Abstract: A document printing system configured to print documents received from a user device. The system comprises a remote server and a first local print server. The remote server is configured to communicate with one or more local print servers located on respective local networks, including the first local print server. The remote server receives print jobs from remote user devices and to maintain a global queue of print jobs. The first local print server is coupled to a first local network and to an input device for receiving user instructions. The first local print server receives print jobs from user devices, maintains a first local queue of print jobs, processes print jobs from the global queue of documents, and instructs a printing device to print a document in accordance with a print job, based upon user instructions received at the input device.

Abstract: A folding board system comprising a plurality of board sections rotatably coupled via one or more hinge assemblies between respective board sections. In various embodiments, the folding board system is configured to assume a plurality of configurations including: a folded configuration wherein the plurality of board sections assume a stacked configuration and an assembled configuration wherein the plurality of board sections extend along a common axis.

Abstract: The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the

Abstract: The present invention relates to a telephony device comprising a near-mouth microphone (M1) for picking up an input acoustic signal including the speaker's voice signal (S1) and an unwanted noise signal (N1,D1), a far-mouth microphone (M2) for picking up an unwanted noise signal (N2,D2) in addition to the near-end speaker's voice signal (S2), said speaker's voice signal being at a lower level than the near-mouth microphone, and an orientation sensor for measuring an orientation indication of said mobile device. The telephony device further comprises an audio processing unit comprising an adaptive beamformer (BF) coupled to the near-mouth and far-mouth microphones, including spatial filters for spatially filtering the input signals (z1,z2) delivered by the two microphones, and a spectral post-processor (SPP) for post-processing the signal delivered by the beam-former so as to separate the desired voice signal from the unwanted noise signal so as to deliver the output signal (y).

Abstract: A device (1) for acoustic feedback compensation comprises an adaptive filter (4), an adjustment unit (5) for adjusting coefficients of the adaptive filter, a combination unit (3) for subtracting an adaptive filter signal from an input signal so as to produce a residual signal, and a noise unit (8) for producing a noise signal. The noise unit (8) is arranged for producing masked noise on the basis of the residual signal, a noise signal having a frequency spectrum controlled by the residual signal. An auditory masking model may be used to shape the noise spectrum. The noise signal may have a smaller amplitude than the residual signal at frequencies where the residual signal has a relatively large amplitude, and a larger amplitude than the residual signal at some frequencies where the residual signal has a relatively small amplitude.

Abstract: Disclosed is an echo canceller comprising two or more adaptive filters for calculating echo estimates, whereby the adaptive filters each have adaptation control mechanisms for applying individual update control criteria. The adaptive filters are arranged in series. Each of the adaptation control mechanisms of the adaptive filters may apply individual update control criteria for both direct echo and diffuse echo. Several step-size reduction strategies are presented.

Abstract: The relatively robust adaptive beamformer, comprises: a filtered sum beamformer (107) to process input audio signals (u1, u2, u3) from an array of respective microphones (101, 103, 105), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160); and a noise estimation e.g. when incorporated in a sidelobe canceller topology an adaptive noise estimator (150), arranged to derive a noise signal (y) which is subtracted from the first audio signal (z) to obtain a noise cleaned second audio signal (r), and further comprises a scaling factor determining unit (170) arranged to provide a scale factor (S) as a function of a ratio (Q) of the sidelobe canceling, and being arranged to scale the adaptation step size with the scale factor (S), so that the sidelobe canceller only adapts quickly if it is relatively well locked on the desired audio source, but is rather insensitive to interference from noise sources.

Abstract: A method for reconstructing a data spectrogram disturbed by noise and/or echo is described, wherein spectrogram data is subjected to an awarding of a reliability measure, and wherein the spectrogram data having a low reliability measure is replace by more reliable data. In particular the replacement is carried out by employing spectrogram data having a higher reliability measure as a means for selecting a code-book entry where said more reliable data is stored. Such a code-book is easy to implement, and this method avoids correlation calculations, inversions of matrices and limitations as to the specific types of used statistical models. The reconstruction method improves speech recognition results, which is important for voice controlled devices.

Abstract: An echo canceller (1) is described, which has dedicated non stationary echo cancelling properties. The non stationary echo could be estimated directly, or indirectly by subtracting the stationary echo determined by a stationary noise estimator. The echo canceller may comprise an adaptive filter (4) and a residual echo processor (6) coupled to the adaptive filter, which residual echo processor is equipped with the non stationary echo canceller. Such a non stationary echo canceller is proposed in order to prevent the stationary component in the echo estimate, especially in the residual echo from continuously distorting near end speech. This improves echo canceller performance in terms of speech quality and speech intelligibility, which is particularly important in cases of single talk near end speech, as opposed to double talk.

Abstract: An interference canceller is described, comprising an adaptive filter for modeling an interference, such as an echo or noise, and a spectral processor for processing the modeled interference together with near end speech and the interference. The interference canceller further comprises an interference model mismatch compensator coupled to the adaptive filter for providing a mismatch signal to the spectral processor, said mismatch signal showing a speech independent decay. With the model mismatch signal of the interference canceller the interference power spectrum can be estimated very accurately leading to a significant convergence improvement of the acoustic canceller. Especially in the initial convergence phase at the start of a communication session a high quality operation of the acoustic canceller is achieved, which is important as this determines the first quality impression of the user.

Abstract: An audio enhancement system is described, comprising audio signal inputs for a distorted desired signal and at least a reference signal, and a spectral processor coupled to the microphone array for processing the distorted desired signal by means of the reference signal acting as an estimate for the distortion of the desired signal. The spectral processor is arranged for modifying said processing such that the estimate for the distortion depends on A times the spectral power of the reference signal, where A is the ratio between the time averaged spectral power of the distortion of the distorted desired signal and the time averaged spectral power of the reference signal.

Abstract: An echo canceller comprises a signal input for a far end signal, an audio input for a distorted desired signal, an echo estimator coupled to the signal input, and a spectral subtracter coupled to the echo estimator and the audio input. The echo estimator further comprises digital filter means covering a time span of at least a part of the echo to be cancelled. Spectral subtraction of the echo part does not make use of echo phase information. Consequently this saves memory and processing power of calculations made in the echo canceller. Futhermore these calculations are not restricted to a particular decaying course of the room impulse response, as any kind of echo tail course may be modelled. This provides a larger degree of freedom in practical embodiments and broadens the application area of the echo canceller.