Abstract: Techniques of rendering sound for a listener involve producing, as the amplitude of each of the source driving signals, a sum of two terms: a first term based on a solution s† to the equation b=A·s, and a second term based on a projection of a specified vector ? onto the nullspace of A, ? not being a solution to the equation b=A·s. Along these lines, in one example, the first term is equivalent to a Moore-Penrose pseudoinverse, e.g., AH(AAH)?1·b. In general, any solution to the equation b=A·s is satisfactory. The specified vector that is projected onto the nullspace of A is defined to reduce the coherence of the net sound field. Advantageously, the resulting operator is both linear time-invariant and idempotent so that the sound field may be faithfully reproduce both inside the RSF and at a sufficient range outside the RSF to cover a human head.

Abstract: A method includes: receiving time instants of audio signals generated by a set of microphones at a location; determining a distortion measure between frequency components of at least some of the received audio signals; determining a similarity measure for the frequency components using the determined distortion measure; and processing the audio signals based on the determined similarity measure.

Abstract: Techniques of performing linear acoustic echo cancellation performing a phase correction operation on the estimate of the echo signal based on a clock drift between a capture of an input microphone signal and a playout of a loudspeaker signal. Along these lines, the existence of the clock drift, i.e., a small difference in the sampling rates of the input microphone signal and the loudspeaker signal, can cause processing circuitry in a device configured to perform LAEC operations to generate a filter based on the magnitudes of the short-term Fourier transforms (STFTs) of the input microphone signal and the loudspeaker signal. Such a filter is real-valued and results in a positive estimate of the acoustic echo signal included in the input microphone signal. The phase of this estimate may then be aligned with the phase of the input microphone signal.

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: 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: Existing post-filtering methods for microphone array speech enhancement have two common deficiencies. First, they assume that noise is either white or diffuse and cannot deal with point interferers. Second, they estimate the post-filter coefficients using only two microphones at a time, performing averaging over all the microphones pairs, yielding a suboptimal solution. The provided method describes a post-filtering solution that implements signal models which handle white noise, diffuse noise, and point interferers. The method also implements a globally optimized least-squares approach of microphones in a microphone array, providing a more optimal solution than existing conventional methods. Experimental results demonstrate the described method outperforming conventional methods in various acoustic scenarios.

Abstract: Existing post-filtering methods for microphone array speech enhancement have two common deficiencies. First, they assume that noise is either white or diffuse and cannot deal with point interferers. Second, they estimate the post-filter coefficients using only two microphones at a time, performing averaging over all the microphones pairs, yielding a suboptimal solution. The provided method describes a post-filtering solution that implements signal models which handle white noise, diffuse noise, and point interferers. The method also implements a globally optimized least-squares approach of microphones in a microphone array, providing a more optimal solution than existing conventional methods. Experimental results demonstrate the described method outperforming conventional methods in various acoustic scenarios.

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: 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: 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: 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: Provided are methods and systems for spatially selecting acoustic sources using a post-processor that consists of a selection of one postfilter from a set of postfilters, or a cascade of postfilters, where each postfilter is optimal for a particular scenario. Each postfilter individually is based on optimizing the gain for each time-frequency bin based on knowledge of (i) a spatial covariance matrix for the desired source, (ii) a spatial covariance matrix for the interfering sources, and (iii) microphone signals in some neighborhood of the current time-frequency bin.

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: 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.

Abstract: A system and a method for evaluating an acoustic transfer function, wherein the acoustic transfer function is a transfer function from one acoustic source to a reproduction area sampled by a limited number of microphone modules.

Abstract: 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.

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: 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.

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: Implementations of independent temporally concurrent video stream coding may include encoding a plurality of input frames from an input video sequence, wherein the plurality of input frames includes a first input frame. Encoding the plurality of input frames may include generating a first plurality of encoded frames based on the plurality of input frames such that the first plurality of encoded frames includes a first encoded I-frame corresponding to the first input frame, and generating a second plurality of encoded frames based on the plurality of input frames such that the second plurality of encoded frames includes a first encoded P-frame corresponding to the first input frame. Implementations of independent temporally concurrent video stream coding may include including the first plurality of encoded frames and the second plurality of encoded frames in an output, and transmitting the output to a decoder.