113results about "Digitally weighted transducing elements" patented technology

An audio path is constructed to include a multi-bit sigma-delta converter for converting an N-bit digital input to an n-bit output representing an over-sampled, lower resolution n-bit version of the N-bit digital input; a formatter for converting the n-bit output to an m signal output (e.g., as a thermometer code, a SDM format or a PWM format); an m-by-m switching matric for receiving the m output signals and for reordering the m output signals, m class-D drivers individual ones of which are driven by one of the reordered m output signals for driving one of m speakers; and a dynamic element matching (DEM) block coupled to the switching matric for controlling the reordering of the m output signals driving the m class-D drivers for spreading the distortion due at least to driver-speaker pair mismatch to wide band noise. The DEM may operate to generate white noise, or it may generate shaped (colored) noise.

A semiconductor die with an integrated electronic circuit, configured so as to be mounted in a housing with a capacitive transducer e.g. a microphone. A first circuit is configured to receive an input signal from the transducer at an input node and to provide an output signal at a pad of the semiconductor die. The integrated electronic circuit comprises an active switch device with a control input, coupled to a pad of the semiconductor die, to operatively engage or disengage a second circuit interconnected with the first circuit so as to operate the integrated electronic circuit in a mode selected by the control input. That is, a programmable or controllable transducer. The second circuit is interconnected with the first circuit so as to be separate from the input node. Thereby less noise is induced, a more precise control of the circuit is obtainable and more advanced control options are possible.

In a first aspect the present invention provides a compact high efficiency driver suitable for driving a loudspeaker, such as a miniature loudspeaker for mobile phones or hearing aids. The driver comprising an interface for receiving an input signal, a three-level modulator, and a three-level H-bridge. The interface may be adapted to receive a digital input signal. In a preferred embodiment of the driver the H-bridge is controlled by a correction circuit which is operated by a Return-To-Zero scheme. In a further preferred embodiment the driver comprises a supply voltage step-up circuit for increasing a voltage supplied to the H-bridge. In a second aspect the present invention provides a miniature loudspeaker assembly having a built-in driver. In a third aspect the present invention provides a mobile device with a built-in miniature loudspeaker assembly.

A wireless microphone attached to a ruggedized, mobile digital video recording system may be used to record audio of events when activated. To conserve file space in a storage device, a sensor may be used to activate the microphone for audio recording, and to cause video recording to begin at the same moment through a pairing mechanism between the wireless microphone and the mobile digital video recording system. This sensor activation provides a complete audio/visual record of the event, or series of events, that caused the sensor to trigger. The microphone may also be manually triggered, thus providing a user with a means to capture audio/visual records in an ad hoc fashion. All recorded events are saved to files in a high-capacity storage device for later retrieval and review.

The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a headliner speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Digital loudspeaker comprising a support, a plurality of first membranes suspended on the support, said first membranes being bistable, and said loudspeaker comprising actuator for each of the first membranes that can change each of the first membranes from a first stable state to a second stable state and vice versa, and a controller for controlling said first actuator.

A digital/analog conversion apparatus for converting a digital signal into an analog signal. The digital/analog conversion apparatus can generate a high-quality analog signal, even when elements configuring the digital/analog conversion apparatus have variance, with high resolution and a small circuit size. The data conversion apparatus is provided with a first data converter for reducing the number of bits of an input signal, a second data converter for converting the format of the first output signal, and a third data converter for conversion into a code which corresponds to the history of the output from the second data converter.

A digital sound system suitable for a digital speaker device for directly converting analog sound by a circuit using a ΔΣ modulator and a mismatch shaping filter circuit to output a plurality of digital signals and a plurality of speakers driven by the plurality of digital signals. A digital speaker driving device includes a ΔΣ modulator, a post filter, s driving circuits, and a power supply circuit to supply power to the ΔΣ modulator, the post filter and the s driving elements and the s driving circuits are adapted to s digital signal terminals.

The present invention provides for methods and systems for digitally processing an audio signal. In various embodiments, a method comprises receiving a profile comprising a plurality of filter equalizing coefficients, configuring a plurality of filters of a graphic equalizer with the plurality of filter equalizing coefficients from the profile, receiving a first signal for processing, adjusting the plurality of filters using a first gain, equalizing the first signal using the plurality of filters of the graphic equalizer, outputting the first signal, receiving a second signal for processing, adjusting the plurality of filters, previously configured with the filter equalizing coefficients from the profile, using a second gain, equalizing the second plurality of frequencies of the second signal with the plurality of filters of the graphic equalizer, and outputting the second signal.

The invention discloses a multi-bit Delta-Sigma modulation-based digital speaker system implementation method and device. The method comprises the following steps: 1) converting digital input formats; 2) performing oversampling interpolation filtering; 3) performing multi-bit Delta-Sigma modulation; 4) encoding a thermometer; 5) performing multi-channel mismatch-shaping; 6) converting encoding formats; 7) performing multi-channel digital power amplification; 8) driving a speaker array or a multi-voice coil speaker unit to phonate. The device comprises a digital input interface, an oversampling interpolation filter, a multi-bit Delta-Sigma modulator, a thermometer encoder, a multi-channel mismatch-shaper, an encoding format converter, a multi-channel digital amplifier, and a speaker array or multi-voice coil speaker unit which are sequentially connected. By adopting the method and the device, the high-power output under low-voltage power supply can be realized; the power consumption is reduced; the single chip integration of a multi-channel playback system is realized; the volume weight and the implementation cost of the system are reduced; the quality of playback voice is improved.

The present invention provides for methods and systems for digitally processing an audio signal and setting a peak signal threshold to a level that is determined to be safe, particularly for children or listeners with sensitive hearing. In various embodiments, a method comprises adjusting a gain of the signal based upon a received profile value minus a safety parameter, filtering the signal with a low shelf filter and/or high shelf filter, passing the signal through a first compressor that sets a peak threshold based upon the profile value minus the safety parameter, filtering the signal again with a low shelf filter and/or high shelf filter, processing the signal with a graphic equalizer, passing the signal through a second compressor that sets a peak threshold based upon the profile minus the safety parameter, and outputting the signal.

A wireless speaker system that transmits and receives both analog and digital audio signals simultaneously over a single channel at frequencies above 900 MHz. The present invention advantageously uses continuously variable slope delta (CVSD) modulation to perform analog-to-digital (A/D) and digital-to-analog (D/A) conversion of audio signals in the transmitter and receiver, respectively, using complementary filters and CVSD modulation circuits in both the transmitter and receiver. Analog signals are modulated in the transmitter with a high-frequency carrier and band-pass filtered before being combined with the digitized audio signal to modulate a high-frequency carrier, i.e. over 900 MHz, which is transmitted to the receiver. Similarly, the receiver band-pass filters and demodulates the received signal to recover the analog signal components.

The invention relates to a device for generating or recovering acoustic energy, including:

• • at least one first deformable cavity (20) for receiving an ambient atmosphere, made in a first substrate, delimited by at least one wall including at least one mobile or deformable wall (25), and a means for making the cavity communicate with an ambient atmosphere,
  • a means (24, 24′, 24₁, 24′₁) for actuating a displacement or deformation, in the plane of the sensor, or to recover energy resulting from a displacement or deformation, in the plane of the sensor, of said mobile or deformable wall.

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

The present disclosure refers to a digital microphone device providing a single-bit Pulse Density Modulation PDM output signal. The digital microphone comprises a microphone, arranged to convert an acoustic input signal into an analog electrical signal, and a preamplifier, having a variable gain, arranged to receive the analog electrical signal and to provide an amplified analog electrical signal, depending on the variable gain. The variable gain depends on a gain control signal. The digital microphone further comprises an Analog-to-Digital Converter block, arranged to receive the amplified analog electrical signal and to convert it into a respective digital signal; and a compensation block, arranged to receive the digital signal and to perform a digital operation on such digital signal, on the basis of a compensation signal, to generate a compensated signal. Furthermore, the digital microphone comprises an Automatic Gain Controller block 25, arranged to detect the digital signal and to generate said gain control signal, on the basis of the detected digital signal. The Automatic Gain Controller block is further arranged to generate the compensation signal, on the basis of the control signal, and to provide the compensation signal to the compensation block, to compensate a variation of the digital signal resulting from the variable gain of the preamplifier. Finally, the digital microphone device comprises a conversion block, arranged to receive the compensated signal and to convert it into the single-bit PDM output signal.

A digital microphone device outputs a digital audio signal to a digital signal processing system. The digital microphone device includes an acoustic transducer for producing an analog voltage signal representative of an acoustic signal, and an integrated circuit including an input multiplexer. The input multiplexer has a first analog input coupled to an output of the acoustic transducer, and a second analog input to be connected to an output of a remote external analog microphone providing an analog voltage signal. A variable gain analog signal pre-amplifier is coupled to an output of the input multiplexer.

A digital pulse-width-modulation (PWM) generator comprising:

• • an n bit digital magnitude comparator having first and second n bit inputs and an output indicative of the relative values of the signals applied at the first and second inputs;
  • a first n bit digital up/down counter having a count direction input coupled to receive a sign bit of a digital unary input signal, an n bit parallel binary count output connected to the first n bit input of the magnitude comparator, and a clock input;
  • a second n bit counter having a clock input coupled to receive a constant rate clock signal and an n bit parallel binary count output connected to the second n bit input of the magnitude comparator;
  • an AND gate having a first input coupled to receive the constant rate clock signal in frequency divided form and a second input coupled to receive a magnitude portion of the digital unary input signal, and further having an output connected to the clock input of the first counter; and
  • wherein the comparator continually generates an output signal indicative of the relative magnitudes of the counts of the first and second counters, whereby said output signal is a PWM output signal with an average value representing a ramp voltage having a slope determined by magnitude portion of the digital unary input signal with a direction of a slope of the output signal being determined by the polarity of the sign bit.

A wiring structure is provided for an electroacoustic transducer directly converting digital signals from a single sound source to sound waves without conversion to analogue signals. The structure includes a diaphragm, and a plurality of voice coils fixed to the diaphragm. The voice coils are connected to respective cables each consisting of a stranded pair of positive and negative input lines.

A digital microphone system, audio control device and control method thereof is related to the method for controlling a digital microphone circuit including receiving a clock signal; detecting that the clock signal is maintained at a predetermined level for a duration; when the duration reaches a given time, switching a transmission type of a data pin; in a data mode, outputting a digital audio signal to a data line via the data pin in an output type; and in a command mode, receiving a command signal from the data line via the data pin in an input type.

A digital microphone system and a method thereof are described. The digital microphone comprises a receiving unit, an amplifier, a modulator, an audio processing unit and a high definition audio link. After the receiving unit receives an analog audio signal, the analog audio signal is amplified by the amplifier. The analog audio signal is then converted into a first digital data by the modulator. The high definition audio link is set between the modulator and the audio processing unit for transmitting the first digital data. The audio processing unit comprises a digital filter for converting the first digital data into a second digital data. The audio processing unit and the digital filter are integrated and implemented by software.