Abstract: A method conceals dropouts in one or more audio channels of a multi-channel arrangement. The method maps transmitted signals into a frequency domain during an error-free signal transmission of two or more channels. A magnitude spectra and spectral filter coefficients are derived. The spectral filter coefficients relate the magnitude spectrum of the audio channel to the magnitude spectrum of at least one other channel. When a dropout occurs, a replacement signal is generated through the filter coefficients and a substitution signal. The filter coefficients may be generated prior to the detection of the dropout.

Abstract: A transducer membrane enhances sound reproduction. Curved portions of the membrane periphery contribute to the enhanced sound reproduction. The transducer membrane may add or improve sound reproduction capability in cell phones, gaming systems, personal data assistants, or other devices.

Abstract: A transducer generates acoustic energy. The transducer is suitable for incorporation into any device that needs sound reproduction capability, including devices with generally rectangular geometries such as cell phones, PDAs, and portable gaming devices. The transducer includes a displaceable membrane with a deformable corner. The deformable corner may extend the frequency range over which the transducer generates acoustic energy without distortion. The deformable corner may be part of a membrane periphery around the displaceable membrane. The membrane periphery may be square, triangular, or may take any other polygonal shape.

Abstract: The invention relates to an array microphone with several individual microphones (1-4) connected with a signal processor (11) that for each individual microphone, comprises at least one digital filter, in particular for voice recognition.

Abstract: A method and an apparatus for filtering an audio signal include the steps of making the audio signal available in digitalized form, wherein the duration of the sampling interval is half or less than half of the period duration of the highest frequency to be expected in the audio signal, a digitalized impulse response is made available in accordance with the desired filtering effect, and a convolution sum is formed from the samples of the impulse response and the samples of the audio signal, wherein, (i) several adjacent samples of the impulse response define a corresponding interval within at least a time portion of the impulse response which is shorter than the impulse response, (ii) within the interval defined in this manner, the samples of the impulse response corresponding to the interval defined in this manner are equated to a value which is a function of one or more of the samples of the digitalized impulse response falling within the interval defined in this manner, and (iii) the steps (i) and (ii) are

Abstract: A method of simulating a room impression and/or sound impression occurring at a representative listening location in a room with monophonic, stereophonic or multichannel reproduction includes selecting a room whose sound is to be simulated. A location of a representative listening location is then determined. Subsequently, the corresponding room impulse response at least for one channel is determined at the representative listening location. A threshold value which exceeds over at least a portion of the duration of the determined room impulse response is determined for the determined room impulse response. By comparing the determined room impulse response with the threshold value, a reduced room impulse response is produced which within the portion of the duration of the determined room impulse response only includes those contents of the determined room impulse response in which a momentary amplitude is above the threshold value.