32results about How to "Improve audio signal quality" patented technology

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.

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 processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

The present invention relates to a system for processing of an audio signal relating to a diver in a deep diving environment. The system comprises an input device structured to receive the signal, a pitch changing module configured to change the pitch of the received signal, a plurality of processing modules collectively configured to process the pitch changed signal and produce a processed signal. The present invention is further directed to a method for processing of an audio signal relating to a diver in a deep diving environment.

A system and method for improving hearing is disclosed. The system includes a microphone array that includes an enclosure, a plurality of beamformer microphones and an electronic processing circuitry to provide enhanced audio signals to a user by using information obtained on the position and orientation of the user. The system is preferably in the form of a smartphone having a retractable piece having the beamformer microphones mounted thereon. The system may also include a corresponding method for improving hearing and a non-transitory computer storage media having instructions stored thereon which, when executed, execute the method.

Wearable devices (100) capable of measuring a core body temperature and other vital signs of a user in a range of situations are described herein. The wearable device is arranged to be retained within the ear canal of the ear, in order to prevent the wearable device from inadvertently removing itself from the ear. Providing an infrared thermopile (101) at the innermost end of the ear insert ensures that the infrared thermopile is provided as close as possible to the tympanic membrane which will be used to provide an indication of the core body temperature. The device has an audio conduction channel (111) at least partly defined within an ear canal extending member (114), the audio conduction channel configured as a waveguide to conduct sound through a blocking member (212) to a distal portion of the ear insert.

An embodiment of the invention discloses an audio signal processing method and device. The method comprises steps as follows: to-be-processed audio signals generated in a direct broadcasting process are acquired, and audio frames are extracted from the to-be-processed audio signals; a first probability, a second probability and a third probability are determined according to a transcendental audio model; the first probability is the probability that the audio frame belongs to voice, the second probability is the probability that the audio frame belongs to music, and the third is the probability that the audio frame belongs to noise; if the first probability is smaller than a first threshold or the second probability is smaller than a second threshold and the third probability is larger than a third threshold, the fact that the audio frame contains noise is determined; after the fact that the to-be-processed audio signals contain noise is determined, the audio frame belonging to noise is denoised. The method and the device can be suitable for the application scene of direct broadcasting, denoising of the audio signals is realized, and the quality of the audio signals is improved.

The present invention relates to a sound device (SD1), connected to a computer (P1), for handling of asynchronously transferred digital audio packets (5) on a network (LAN1). The computer has an interface (3) connected to a telephony application (1), a driver (D3) and a bus (4). The sound device (SD1) is connected (9) via the bus (4) and includes a software frame buffer (B2), codecs (C2) and an A/D-D/A converter (AD2), which is connected to in/out devices (10, 11, 12). The sound packets (5) are transferred asynchronously through the computer (P1), are buffered in the sound device frame buffer (B2), decoded in the codec (C2) and D/A converted into an analog signal for the in/out devices. Speech to the in devices (11, 12) is processed in a corresponding manner. Having the buffer (B2) close to the codec (C2) enables processing of the sound packets, e.g. with respect to the varying time delay in the computer (P1), restoring lost packets and producing replacement frames. The sound device (SD1) relieves the computer (P1) of the heavy workload of processing the sound packets (5).

For the quantisation of spectral data in an audio transform encoder psycho-acoustic information is required, i.e. an approximation of the true masking threshold. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window/transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied.

The embodiment of the invention provides an anti-grounding loop noise circuit, an anti-grounding loop noise system and unbalanced audio equipment, which are applied to the unbalanced audio equipment and comprise a differential amplification circuit, the positive input end of the differential amplification circuit is connected with the signal end in an audio interface, the negative input end of the differential amplification circuit is connected with the grounding end in the audio interface, and the negative input end of the differential amplification circuit is connected with the signal end in the audio interface. The audio interface is an input interface of the audio equipment, and the differential amplification circuit is used for amplifying a differential audio signal sent by source end audio equipment and inhibiting ground loop noise. According to the embodiment of the invention, ground isolation between the two audio devices is realized from the perspective of hardware design, and balance and non-balance output ports of the audio device at the source end are compatible, so that additional adaptation is not needed. By adopting the technical scheme, the design of field wiring is greatly simplified, and the audio signal quality and the stability of an audio system are improved.

An electronic accessory apparatus and audible signal transmission method thereof are provided. The electronic accessory apparatus includes a type C USB connector, a first switch circuit, a second switch circuit, a grounding switch circuit and a first channel selector. The type C USB connector has a plurality of data pin sets, a first ground pin, and a second ground pin. The first switch circuit transports a data signal set or an audible output signal set to a first selected data pin set according to a first control signal. The second switch circuit transports the audible output signal set to the first selected data pin set according to the first control signal. The grounding switch circuit is turned on or cut off according to a second control signal. The first channel selector sets a signal transmission relation between the first audible input signal set and the first and second ground pins.

The present invention provides an audio processing unit having a microchip for processing an inputted audio signal. The microchip operates by duplicating the inputted audio signal, splitting the duplicated signal into at least two exact duplicates of the original inputted signal, layering the signals, processing the layers, and recombining the layers to provide an enhanced or processed audio signal. Thereafter, the processed audio file may be further processed by a user by utilizing any number of knobs, buttons, and the like disposed on the unit. Preferably, the unit is mountable on an open-frame rack.

An encoding system includes a sampling unit, a computing unit, a comparing unit, a quantifying unit, and an encoding unit. The sampling unit obtains first sample data of a current sampling point and second sample data of a previous sampling point. The computing unit computes a data difference between the first sample data and the second sample data. The data difference includes a numeral and a sign. The comparing unit determines whether the data difference is more than or equal to 0 and outputs a determining result. The quantifying unit quantifies the numeral of the data difference. The encoding unit encodes the numeral of the data difference with or without the sign according to the determining result.