57 results about "Bit Rate Reduction" patented technology

Described herein are systems and methods for processing and transmitting data included in packets with jitter. The systems and methods combine a jitter buffer and a buffer scanned by a look ahead window as a combined buffer. The combined buffer varies look ahead window and performance of a process that uses the look ahead window based on the current jitter support in the buffer. More specifically, the flexible buffering techniques flexibly adapt the size of the look ahead window based on the amount of buffer space used for jitter support. When jitter support requirements are low, the size of the look ahead window is increased. When the look ahead window is implemented with a multiplexer, the multiplexer may benefit from a larger look ahead window and improve multiplexer performance. Improved multiplexer performance may lead to less or more intelligent re-encoding and bit rate reduction for video data transmission applications, leading to improved decoded output video quality.

A video frame is input to a spectrum analyzer, which transforms the frame from a spatial domain into a frequency domain. A resolution determiner analyzes the output of the spectrum analyzer, thereby selecting a resolution appropriate for the video frame. Then, a resolution converter converts a resolution of the video frame into the resolution selected by the resolution determiner. Thereafter, the video frame is subjected to a coding process by block divider, DCT transformer, quantizer and variable-length coder so as to be output as a coded bit stream. By coding the input video frame at a resolution corresponding to the high-frequency component in the frame, a video of higher quality can be obtained with the bit rate reduced.

A system apparatus and method for concurrent delivery of a plurality of video streams comprising one or more transcoders for bitrate reduction and a cache memory to save copies of transcoded segments. When the total outgoing bitrate of the system exceeds a predefined threshold, the system decides which of the video segments to transcode and the magnitude of bitrate reduction. Before assigning a transcoder for the bitrate reduction, the system first checks the cache for saved transcoded segments. If appropriate transcoded segments are not found in cache, the system may assign a transcoder for the video segments designated for bitrate reduction.

The present invention relates to methods and apparatus for quantitatively measuring the performance of a statistical remultiplexer. The methods of the present invention provide for generation of evaluation metrics associated with at least one of amount of bit rate reduction performed by the statistical remultiplexer, change in video quality attributable to the statistical remultiplexer, wasted output bandwidth by the statistical remultiplexer, decoder buffer level fullness, bit rate reduction characteristics of the statistical remultiplexer, and time delay attributable to the statistical remultiplexer.The methods of the present invention may be implemented by an evaluator that may be incorporated into a statistical remultiplexer or may be separate from and connectable to a statistical remultiplexer.

A video transmission system that achieves uninterrupted video transmission, even when there are large bit rate fluctuation due to receiving terminal movement and so forth, in a network such as a wireless network in which transmission bit rate fluctuations occur. In this system, when a video receiving apparatus (receiving terminal) 150 is moving, a video transmitting apparatus (transmitting terminal) 100 lowers the base layer bit rate of layered-coded data to the limit. When the base layer bit rate is lowered in this way, the bit rate of the lowest enhancement layer is raised and effects on the received image quality of other terminals due to lowering of the base layer are suppressed, or the lowest enhancement layer is divided finely and the adjustability to a bit rate under bit rate fluctuations is improved.

An improved communication method for sending media between a sending node and receiving node during a conversation. When network bandwidth is insufficient to transmit a full bit rate representation of time-sensitive media, then a reduced bit rate representation of the media is transmitted for the purpose of increasing the ability of the recipient to review the media upon receipt and continue the conversation in the real-time mode when the bandwidth on the network is insufficient to support the transmission of the full bit rate representation. Media that is ascertained as not time-sensitive on the other hand is transmitted when bandwidth in excess of what is needed for time-sensitive media becomes available. When the media ascertained as not time-sensitive is transmitted, the rate of transmission is adjusted at the sending node based on network conditions, the adjusted rate of transmission being set for network efficiency and reliable delivery of the media ascertained as not time-sensitive relative to the timeliness of the delivery of the media ascertained as time-sensitive.

The semi-fragile watermark comprises a fragile watermark component and a robust watermark component. Features are extracted from the video stream and subsequent hashing and encryption processes are performed to generate the fragile watermark. The fragile watermark includes control data information at the block level as well as frame and group level information. The fragile watermark is added on top of the robust watermark, giving the system the ability to detect alteration at the block level as well as the group level. The resulting semi-fragile watermark has the advantage of being both sensitive to malicious attack while being robust enough to survive bit rate reduction and other types of manipulation typically performed on digital multimedia signals.

A method and apparatus to encode / decode an audio signal, in which a bit rate for each bit plane can be controlled. A method of encoding an audio signal for each of a plurality of bit plane can include dividing the audio signal into a plurality of frequency bands and encoding the bit planes of the frequency bands from a low frequency band to a high frequency band, wherein, in the encoding the bit planes of the frequency bands, the bit planes are encoded from the most significant bit (MSB) to the least significant bit (LSB) within bits allocated for the frequency bands, and when there are allocated bits remaining after the encoding of the currently encoded frequency band, un-encoded bit planes corresponding to the MSB in a frequency band that has the fewest encoded bit planes among frequency bands with a lower frequency than the currently encoded frequency band are encoded using the remaining allocated bits. Accordingly, when encoding / decoding an audio signal, an encoding sequence of bit planes is determined so that an audio signal that significantly affects audio quality during decoding is first encoded, thereby reducing audio quality deterioration at a low bit rate.

A method and apparatus for controlling the recording of streaming data by a mobile recording apparatus is disclosed. A streaming data input signal is received and the streaming data signal is compressed in a scalable manner using a scalable encoder to create layered encoded streaming data streams. Storage of the layered encoded streaming data streams is begun in a storage device at a first bit-rate. The remaining battery life for the apparatus is determined at the first bit-rate. The bit rate is reduced to a second bit-rate by stopping the storage of at least one of the layered encoded streaming data streams to lengthen the remaining battery life of the apparatus.

Method and apparatus for encoding video is described. In one example, average of motion vectors for each of a plurality of blocks is maintained over a threshold number of inter-coded frames in the video. The running average of motion vectors for each of the plurality of blocks is compared to a threshold value. Each of the plurality of blocks the running average of which does not satisfy the threshold value is encoded using a first quantization parameter. Each of the plurality of blocks the running average of which satisfies the threshold value is encoded using a second quantization parameter. The second quantization parameter results in a coarser quantization of transformed coefficients than the first quantization parameter.

Transmission channel monitoring, detection, classification, and adaptation provide solutions to severe information blockage problems during transmission. Channel monitoring is achieved through two types of detection and re-transmission strategies. Channel detection is accomplished by means of counting the number of negative acknowledgements, which classifies the channel degradation level and determines which alternatives to be adapted in order to combat specific blockage phenomenon. Blockage level is first translated into and related to transmission link parameters in terms of total carrier power to noise ratios, from which the methods of bit rate reduction, modem density increasing, and higher encoding rate are brought forth. A blockage identifier and the optimal smearing technique are used. Frequency-diversity and space-diversity, a method for combating complete blockage and an adaptive programmable channel equalization technique independent of channel characteristics are described. A cascading technique is taught to improve significantly false- and miss-detection probabilities. Performance measures are established in terms of system efficiency and error probability for blockage mitigation with unequal terminal capabilities.

A digital broadcast recording / reproducing apparatus which is able to determine the bit rate at the time of recording operation in accordance with the bit rate of the recorded video data and audio data, so that the bit rate at the time of reproducing operation is reduced lower than the bit rate at the time of recording operation is obtained. When reserving the reproduction bit rate, it prevents taking a large band superfluously. The amount of records and time for every picture which are the coding unit of the video signal of MPEG-2 are held at the time of recording operation, and the bit rate for every picture is calculated from these data. The value somewhat higher than the maximum or maximum as the reproduction bit rate is set up, and it reproduces. By doing in such a way, the bit rate at the time of reproducing operation can be decided in accordance with the bit rate of the recorded video and audio signals, so that the bit rate at the time of reproducing operation can be reduced lower than the bit rate of the TS at the time of recording operation.

A method and apparatus for reducing the bitrate of a given datastream is provided. The bitrate of a given datastream is reduced by altering the payload of an incoming packet, while still providing sufficient data as to not cause a malfunction within the receiving node. The altered payload represents data which will trigger commonly used error concealment methods, within the receiving node. The determination as to which packets undergo alteration is based on a relationship between network congestion and the priority of a given packet. A packet's priority is measured in terms of its effect on the regeneration of high quality signal. As the need to reduce the bitrate of a given datastream increases, the occurrence of packet alteration increases.

A fine granularity scalability encoding / decoding apparatus includes a first quantizer for performing a DCT on a motion-compensated image and quantizing a resulting value, a second quantizer for re-quantizing the value obtained by the first quantizer, an inverse-quantizer for re-quantizing a value re-quantized by the second quantizer, and a number of subtracters. A first subtracter obtains a difference between a value of N times the re-quantized value and the inverse-quantized value. A second subtracter obtains a difference between the value quantized by the first quantizer and the value quantized by the inverse-quantizer. And, a third subtracter subtracts the output value of the first subtracter from the output value of the second subtracter. Coding efficiency is increased and a bit-rate reduction of an entire coding stream is achieved by making an enhancement layer signal and a base layer signal have the same code.

A multiplexer applies dynamic bit rate reduction at the multiplexer level in accordance with the types of video input streams as determined from information contained in units of the video input streams. The multiplexer parses the Network Abstraction Layer (NAL) headers of said units to determine their relative importance and passes them on to its output accordingly. The multiplexer can also take advantage of the relation between streams if they are related, as in the case of Scalable Video Coding (SVC).

The method is characterized in that:if the current macroblock was coded according to the “skipped macroblock” mode and if at least one corrected coefficient block of said macroblock has coefficients non-null or greater than a predefined threshold, the coding mode of the macroblock is modified into a mode other than “skipped macroblock” and the data of the data stream is modified or added to specify this mode and the value of the coefficients.

A new dequantization scheme for DCT-based transform coding, such as JPEG, MPEG and H.26x, is disclosed. The new approach drastically reduces blocking artifacts without smoothing the decoded image. Most discrete cosine transform (DCT) based video coding suffers from blocking artifacts where boundaries of 8×8 DCT blocks become visible on decoded images. The blocking artifacts become more prominent as the bit rate is lowered. In the present invention, a new dequantization technique is disclosed for discrete cosine transform (DCT) based encoding to sharply reduce the blocking artifacts. The dequantization scheme of the present invention sharply reduces blocking artifacts in decoded images through regularization. The performance comparison with the standard JPEG as well as MPEG and H.26x decoding shows visual improvements as well as numerical improvements in terms of the peak-signal-to-noise ratio (PSNR) and the blockiness measure (BM) to be defined.

Method and apparatus for encoding video is described. In one example, average of motion vectors for each of a plurality of blocks is maintained over a threshold number of inter-coded frames in the video. The running average of motion vectors for each of the plurality of blocks is compared to a threshold value. Each of the plurality of blocks the running average of which does not satisfy the threshold value is encoded using a first quantization parameter. Each of the plurality of blocks the running average of which satisfies the threshold value is encoded using a second quantization parameter. The second quantization parameter results in a coarser quantization of transformed coefficients than the first quantization parameter.

The invention provides a depth signal detection method for a high-speed rail, and the method comprises the steps: firstly, collecting data, and collecting a plurality of sending signals and receivingsignals in each scene along the high-speed rail according to different environment types along the high-speed rail; secondly, dividing scenes, and further dividing each scene into a plurality of regions through data analysis to meet the compatibility of the neural network; thirdly, establishing a deep high-speed rail signal detection neural network model; secondly, training a high-speed rail signal detection neural network offline; and finally, carrying out online real-time signal detection, determining the position information of the high-speed rail through a GPS in the driving process of thehigh-speed rail, judging the area where the high-speed rail is located, selecting a corresponding neural network model, then inputting signals received in real time into the trained neural network, and outputting signals sent by the base station end in real time. The system performance is greatly improved, the signal detection bit error rate is reduced, and the algorithm is more robust. The method used in the invention does not need to estimate the channel, thereby saving the pilot cost.

The invention discloses an interference noise elimination method suitable for a power line carrier (PLC) communication network, which is mainly used for solving the impulse interference noise which appears randomly and has high energy causes performance attenuation of the whole communication system in the PLC network. The method mainly comprises two parts of receiver-side signal amplitude pretreatment and robust channel estimation. The initial filtering of the impulse interference noise is performed according to the statistical characteristics of received signals by the receiver-side signal amplitude pretreatment, and the influence on the residual noise is further weakened by the robust channel estimation. Compared with other conventional power line noise elimination algorithms, the method provided by the present invention is characterized by a relatively low complexity and being insensitive to noise change. The method eliminates the power line noise from the viewpoint of practical feasibility and effectiveness, so that the bit error rate is reduced during the power line carrier transmission, and the system reliability can be improved.

When the bit rate of an input transport stream is reduced to a lower bit rate S by reencoding, the bit rate Ro of streams other than video is set to be the same as the bit rate of the streams other than video in the input transport stream. Then, the value obtained by subtracting Ro from S (S−Ro) is set as the bit rate for the video stream to be reencoded.

The invention discloses a lash-up JPEG bit rate reduction control device, which includes a JPEG module and a buffer module, as well as a buffer management module. The JPEG module is used to compress input data block and output data stream to the buffer module. The buffer management module is used to control JPEG module whether to discard quantified AC coefficient and the discard number, according to the buffer filling state of the buffer module. The invention also discloses a lash-up JPEG bit rate reduction control method. The invention can greatly reduce data volume in video stream compression, and avoid transfer buffer overflow, with no image quality change sensed by users.

In this disclosure, a novel method for direct mode enhancement in B-pictures and skip mode enhancement in P-pictures in the framework of H.264 (MPEG-4 / Part 10) is disclosed. Direct mode and skip mode enhancements are achieved by clustering the values of the Lagrangian, removing outliers and specifying smaller values of the Lagrangian multiplier in the rate-distortion optimization for encoding mode selection. Experimental results using high quality video sequences show that bit rate reduction is obtained using the method of the present invention, at the expense of a slight loss in peak signal-to-noise ratio (PSNR). By conducting two different experiments, it has been verified that no subjective visual loss is visible despite the peak signal-to-noise ratio change. In relationship to the existing ratedistortion optimization methods currently employed in the (non-normative) MPEG4 / Part 10 encoder, the method of the present invention represents a simple and useful add-on. More importantly, when other solutions such as further increasing the values of the quantization parameter are not applicable, as inadmissible artifacts would be introduced in the decoded pictures, the method of the present invention achieves bit rate reduction without introducing visible distortion in the decoded sequences. Even more, despite the fact that the present document makes use of the H.264 framework, the proposed method is applicable in any video encoding system that employs ratedistortion optimization for encoding mode selection.

A method for dispersion compensation regulation wherein, in order to transmit high-bit-rate optical signals, in a first step a bit-rate-reduced optical signal is formed from the high-bit-rate optical signal by the N-fold repetition at the transmitting end of each logic 1 and each logic 0, and in a second step the bit-rate-reduced optical signal is transmitted. In a third step the bit error rate of the transmitted bit-rate-reduced optical signal is determined at the receiving end, and in a fourth step the absolute-magnitude compensations of the dispersion compensation units are regulated in such a way that the determined bit error rate is minimized. After the bit error rate has been minimized by regulations the second to fourth steps are repeatedly carried out for the high-bit-rate optical signal.

The invention discloses a high-speed rail traction power supply monitoring architecture and steps. A data microcontroller MCU is connected with a storage module; the storage module can be a flash memory; the storage module can include a time-consuming change recognition program for identifying a change in time-consuming of a present GPRS mobile communication network via a total of GPRS mobile communication network time-consuming of a preset time length range, an recognition derivation program, and a bit rate reduction program for transmitting the processed information, wherein the affirmationexport program is used for affirming the situation that the GPRS mobile communication network has an obstruction situation by virtue of the time-consuming change of the current GPRS mobile communication network; and in combination with other structures and methods, the defects that in the prior art, the GPRS mobile communication network with a small bit rate is often blocked by large hysteresis, more time is consumed for trimming the GPRS mobile communication network, and the timeliness of processed information transmission is unfavorable are effectively avoided.

The invention concerns a method of encoding signals, in particular digitised audio signals, with an encoding device for encoding the signal in an encoding format and a processing device for processing the encoded signal. Methods of that kind are known for example from European patent specification No 290 581. In that case, in the bit rate-reducing encoding of audio signals which are already present in digitized form, for example 48 kHz sampling frequency / 16-bit resolution, psycho-acoustic phenomena of the perception of audio signals are used in such a way that the original bit rate of the audio signals is considerably reduced. Such methods are also familiar and standardised under the heading of ‘source encoding’ (ISO 11172 and 11318). The object of the invention is to provide a method of the kind set forth in the opening part of this specification, which resolves the above-indicated problems and in which re-coding operations, once encoding has been effected, are very substantially avoided.