Abstract: A novel technique for combining deinterleaving operation with Fast Fourier Transformer (FFT) modules and other post FFT modules in a receiver to reduce processing time. In one example embodiment, the deinterleaving operation, in the post FFT module, is combined with FFT and demapper operations to reduce the processing time and complexity.

Abstract: A synchronizer and a method for synchronizing a communication signal are presented. The synchronizer comprises a first unit arranged for receiving a plurality of inputs. The plurality of inputs include at least an indicative of a source clock period of a source clock domain, an indicative of destination clock period of a destination clock domain and a communication signal. The first unit being configured for stretching the communication signal according to the indicative of the destination clock period and indicative of the source clock period. The synchronizer is further provided with a second unit. The second unit is configured for operating according to the indicative of destination clock period. The output of the first unit is provided to the second unit. The second unit is configured for providing a synchronizer output signal corresponding to the communication signal operable in the destination clock domain.

Abstract: A technique for improving the performance of OFDM (Orthogonal Frequency Division Multiplexing) systems addresses the Sampling Frequency Offset (SFO) estimation and correction problems of conventional techniques and provides improved performance in a wide range of operating conditions. The technique teaches a correction scheme that uses known pilots in each OFDM symbol to estimate slope of phase error and subsequently calculate the correction factor for each of the sub-carriers in the associated OFDM symbol. The present subject matter further teaches an averaging slope estimates over a window of predetermined number of OFDM symbols to average out noise and get a refined correction factor. In addition, the present technique teaches tracking rate of growth of slope across OFDM symbols to estimate SFO, which can be used for the timing error correction in the time domain.

Abstract: A motion estimation algorithm finds the best match for a given block or macroblock so that the resulting error signal has very low energy level which is computed, for e.g., by the SAD method. The motion estimation algorithm also provides for an optional sub-pixel level estimation and an inter4v search, and allows for restricting the number of searches for a Frame-frame ME (motion estimation) using Top-Top and Bottom-Bottom field MEs. The algorithm provides for a selective early exit and enables selecting a suitable search area with N candidate points (4 to 8) for starting the search. The search is conducted progressively till a minimum error signal (low energy level signal) is reached. The candidate points for search may be in a diamond shaped configuration, and there may be a plurality of successive diamond configurations, the number of which is configurable. The invention has application in MPEG-4 and H.264 standards.

Abstract: In a video application, a method and system provide different sizes of data-fetch where the data transfer rate between a decoder and an external memory (e.g., DDR memory) is extremely high, as for example in HDTV systems. The invention in one form divides a reference frame into different tiles where each tile is hierarchically divided into smaller tiles to a level where the minimum tile size is the same as the fixed burst size of the DDR memory. The method also provides for arranging the biggest tiles into different banks and pages so that even if the block to be fetched falls across tile boundaries, the latency penalty in the tile transition will be minimized. The invention provides advantages also for progressive and interlaced data fetch.

Abstract: A device for Frame Synchronization and coarse frequency offset estimation (CFE) and a method thereof is provided. The method may include the step of receiving a communication signal in a matched filter. The matched filter may be configured for selecting one or more blocks from the communication signal. The matched filter may be further configured for filtering the blocks to obtain an output of the matched filter. The method may also include the step of providing the output of the matched filter to a frame synchronizer and to a coarse frequency estimator. The method may include the step of obtaining substantially simultaneously a symbol boundary using the frame synchronizer and a plurality of metrics using the coarse frequency estimator. Each of the metric may be an indicative of an estimate of coarse frequency offset. The method may further include a step of selecting one of the metric from the plurality of metrics according to the symbol boundary for obtaining an estimate of the coarse frequency offset.

Abstract: An improved decoding process as applied to MPEG AAC decoders of the type that use codebooks, reduces redundant traversing steps while finding the location of a specific binary stream codeword in the codebook. To this end, the codebooks are divided into a plurality of node-tables, each table corresponding to a node. From the number of bits “k” in a given node-table, the decimal value of the first “k” bits in a binary stream to be decoded is ascertained and compared with the decimal value of the maximum code word in the given node-table. The comparison is used for the identification of the location of the codeword in the node-tables to assist in decoding without unnecessary traversing of all the nodes in the codebook.

Abstract: A method (and algorithm) for detection of 802.11a and 802.11b frames in an 802.11 transmission, calculates DCC (delayed copy correlation) and LCC (local copy correlation) magnitudes and compares their normalized values with threshold values. When a signal is detected, (i.e., AGC locks) the algorithm checks for a valid 802.11a frame, and it exploits the DCC properties of the received symbols. Because of the repetitions of short preambles in received symbols, the DCC is high when the short preamble symbols are present. Further, 802.11b detection may be based on Barker correlation properties simultaneously. A counter array is maintained for each position in a Barker window, and is incremented if the normalized LCC of the incoming sequence with the Barker code exceeds a threshold. An 802.11b packet is declared if the maximum value of the counter array exceeds a predetermined number in a specified number of Barker windows.

Abstract: A non-iterative and computationally efficient bit allocation technique for perceptual audio coders employing uniform quantization schemes. This is achieved by computing a target MNR for all critical bands in a frame using a target bit rate and associated SMRs. Associated SNRs are then computed for the critical bands using the computed target MNR and the associated SMRs. Bits are then allocated to the critical bands based on the computed associated SNRs.

Abstract: A method for achieving bit rate controlled encoding, e.g., constant bit rate, uses a bit rate control model based on bits allocated per macroblock and based on error in estimating bits consumed in an encoding process in a macroblock. The method computes bits consumed per macroblock as a function of the formed bit rate control model, and allocates bits for the macroblock. To this end, a quadratic (second order) equation is used for the model, which equation might default into a first order equation requiring fewer computations, in the event that no second order solutions exist. In one form, the bit rate control model computes the bit rate to satisfy video buffer verifier (VBV) compliance in accordance with MPEG requirements. The bit rate control model makes allowances for the type of macroblock, i.e., inter, intra or bidirectional, and provides for any frame-skips.

Abstract: A technique and method provide virtual N-way video conferencing to participants having IP stations/videophones, which have limited processing capacity, by auto-selecting a limited subset of the participants for video display and restricting the rest of the participants to voice-only mode which requires reduced processing capability. The method uses a virtual n-way conferencing module (VNCM) to provide the video conferencing, and an auto caller select module (ACSM) for causing selective video display. Some of the participants might have VoIP terminals instead of IP videophones. The invention includes a technique, e.g., by using a video transmission control module (VTCM) for reducing the bandwidth requirement of IP videophones by signaling the participants in voice-only mode not to transmit the video information in bandwidth-constrained situations.

Abstract: A method for configuration of an Auxiliary Processing Unit (APU) of multiprocessor system is presented. The multiprocessor system has at least a Main Processing Unit (MPU) coupled to the APU via a communication link. The APU has at least a first memory and a second memory. The method includes a plurality of steps. At step the first memory is divided into an application sector, a boot sector and a common sector. At another step interrupts of the APU except interrupt/s that is/are being received via the communication link are disabled. At a further step interrupt vector/s pertaining to the communication link is/are mapped to the boot sector of the first memory. At another step a configuration code is received selectively into the application sector of the first memory and into the second memory. At a further step the interrupt/s that is/are being received via the communication link are disabled. At a further step the common sector of the first memory is updated from the second memory.

Abstract: A technique for signal distortion compensation using a filter that can support a higher delay spread distortion without substantially increasing hardware complexity. In one example embodiment, an adaptive receiver design compensates for signal distortions with the use of non-uniform tap delay filters. The non-uniform tap delay filters are used to output an adaptively channel matched signal for decoding.

Abstract: A method and device for synchronization and rate control of media data comprising adaptively computing an offset time using the reference time of a server, a local time of a client, and an associated time for presentation of the data, stamping the offset time on the media data, receiving the offset time stamped media data into a first buffer, processing the offset time stamped media data, computing an altered offset time, including the altered offset time with the processed media data, storing the processed media data in a second buffer, and selectively supplying the processed media data to a presentation sink unit according to the altered offset time, the server reference time, and the associated time of the media data.

Abstract: A technique to encode an audio signal based on a perceptual model. In one example embodiment, this is accomplished by shaping quantization noise in the spectral lines on a band-by-band basis using their local gains. The noise shaped spectral lines are then fitted within a predetermined bit rate to form an encoded bit stream.

Abstract: A method for improving the performance of OFDM (Orthogonal Frequency Division Multiplexing) systems addresses the channel estimation, equalization and decoding problems of conventional techniques and provides improved performance in a wide range of operating conditions that include varying Signal to Noise Ratio (SNR) levels, data rates, and fading channels. The method teaches SNR based selection in the use/choice of a smoothing filter. The invention further teaches using a smoothing filter in combination with Decision Feedback Equalization (DFE), using a smoothing filter in combination with WCSI (Weighted Channel State Information) decoding, which may be combined with DFE, and data rate based selection (for example, in IEEE 802.11a/g applications) in employing a decision feedback based correction.

Abstract: A technique to enhance audio quality of a quantized audio signal when a perceptual audio coder is operating at low bit rates. The perceptual audio coder uses a modified two-loop quantization technique that maintains audio quality at medium to high bit rates while eliminating artifacts at low bit rates. The perceptual audio coder saves vanishing bands by stealing bits from surviving bands to reduce artifacts at low bit rates.

Abstract: A method for designing a multi-threaded processing operation that includes, e.g., multimedia encoding/decoding, uses an architecture having multiple processors and optional hardware accelerators. The method includes the steps of: identifying a desired chronological sequence of processing stages for processing input data including identifying interdependencies of said processing stages; allotting each said processing sage to a processor; staggering the processing to accommodate the interdependencies; selecting a processing operation based on said allotting to arrive at a subset of possible pipelines that offer low average processing time; and, choosing one design pipeline from said subset to result in overall timing reduction to complete said processing operation. The invention provides a multi-threaded processing pipeline that is applicable in a System-on-Chip (SoC) using a DSP and shared resources such as DMA controller and on-chip memory, for increasing the throughput.

Abstract: Efficient techniques are provided to detect and correct channel errors found in an encoded video signal. In one example embodiment, header and global information in a video frame is analyzed to detect channel errors in the video frame. The video frame is then corrected for detected channel errors by isolating the detected channel errors to a few macroblocks in the video frame to reduce data loss and improve video quality.

Abstract: Efficient techniques are provided to detect and correct channel errors found in an encoded video signal. In one example embodiment, header information in a video frame is analyzed to detect channel errors in the video frame. The video frame is then corrected for detected channel errors by isolating the detected channel errors to a few macroblocks in the video frame to reduce data loss and improve video quality.