522 results about "Frame size" patented technology

Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

Client signals to be transported in a transmission network, particularly an optical transmission network, may have different payload envelope rates and are digitally mapped on the client egress side into first transport frames (also referred to as iDTF frames, or intra-node or internal digital transport frames), at the client side for intra-transport within terminal network elements (NEs) and further digitally mapped into second transport frames (also referred to as DTFs or digital transport frames) for inter-transport across the network or a link which, through byte stuffing carried out in the first transport frames so that they always have the same frame size. As a result, the system of framers provides for a DTF format to always have a uniformly universal frame rate throughout the network supporting any client signal frequency, whether a standard client payload or a proprietary client payload, as long as its rate is below payload envelope rate of the client signal. At the client signal ingress side, the signal are digitally demapped from the second transport frames (DTF format) into the first transport frames where the stuff bytes are removed and accordingly processed at an intermediate node element before further transport, or digitally demapped from the first transport frames (iDTF format) to reproduce or reassemble the client signal or signals comprising the client payload at the client payload envelope rate for reception at the client's equipment. Among various features disclosed, two predominate features are (1) a single channel or network rate for transport of all signals between network elements (NEs) and end terminal network elements and (2) the digitally wrapping of different types of payloads into N client side or first frames using stuff bytes to render each client side frame size equal to a predetermined value. Then the stuffed first frames are wrapped into line side or second frames for transport over the network at the same high speed line rate for all digitally wrapped client signals. The client side framers may be, for example, running at the lowest signal rate encountered, to digitally wrap then into parallel N client signals or digitally wrap a client signal multi-sected into N parts, where these two different client signals have different payload rates.

An audio encoder for encoding audio samples has a first time domain aliasing introducing encoder configured to decode audio samples in a first encoding domain and having a first framing rule, a start window and a stop window. The audio encoder further has a second encoder configured to encode samples in a second encoding domain and having a predetermined frame size number of audio samples, and a coding warm-up period number of audio samples, the second encoder having a different second framing rule, a frame of the second encoder being an encoded representation of a number of successive audio samples that is equal to the predetermined frame size number of audio samples. The audio encoder further has a controller switching from the first to the second encoder and for modifying the second framing rule or for modifying the start or the stop window of the first encoder.

The apparatus generates simulated strobe effects in the form of video or still image output in response to receipt of a video stream, and without the need of additional strobe hardware. Videos of a moving target object are categorized into one of multiple categories, from which a strobe generation process is selected. In one mode, the two categories comprise target objects with either small motion or large motions in relation to the frame size. Interoperation between image registration and cloning are utilized to produce simulated strobe motion videos or pictures. Motion segmentation is applied to the foreground object in each image frame, and a foreground mask is updated as each checkpoint is reached along the object trajectory, such as in response to time differences between checkpoints. Potential applications include special features for camcorders, digital cameras, or computer software.

What is provided is an enhanced dynamic address translation facility. In one embodiment, a virtual address to be translated and an initial origin address of a translation table of the hierarchy of translation tables are obtained. Dynamic address translation of the virtual address proceeds. In response to a translation interruption having occurred during dynamic address translation, bits are stored in a translation exception qualifier (TXQ) field to indicate that the exception was either a host DAT exception having occurred while running a host program or a host DAT exception having occurred while running a guest program. The TXQ is further capable of indicating that the exception was associated with a host virtual address derived from a guest page frame real address or a guest segment frame absolute address. The TXQ is further capable of indicating that a larger or smaller host frame size is preferred to back a guest frame.

The subject matter herein generally relates to transcoding content, typically audio/video files though not limited to such, from one version to another in preparation for online streaming or other delivery to end users. Such transcoding may involve converting from one format to another (e.g., changing codecs or container formats), or creating multiple versions of an original source file in different bitrates, frame-sizes, or otherwise, to support distribution to a wide array of devices and to utilize performance-enhancing technologies like adaptive bitrate streaming. A transcoding platform is described herein that, in certain embodiments, leverages distributed computing techniques to transcode content in parallel across a platform of machines that are preferably idle or low-utilization resources of a content delivery network. The transcoding system also utilizes, in certain embodiments, improved techniques for segmenting the original source file so as to enable different segments to be sent to different machines for parallel transcodes.

The multi-window system includes: a main storage unit storing display data including a display frame and a display element, a standard display frame size, and a standard text size; a display area size obtaining unit obtaining a display area size; a text size setting unit determining a text size for correcting the standard text size, based on the display area size with reference to the standard display frame size, and setting the text size to the display data as the actual text size of the text to be added to the display element; and a drawing unit generating, using the display data, the image data of the display frame and the display element added with the text having the text size, and displaying the image data in the display area by storing the image data in a display data storage unit different from the main storage unit.

Methods and systems for transcoding in a cloud computing platform are disclosed. According to an embodiment, a receiver receives an uploading file by one data block at a time, and stores the received data blocks in various storage modules. Small segment files are then generated when the size of the received data blocks is larger than a threshold. A transcoder transcodes the small segment files from one format such as a bit rate or a frame size to another while the receiver is still receiving a new data block. The transcoded small segment files may be stitched together to form a stitched file, which may be stored in a storage module to be downloaded through a content distribution network (CDN). The transcoded small segment files may be passed to streaming servers for streaming over a network while the receiver is still receiving a new data block of the uploading file.

The present invention relates to a method for predicting an optimum transmission frame size in a wireless computer network. The method comprises assessing transmission channel quality in the network, calculating an optimum length for the transmission frame, adjusting the length of the transmission frame to the predicted optimum length, transmitting the frame at its adjusted length and assessing the quality of the transmission of the frame. Prediction of the optimum frame size employs a Kalman filter which employs the parameters of bit error rate and random processing noise in the calculation of the predicted optimum frame length.

Rate matching and rate shaping methods and systems for managing data traffic on wireless channels are disclosed. Rate matching embodiments dynamically determine downlink video data transmission rates based on frame decoding deadlines and frame sizes. In addition, if the channel capacity cannot support the video data transmission rate, the rate matcher can adjust the video stream accordingly so that it is supportable by the channel capacity. In turn, rate shaping implementations can tailor the transmission rate shaping based on the state of the wireless data channel. Shaper operations can be specifically adapted to a saturated state of the wireless channel, to an unsaturated state of the wireless channel and to transitions between the unsaturated and saturated states.

Disclosed are a display panel, a display device and a driving method thereof. The display panel includes a plurality of sub-pixel cells arranged in an array including N rows and M columns and a gate driving circuit. Each of the plurality of sub-pixel units includes a light emitting unit, a pixel driving circuit for driving the light emitting unit to emit light, and a sensing circuit for sensing the pixel driving circuit. The gate driving circuit includes N+1 sequentially arranged output terminals and is configured to output a gate scanning signal that turns N rows of sub-pixel cells of the array on line by line. The pixel driving circuit of the nth row sub-pixel unit and the nth output terminal of the gate driving circuit are connected to receive a gate scanning signal as a scan driving signal, and the sensing circuit of the nth row sub-pixel unit and the n+1 output terminal of the gate driving circuit are connected to receive a gate scanning signal as a sense driving signal. The display device using the display panel can reduce the frame size.

Provided herein are exemplary embodiments of a system for and method of initially and dynamically allocating the available resources that affect video quality in an asymmetric endpoint network so as to provide an enhanced quality of video exchange. Relevant variables that may be dynamically configured to allocate resources include, but are not limited to, frame size, frame rate, choice of codec (e.g., MPEG4, H263, H263+, H264), codec bit rate and size of rendering window (which may or not be identical to the frame size).

A method and device for improving rate controlling in video coding of sequences including a series of Inter frames separated by Intra frames, when a decoding delay is considered, comprise for each Inter frame of the series: computing a target frame size, computing a maximum buffer level related to a position of each Inter frame relative to a previous Intra frame and an upcoming Intra frame, and optimizing a transmission buffer level in response to the computed target frame size and the computed maximum buffer level.

An enhanced radio link protocol (RLP) in a wireless access network that is network aware is disclosed. The RLP increases radio link quality by various ARQ mechanisms. The RLP framing structure is included that supports and enables at least network layer packet boundary detection, dynamic and adaptive ARO schemes for QoS support on a per-packet basis, and a flexible RLP frame structure for fast adaptation to physical layer channel rate/RLP frame sizes. Optional uses include supporting negative acknowledgment (NAK) based ARQ.

The invention relates to an intra-frame prediction method, which increase the intra-frame prediction compression efficiency and comprises the following steps: (1) a brightness intra-frame prediction processing unit and a chroma intra-frame prediction processing unit are selected according to the image coding (decoding) frame size; (2) the sizes and the prediction modes of a brightness intra-frame prediction subblock and a chroma intra-frame prediction subblock are selected respectively; (3) the intra-frame prediction is performed according the selected sizes and the prediction modes of the brightness intra-frame prediction subblock and the chroma intra-frame prediction subblock to obtain an intra-frame prediction sampling value and a residual error of the intra-frame prediction subblocks; and (4) the coding mode of the intra-frame prediction subblocks, the brightness intra-frame prediction mode as well as the chroma intra-frame prediction mode and the residual error are coded respectively and then are sent to a code stream. The classification of the intra-frame prediction method is mainly aimed at the circumstances that the sizes of the intra-frame prediction subblocks are 4*8, 8*4, 8*16 and 16*8, and five optional prediction modes are matched, so that 3 percent to 5 percent of code stream bandwidth is saved under the condition that the image quality is identical, and the image encoding compression efficiency is increased; and the effect of simplifying the computational complexity can be also achieved due to the selection of partial prediction modes.

Provided is a method for performing hierarchical B picture-based coding on a video sequence using the structure of adaptively divided group of pictures (GOP). The method includes the steps of, for each predefined 2^N frame-sized group of pictures (GOP) of the video sequence, (a) encoding the 2^N frame-sized GOP of the video sequence based on each of the different GOP sizes from the maximum size, 2^N, to the minimum size, 2^M (M is an integer between 1 and N) and obtaining different values between frames reconstructed after the encoding is performed and frames after the hierarchical B-picture prediction is performed, based on each of the different GOP sizes; (b) selecting at least one sub-GOP based on the difference values obtained by encoding the 2^N frame-sized GOP of the video sequence based on each of the different GOP sizes; and (c) generating a bitstream by encoding the 2^N-frame-sized GOP based on the at least one selected sub-GOP. Thereby, the hierarchical B picture-based video coding is performed by adaptively dividing the GOP size based on performance and thereby obtains high coding efficiency.

A method includes receiving streaming media data having a media frame and a frame index referencing the media frame; allocating a frame budget for an output media frame by estimating a frame size of the output media frame based on the frame index; generating the output media frame in real-time by processing the media frame based on first processing parameters and, if the allocated frame budget is greater than a frame size of the processed media frame, padding the processed media frame; and providing the output media frame.

A system and method for adaptively managing frame size in a wireless multihop network (100) is disclosed. In one embodiment, a packet is transmitted from a source to a destination (420). A acknowledgement packet (422) is received and a successful acknowledgement packet counter is incremented if the acknowledgement packet arrives prior to a time-out of a timer (444). A frame size is increased if the successful acknowledgement packet counter reaches a specified value (446, 448). If the acknowledgement packet arrives after the time-out of the timer, the successful acknowledgement packet counter (460) is reset and the frame size (462) is decreased. These procedures can be repeated until the frame size is greater than or equal to a maximum frame size (450) or less than or equal to a minimum frame size (464).

There is provided a speech coding/decoding apparatus and method, in which the input speech signals are classified into several classes in accordance with characteristics of the input speech signals and the input speech signals are coded using frame sizes, quantizer structures, and bit assignment methods corresponding to the determined classes, or in which the frame sizes can be adjusted in accordance with network conditions or codec type of a counter part. Therefore, by optimally adjusting the frame size, the quantizer structure, and the bit assignment method in accordance with the characteristics of input speech, it is possible to improve the performance of the speech coding apparatus, and by adjusting the frame size in accordance with the speech codec type of a counter part, it is also possible to reduce the total end-to-end delay.