Patent application number | Description | Published |
20080205520 | Method of coding a video signal - The invention relates to methods and apparatuses for encoding and decoding of a video sequence. In connection with encoding/decoding a video sequence it is desirable to increase the video quality without having to increase the bit-rate for the encoded video too much, thereby still providing a bit-efficient representation of the video. If multiple descriptions of the video sequence is used the invention improves the video quality without any increase of the bit-rate. According to the invention, this is achieved by using two or more coding units for encoding the same video sequence, wherein the encoding units perform their encoding operations displaced in time in relation to each other. Correspondingly, two or more decoding units are used for decoding the same video sequence, wherein the decoding units perform their decoding operations displaced in time in relation to each other. | 08-28-2008 |
20140079123 | INDEPENDENT TEMPORALLY CONCURRENT VIDEO STREAM CODING - Implementations of independent temporally concurrent video stream coding may include generating a sequence of encoded frames by encoding a plurality of input frames, wherein encoding the plurality of input frames may include generating a first plurality of encoded frames based on the plurality of input frames, the first plurality of encoded frames including a first plurality of intra-coded frames and a first plurality of inter-coded frames, and independently generating a second plurality of encoded frames based on the plurality of input frames, wherein the second plurality of encoded frames includes a second plurality of intra-coded frames and a second plurality of inter-coded frames, such that the first plurality of encoded frames and the second plurality of encoded frames are temporally concurrent, and such that the intra-coded frames from the second plurality of intra-coded frames are temporally nonconcurrent with the intra-coded frames from the first plurality of intra-coded frame. | 03-20-2014 |
Patent application number | Description | Published |
20120027028 | ADAPTIVE, SCALABLE PACKET LOSS RECOVERY - A system for transmitting data packets representing a source signal across a packet data network is provided. Additionally provided are methods and an apparatus for encoding parameters representing the source signal and also decoding these parameters. The system allows adaptation to the loss scenario of data packets transmitted across the packet data network. A redundancy encoding is generated with a bit rate continuously scalable, the bit rate being provided by a bit rate controller that uses input from the network and packet-loss rate information. The specification can be changed for each coding block. At the decoder, recovery is performed by a parameter estimator based on a dynamically generated statistical model of the effect of the quantizers. The method may be added to existing lossy source coding systems or may be used to enhance the quality of the reconstructed source signal even in scenarios without packet loss. | 02-02-2012 |
20120063572 | DELAY ESTIMATOR - The present invention provides a method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the referenced point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences. | 03-15-2012 |
20120281914 | DETECTION AND SUPPRESSION OF FLICKER IN A SEQUENCE OF IMAGES - The invention relates to a method, device and computer-program product for detection of undesired temporal variations (flicker) in a sequence of video frames. In one embodiment, frame-wise luminance means are compared with a reference level and the crossing frequency is compared with expected variation frequencies, such as frequencies associated with an illumination frequency through aliasing. The crossings count can be refined by introducing a latency zone around the reference level. In case of a positive detection of an undesired temporal variation, there is further provided a correction method, device and computer-program product using cumulated distribution functions. The visual detriment of flicker-induced saturation of pixels is alleviated either by brightening non-saturated pixels or by replacing the saturated pixels by randomly sampled values in accordance with a reference cumulated distribution function. The invention provides embodiments suitable for real-time processing of streamed video sequences. | 11-08-2012 |
20120295649 | DISTRIBUTED BLIND SOURCE SEPARATION - Systems and methods for using distributed processing in conjunction with blind source separation techniques for signal processing and acquisition in sensor network environments are provided. In the distributed blind source separation framework, sensors each perform some processing of sensor signals rather than transmitting such signals over long distances, and/or outside of the sensor network, to be processed at a central location. Sensors attempt to own a source signal, and a source signal can only be owned by one active sensor. Sensors that own a source signal broadcast the source signal directly or indirectly so that it is perceived by users. Sensors receive information from other sensors in their sensor neighborhood, including the observed signals of the other sensors and the estimated source signals of the sources owned by the other sensors. This allows all owning sensors to extract the respective source signals associated with the sources they own and all redundant sensors to check if there are any non-owned source signals present. | 11-22-2012 |
20130163698 | LOW COMPLEX AND ROBUST DELAY ESTIMATION - A method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the reference point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences. | 06-27-2013 |
20140026020 | ADAPTIVE, SCALABLE PACKET LOSS RECOVERY - A system for transmitting data packets representing a source signal across a packet data network is provided. Additionally provided are methods and an apparatus for encoding parameters representing the source signal and also decoding these parameters. The system allows adaptation to the loss scenario of data packets transmitted across the packet data network. A redundancy encoding is generated with a bit rate continuously scalable, the bit rate being provided by a bit rate controller that uses input from the network and packet-loss rate information. The specification can be changed for each coding block. At the decoder, recovery is performed by a parameter estimator based on a dynamically generated statistical model of the effect of the quantizers. The method may be added to existing lossy source coding systems or may be used to enhance the quality of the reconstructed source signal even in scenarios without packet loss. | 01-23-2014 |
20140112481 | HIERARCHICAL DECCORELATION OF MULTICHANNEL AUDIO - Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled. | 04-24-2014 |
20140204716 | SELF-LOCALIZATION FOR A SET OF MICROPHONES - Provided are methods and systems for finding the location of sensors (e.g., microphones) with unknown internal delays based on a set of events (e.g., acoustic events) with unknown event time. A localization algorithm may iteratively run to compute the acoustic event times, the observation delays, and the relative locations of the events and the sensors. | 07-24-2014 |
20140207473 | REARRANGEMENT AND RATE ALLOCATION FOR COMPRESSING MULTICHANNEL AUDIO - Provided are methods and systems for rearranging a multichannel audio signal into sub-signals and allocating bit rates among them, such that compressing the sub-signals with a set of audio codecs at the allocated bit rates yields an optimal fidelity with respect to the original multichannel audio signal. Rearranging the multichannel audio signal into sub-signals and assigning each sub-signal a bit rate may be optimized according to a criterion. Existing audio codecs may be used to quantize the sub-signals at the assigned bit rates and the compressed sub-signals may be combined into the original format according to the manner in which the original multichannel audio signal is rearranged. | 07-24-2014 |
20140355767 | METHOD AND APPARATUS FOR PERFORMING AN ADAPTIVE DOWN- AND UP-MIXING OF A MULTI-CHANNEL AUDIO SIGNAL - A method and apparatus for performing an adaptive down-mixing of a multichannel audio signal comprising a number of input channels, wherein a signal adaptive transformation of said input channels is performed by multiplying the input channels with a downmix block matrix comprising a fixed block for providing a set of backward compatible primary channels and a signal adaptive block for providing a set of secondary channels | 12-04-2014 |
20150055800 | ENHANCEMENT OF INTELLIGIBILITY IN NOISY ENVIRONMENT - Provided are methods and systems for enhancing the intelligibility of an audio (e.g., speech) signal rendered in a noisy environment, subject to a constraint on the power of the rendered signal. A quantitative measure of intelligibility is the mean probability of decoding of the message correctly. The methods and systems simplify the procedure by approximating the maximization of the decoding probability with the maximization of the similarity of the spectral dynamics of the noisy speech to the spectral dynamics of the corresponding noise-free speech. The intelligibility enhancement procedures provided are based on this principle, and all have low computational cost and require little delay, thus facilitating real-time implementation. | 02-26-2015 |
Patent application number | Description | Published |
20090063158 | EFFICIENT AUDIO CODING USING SIGNAL PROPERTIES - An audio encoder comprising optimizing means ET OPT adapted to generate an optimized encoding template OET based on properties PV of an input audio signal IN, such as in form of a property vector. The optimized encoding template OET is being optimized with respect to a predetermined encoding efficiency criterion. Encoding means ENC then generates an encoded audio signal OUT in accordance with the optimized encoding template OET. The audio encoder may comprise analyzing means AN adapted to generate the set of input signal properties PV based of the input signal IN. In a preferred embodiment the optimizing means ET OPT is adapted to estimate a resulting distortion associated with an encoding template. The optimizing means ET OPT may further be able to estimate bit rate associated with an encoding template. In one embodiment the optimizing means ET OPT is adapted to optimize a bit rate distribution to a number of sub-encoders based on the input signal properties (PV). In another embodiment, the optimizing means ET OPT is adapted to up-front decide on an adaptive segmentation based on the input signal properties (PV). The encoders according to the invention are advantageous in that complex processes of a plurality of encodings prior to deciding upon an optimized encoding template OET can be avoided since the optimal encoding template OET is found based on input signal properties (PV). | 03-05-2009 |
20100054279 | ADAPTIVE, SCALABLE PACKET LOSS RECOVERY - A system for transmitting data packets representing a source signal across a packet data network is provided. The encoder comprises a first encoder ( | 03-04-2010 |
20100215092 | Method and Apparatus for Multiple Description Coding - The present invention relates to a method and apparatus to be used in designing an index assignment matrix for use in multiple description coding of an information signal. The bandwidth of the index assignment matrix is selected in dependence of transmission condition information relating to a transmission condition of a communication channel onto which a description of the information signal can be transmitted. | 08-26-2010 |
20110050997 | FLICKER SUPPRESSION - The invention relates to a method, device and computer-program product for suppression of undesired temporal variations, notably flicker, in a sequence of video frames. Histogram-based and similar approaches generally do not remove all flicker. Features that are resolved only in portions of the flicker cycle will manifest themselves as residual flicker. This effect is near-universal in bright regions of a scene. The inventive solution is a mapping that aims to resolve in the output only those features that are resolved in all frames of the flicker cycle. Use of time-maximal quantile values may preserve non-resolution of such image features that are unresolved due to intermittent bright saturation. Thus, in one embodiment, a reduction of resolution is attained by means of a pixel-value mapping based on selecting, over a time window, maximal and minimal quantile values, with maximal values being used for bright spatial regions and minimal values for dark spatial regions. | 03-03-2011 |
20110224975 | LOW-DELAY AUDIO CODER - The present invention relates to methods and devices for encoding and decoding digital audio signals, e.g. a speech signal. An audio coder and a decoder are provided wherein a modeller adds a first distribution model obtained from model parameters of past segments of the digital audio signal and a fixed distribution model, each of the models being multiplied by a weighting coefficient, for obtaining a combined distribution model. The weighting coefficients are selected to minimize a code length of a current segment of the digital audio signal. As the combined distribution model is a sum of several distribution models, wherein at least some of the models is based on the model parameters, flexibility is introduced in the signal model used to encode the digital audio signal. Thus, an audio coder and decoder providing a low bit rate in average, low bit rate variations and low error propagation are provided. | 09-15-2011 |