Abstract: In a method for reconstructing a source signal, which is encoded by a set of at least two descriptions, the method comprises: receiving a subset of the set of descriptions; reconstructing a reconstructed signal at an operating bitrate of a set of operating bitrates upon the basis of the subset of descriptions, the reconstructed signal having a second probability density, wherein the second probability density comprises a first statistical moment and a second statistical moment; and manipulating the reconstructed signal, wherein the reconstructed signal is manipulated such that, irrespective of the operating bitrate, a predetermined minimum similarity between a first statistical moment of a third probability density and a first statistical moment of a first probability density and between a second statistical moment of the third probability density and a second statistical moment of the first probability density is maintained.

Abstract: Provided are methods and systems for improving the intelligibility of speech in a noisy environment. A communication model is developed that includes noise inherent in the message production and message interpretation processes, and considers that these noises have fixed signal-to-noise ratios. The communication model forms the basis of an algorithm designed to optimize the intelligibility of speech in a noisy environment. The intelligibility optimization algorithm only does something (e.g., manipulates the audio signal) when needed, and thus if no noise is present the algorithm does not alter or otherwise interfere with the audio signals, thereby preventing any speech distortion. The algorithm is also very fast and efficient in comparison to most existing approaches for speech intelligibility enhancement, and therefore the algorithm lends itself to easy implementation in an appropriate device (e.g., cellular phone or smartphone).

Abstract: An audio rendering system is provided that comprises a plurality of loudspeakers arranged to approximate a desired spatial sound field within a predetermined reproduction region, wherein the loudspeakers are configured to approximate the sound field based on a weighted series of orthonormal basis functions for the reproduction region.

Abstract: 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.

Abstract: Provided are methods and systems for concealing missing segments and/or discontinuities in an audio signal, thereby restoring the continuity of the signal. The methods and systems are designed for and targeted at audio signals, are based on interpolation and extrapolation operations for sinusoids, and do not rely on the assumption that the sinusoids are harmonic. The methods and systems are improvements over existing audio concealment approaches in that, among other advantages, the methods and systems facilitate asynchronous interpolation, use an interpolation procedure that corresponds to time-domain waveform interpolation if the signal is harmonic, and have a peak selection procedure that is effective for audio signals.

Abstract: In a method for reconstructing a source signal, which is encoded by a set of at least two descriptions, the method comprises: receiving a subset of the set of descriptions; reconstructing a reconstructed signal at an operating bitrate of a set of operating bitrates upon the basis of the subset of descriptions, the reconstructed signal having a second probability density, wherein the second probability density comprises a first statistical moment and a second statistical moment; and manipulating the reconstructed signal, wherein the reconstructed signal is manipulated such that, irrespective of the operating bitrate, a predetermined minimum similarity between the first statistical moment of the third probability density and the first statistical moment of the first probability density and between the second statistical moment of the third probability density and the second statistical moment of the first probability density is maintained.

Abstract: 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.

Abstract: Methods and systems are provided for separating signal-correlated and signal-uncorrelated error components in quantization noise. Such separation leads to a generalization of the conventional rate-distortion optimization problem. For the commonly used assumption of a Gaussian process, a quantizer according to this principle is implemented in a straightforward manner using a dithered quantizer and appropriate pre-filters and post-filters. If the penalization of the signal-uncorrelated error component is increased over that of the signal-correlated error component, then the pre-filter emphasizes the signal spectrum more, reducing the differential entropy rate of the pre-filtered signal. Accordingly, the signal-uncorrelated noise is reduced for a given rate.

Abstract: Methods and systems are provided for implementing a distributed algorithm for beam-forming (e.g., MVDR beam-forming) using a message-passing algorithm. The message-passing algorithm provides for computations to be performed in a distributed manner across a network, rather than in a centralized processing center or “fusion center”. The message-passing algorithm may also function for any network topology, and may continue operations when various changes are made in the network (e.g., nodes appearing, nodes disappearing, etc.). Additionally, the message-passing algorithm may minimize the transmission power per iteration and, depending on the particular network, also may minimize the transmission power required for communication between network nodes.

Abstract: Provided are methods and systems for calibrating a distributed sensor (e.g., microphone) array using time-of-flight (TOF) measurements for a plurality of spatially distributed acoustic events at the sensors. The calibration includes localization and gain equalization of the sensors. Accurate measurements of TOFs are obtained from spatially distributed acoustic events using a controlled signal emitted at known intervals by a moving acoustic source. A portable user device capable of playing out audio is used to produce a plurality of acoustic events (e.g., sound clicks) at known intervals of time and at different, but arbitrary locations based on the device being moved around in space by a user while producing the acoustic events. As such, the times of the acoustic event generation are known, and are spatially diverse. The calibration signals emitted by the acoustic source are designed to provide robustness to noise and reverberation.

Abstract: The invention relates to an estimator for estimating a probability distribution of a quantization index generated from a source coder encoding a source signal, into a sequence of quantization indices, the source signal being described by a signal model, the source coder providing a current quantization index and current side information, the estimator being configured to obtain auxiliary parameters based on a configuration of the source coder and the current available side information and the signal model, the estimator being further configured to adaptively update the probability distribution of a quantization index upon the basis of a probability density function relating to a previous state of the estimator, the auxiliary parameters, the current quantization index and the current side information.

Abstract: Methods and systems for using pitch predictors in speech/audio coders are provided. Techniques for optimal pre- and post-filtering are presented, and a general result that post-filtering is more effective than pre-filtering is derived. A practical paired-zero filter design for the low-rate regime is proposed, and this design is extended to handle frequency-dependent periodicity levels. Further, the methods described provide a general performance measure for a post-filter that only uses information available at the decoder, thereby allowing for the optimization or selection of a post-filter without increasing the rate.

Abstract: 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.

Abstract: 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

Abstract: 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.

Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state 715 located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation 745 of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: 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.

Abstract: 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.

Abstract: The invention relates to an estimator for estimating a probability distribution of a quantization index generated from a source coder encoding a source signal, into a sequence of quantization indices, the source signal being described by a signal model, the source coder providing a current quantization index and current side information, the estimator being configured to obtain auxiliary parameters based on a configuration of the source coder and the current available side information and the signal model, the estimator being further configured to adaptively update the probability distribution of a quantization index upon the basis of a probability density function relating to a previous state of the estimator, the auxiliary parameters, the current quantization index and the current side information.

Abstract: Methods and devices for encoding and decoding are provided. A source signal value is encoded by a quantization index determined using a partition into quantization cells. Decoding of the quantization index takes place by sampling a reconstruction probability distribution, thereby obtaining a reconstructed signal value, such that the reconstructed signal value lies in the same quantization cell as the source signal value. In one embodiment, encoding and decoding are such that their succession preserves the source signal distribution. In another embodiment, the partition and the reconstruction probability distribution are determined in such manner that the quantization error is minimized subject to a constraint on the relative entropy between the source signal and the reconstructed signal.