33results about How to "Improve audio signal quality" patented technology

An audio signal system is provided having, in one embodiment, a magnetostrictive core of varying shapes, sizes, and permeability. The core is wound with a wire, thereby forming an inductor for producing a magnetic field when current is passed through the wire. A speaker, or other audio output device, is positioned in parallel or series with the inductor. In operation, an audio signal is received and transmitted to the speaker or other audio output device, wherein a replication of the received signal is produced. An output of the magnetostrictive inductor couples with the replicated audio signal to modify and enhance the quality of the projected signal. In yet another embodiment, digital signal processing may be used to modify the received signal as well.

An apparatus for calculating filter coefficients for a predefined loudspeaker arrangement has a multi-channel renderer. The multi-channel renderer calculates a filter coefficient for each loudspeaker of a virtual loudspeaker arrangement, being different from the predefined loudspeaker arrangement, based on a property (e.g. position or type) of a virtual source of an audio object to be reproduced by the predefined loudspeaker arrangement. Further, the multi-channel renderer determines an adapted filter coefficient for a loudspeaker of the predefined loudspeaker arrangement based on one or more calculated filter coefficients of one or more loudspeakers of the different virtual loudspeaker arrangement.

To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.

The anti-ground loop noise circuit, system, and unbalanced audio equipment provided by the embodiments of the present application are applied to unbalanced audio equipment, and include: a differential amplifier circuit, a positive input terminal of the differential amplifier circuit is connected to a signal terminal in an audio interface, The negative input terminal of the differential amplifier circuit is connected to the ground terminal in the audio interface, the audio interface is the input interface of the audio device, and the differential amplifier circuit is used to amplify the differential audio signal sent by the source audio device , and suppresses ground loop noise. In the embodiment of the present application, the ground isolation between the two audio devices is realized from the perspective of hardware design, and is compatible with the balanced and unbalanced output ports of the source audio device without additional adaptation. The adoption of the technical solution greatly simplifies the design of the field wiring, and improves the audio signal quality and the stability of the audio system.

A method of processing an audio signal is provided. The method includes receiving at least one digital audio stream from at least one audio source, applying at least one gain value to each of the at least one digital audio stream by a gain adjustment unit, mixing the at least one digital audio stream to which the at least one gain value has been applied, calculating a representative gain value from the at least one gain value, calculating an extra valid headroom based on the representative gain value for correction of the mixed at least one digital audio stream, and post-processing the mixed at least one digital audio stream using the extra valid headroom.

A method of calibrating a sound restitution assembly for a multichannel sound signal, which includes a plurality of loudspeakers. The method includes: obtaining multidirectional impulse responses of the loudspeakers to reproduction of a predetermined audio signal; analyzing the multidirectional impulse responses obtained, in a domain of spatio-temporal representation, over at least one time window encompassing the instants of arrival of the first reflections of the audio signal reproduced to determine a set of characteristics of the first reflections; comparing the amplitude of each of the reflections with a predetermined perceptibility threshold and identifying imperceptible reflections for which the amplitude is below the predetermined threshold; modifying the impulse responses obtained to obtain perceptive impulse responses, by deleting the reflections identified as imperceptible; and determining a filtering matrix on the basis of the perceptive impulse responses for an application of this filtering matrix to the multichannel audio signal before sound restitution.

The invention discloses a loudspeaker control method and device. The method comprises the following steps: acquiring an input digital signal corresponding to an input audio signal; partitioning the input digital signal to obtain a plurality of corresponding digital signal blocks; performing buffer processing on the digital signal block to obtain a corresponding buffer signal; and performing gain adjustment on the buffer signal based on preset loudspeaker parameters, so that the structural performance parameters of the loudspeaker can be controlled within a preset range, and the preset loudspeaker parameters are working state parameters when the loudspeaker works under preset conditions. According to the scheme, the quality of the audio signal output by the loudspeaker can be improved.