Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: A system is provided to protect a loudspeaker (144) by controlling a level of an applied audio signal. A control signal is generated by applying an input audio signal (115) to the collective operations of a gain control system (100). The gain control system (100) uses the input audio signal (115) in conjunction with at least one parameter to derive an estimated stress associated with the loudspeaker (144). The estimated stress is compared with a protection threshold stress (127). If the protection threshold stress is exceeded, a gain applied by a gain component (134) is selectively adjusted to modify the input audio signal (115). The resulting gain-controlled audio signal (116) is employed to drive the loudspeaker (144).

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.

Abstract: Apparatus and method for determining an operating state of an earpiece of a personal acoustic device and/or the entirety of the personal acoustic device through tests to determine the current operating state, wherein the tests differ depending on a current power mode of the personal acoustic device, and wherein at least one lower power test is employed during at least one lower power mode.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: A front surround sound reproduction system which improves the performance of beam steering by using a speaker array arranged geometrically on two or more planes or on one curved surface, and a signal reproducing method of the system. The audio reproduction apparatus to reproduce a multi-channel audio signal by using a plurality of speakers includes a signal distribution unit to duplicate a multi-channel audio signal and to distribute the duplicated signals as one or more groups of multi-channel signals corresponding to one or more speaker array groups, a steering processing unit to form sound beams with steering angles predetermined in relation to each speaker array group, from the groups of multi-channel signals distributed by the signal distribution unit, and a speaker array unit having one or more speaker array groups to reproduce the sound beams of each group formed by the steering processing unit, in the speaker array group.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes skipping extension information included in an input bitstream, extracting a three-dimensional (3D) down-mix signal and spatial information from the input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using a down-mix signal obtained by the removal and the spatial information. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.

Abstract: An acoustic management system configured to compensate for acoustical dampening is provided. The system includes a microphone configured to detect a first acoustic signal from an acoustic environment and a logic circuit. The logic circuit detects an onset of the acoustical dampening between the acoustic environment and the microphone. The logic circuit generates an acoustic damping compensation filter, which is applied to the first acoustic signal to generate a drive signal.

Abstract: The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.

Abstract: A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

Abstract: A method and apparatus of generating a sound field effect is provided. The sound field effect generating apparatus may generate a frequency coefficient that is frequency-transformed from a direct signal, may generate a reflection signal from the frequency coefficient, may generate an output signal using the frequency coefficient and the reflection signal, and may perform an inverse-frequency transform of the output signal.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that can be applied to the entire broadened signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter allowing the reshaping the reconstituted high-frequency part and the correction of compression artifacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: There is disclosed a microphone, a circuit, and a method. A microphone capsule may include an electret microphone and a field effect transistor (FET). A floating DC voltage source may have a first end connected to a drain terminal of the electret microphone capsule and a second end. A load resistor may be connected between the second end of the floating DC voltage source and a source terminal of the electret microphone capsule. A voltage follower may have an output connected to the source terminal of the electret microphone capsule and the first end of the floating DC voltage source. A coupling capacitor may couple an audio signal from the source terminal of the electret microphone capsule to an input of the voltage follower.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.