US09895048 | 2001-06-29 | US07043557B2 | 2006-05-09 | Malena Rosa Mesarina; Yoshio Frank Turner |
| A method and system thereof for reducing the energy consumed when decoding an encoded and synchronized multimedia data stream, wherein the data stream is non-preemptable and subject to precedence constraints. In a client-server environment, the server delivers to the client the stream for decoding. The client has a processor operating on a discrete variable-voltage power supply. Prior to transmitting the stream to the client, the server produces an execution schedule according to the precedence constraints. The server also assigns a voltage setting for each task in the schedule, wherein each task decodes a frame in the stream without preemption. The server transmits the execution schedule and voltage settings to the client with the encoded data stream. The schedule and voltage settings reduce energy consumption by the client while satisfying multimedia timing constraints. | ||||||
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