In one aspect there is provided a method. The method may include receiving a first analog radio frequency signal including a signal of interest and an interference signal caused by a second analog radio frequency signal transmitted in full duplex over a channel from which the first analog transmission is received; combining the first analog radio frequency signal and a portion of the second analog radio frequency signal to generate an output analog radio frequency signal characterized by at least a reduction or an elimination of the interference signal included in the output analog radio frequency signal; and providing the output analog radio frequency signal. Related apparatus, systems, methods, and articles are also described.

Single channel full duplex wireless communications

Blind source separation (BSS) and independent component analysis (ICA) are generally based on a wide class of unsupervised learning algorithms and they found potential applications in many areas from engineering to neuroscience. A recent trend in BSS is to consider problems in the framework of matrix factorization or more general signals decomposition with probabilistic generative and tree structured graphical models and exploit a priori knowledge about true nature and structure of latent (hidden) variables or sources such as spatio-temporal decorrelation, statistical independence, sparseness, smoothness or lowest complexity in the sense e.g., of best predictability. The possible goal of such decomposition can be considered as the estimation of sources not necessary statistically independent and parameters of a mixing system or more generally as finding a new reduced or hierarchical and structured representation for the observed (sensor) data that can be interpreted as physically meaningful coding or blind source estimation. The key issue is to find a such transformation or coding (linear or nonlinear) which has true physical meaning and interpretation. We present a review of BSS and ICA, including various algorithms for static and dynamic models and their applications. The paper mainly consists of three parts: (1) BSS algorithms for static models (instantaneous mixtures); (2) extension of BSS and ICA incorporating with sparseness or non-negativity constraints; (3) BSS algorithms for dynamic models (convolutive mixtures).

/pdf/blind-source-separation-and-independent-component-analysis-a-3snno6yygi.pdf

Blind Source Separation and Independent Component Analysis: A Review

In this chapter, we provide an overview of existing algorithms for blind source separation of convolutive audio mixtures. We provide a taxonomy, wherein many of the existing algorithms can be organized, and we present published results from those algorithms that have been applied to real-world audio separation tasks.

/pdf/a-survey-of-convolutive-blind-source-separation-methods-1tb18ec3v1.pdf

A survey of convolutive blind source separation methods

In this paper, we present a new technique for the estimation of short-term linear predictive parameters of speech and noise from noisy data and their subsequent use in waveform enhancement schemes. The method exploits a priori information about speech and noise spectral shapes stored in trained codebooks, parameterized as linear predictive coefficients. The method also uses information about noise statistics estimated from the noisy observation. Maximum-likelihood estimates of the speech and noise short-term predictor parameters are obtained by searching for the combination of codebook entries that optimizes the likelihood. The estimation involves the computation of the excitation variances of the speech and noise auto-regressive models on a frame-by-frame basis, using the a priori information and the noisy observation. The high computational complexity resulting from a full search of the joint speech and noise codebooks is avoided through an iterative optimization procedure. We introduce a classified noise codebook scheme that uses different noise codebooks for different noise types. Experimental results show that the use of a priori information and the calculation of the instantaneous speech and noise excitation variances on a frame-by-frame basis result in good performance in both stationary and nonstationary noise conditions.

/pdf/codebook-driven-short-term-predictor-parameter-estimation-40rybocmy7.pdf

Codebook driven short-term predictor parameter estimation for speech enhancement

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

A method of reducing noise by cascading a plurality of noise reduction algorithms is provided. A sequence of noise reduction algorithms are applied to the noisy signal. The noise reduction algorithms are cascaded together, with the final noise reduction algorithm in the sequence providing the system output signal. The sequence of noise reduction algorithms includes a plurality of noise reduction algorithms that are sufficiently different from each other such that resulting distortions and artifacts are sufficiently different to result in reduced human perception of the artifact and distortion levels in the system output signal.

Method of cascading noise reduction algorithms to avoid speech distortion

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

Dual microphone noise reduction for headset application

A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem Individual aspects comprehended related to echo and noise reduction, and divergence control

Method and system for clear signal capture

A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.

Two microphone noise reduction system

An improved technique of co-channel speech separation, S-AADP/LMS, and its integration with automatic speech recognition is presented. The S-AADF/LMS technique is based on the algorithms of accelerated adaptive decorrelation filtering (AADP) and LMS noise cancellation, where a switching between the two algorithms is made depending upon the active/inactive status of the co-channel signal sources. The AADF improves the previous adaptive decorrelation algorithm in terms of system stability and estimation efficiency, and leads to better estimation of time-varying and reverberant channels. The S-AADF/LMS further improves the estimation accuracy when only one source signal remains active during certain periods of time. A coherence-function based source signal detection algorithm is also presented, which is successfully used in the switching between AADF and LMS and in extracting speech signals from leakage-corrupted background. Experiments were conducted under a simulated environment based on the measurements made of certain real room-acoustic conditions, and the results demonstrated the effectiveness of the proposed technique for co-channel speech separation and recognition.

/pdf/adaptive-co-channel-speech-separation-and-recognition-1e4anhor9p.pdf

Kuan-Chieh Yen

Papers

Method of cascading noise reduction algorithms to avoid speech distortion

Dual microphone noise reduction for headset application

Method and system for clear signal capture

Two microphone noise reduction system

Adaptive co-channel speech separation and recognition