32results about "Combination control in untuned amplifier" patented technology

Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Apparatus and methods for generating an encoded audio bitstream, including by including program loudness metadata and audio data in the bitstream, and optionally also program boundary metadata in at least one segment (e.g., frame) of the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, e.g., including by performing adaptive loudness processing of the audio data of an audio program indicated by the bitstream, or authentication and / or validation of metadata and / or audio data of such an audio program. Another aspect is an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

A method for control and transfer of audio between analog and computer in digital audio processing including one or a combination of a. controlling data by converting to-and-from analog to control the analog circuit and b. time correction in analogue and digital audio processing with a communication line using USB connection between analog processor and digital computer processor

Various embodiments are disclosed for (possibly simultaneously) applying EQ and DRC to audio signals. In an embodiment, a method comprises: dividing an input audio signal into n frames, where n is a positive integer greater than one; dividing each frame of the input audio signal into Nb frequency bands, where Nb is a positive integer greater than one; for each frame n: computing an input level of the input audio signal in each band f, resulting in a input audio level distribution for the input audio signal; computing a gain for each band f based at least in part on a mapping of one or more properties of the input audio level distribution to a reference audio level distribution computed from one or more reference audio signals; and applying each computed gain for each band f to each corresponding band f of the input audio signal.

A system and method of controlling loudness of an electroacoustic transducer may include: receiving a transfer function data element representing a transfer function between (a) electrical input and (b) sound pressure level (SPL) output of the electroacoustic transducer; applying the transfer function on an incoming electrical signal, to obtain an expected SPL signal, representing expected SPL of the electroacoustic transducer in response to the incoming electrical signal; identifying at least one fundamental acoustic tone in the expected SPL signal; producing at least one electrical compensation signal, corresponding to an acoustic harmonic of the at least one identified fundamental acoustic tone; and controlling the loudness of the electroacoustic transducer, based at least in part on the at least one electrical compensation signal.

A magnetic audio distortion device is provided which produces consistent levels of magnetic flux, non-linear hysteresis, or saturation across the audio spectrum using soft magnetic materials in electromagnetic components such as, for example, transformers or inductors. The device comprises an audio signal connected to an amplified electromagnetic network comprising an electromagnetic component capable of creating non-linear hysteresis and a series resistance. The output of the amplified electromagnetic network is approximately the derivative of the magnetic flux in the electromagnetic component. The output of the network is integrated, producing the magnetic flux in the magnetic core. The output of the system is magnetically distorted consistently across all audio frequencies. The magnetic audio distortion device provides the sonic characteristic of magnetic audio tape without requiring mechanical apparatus or hard magnetic materials enabling the sonic characteristics of magnetic tape in a compact and reliable form.

Some noise compensation methods involve receiving microphone signals corresponding to ambient noise from a noise source location in or near an audio environment, determining or estimating a listener position in the audio environment and estimating at least one critical distance, which is a distance from the noise source location at which directly propagated sound pressure is equal to diffuse field sound pressure. Some examples involve estimating whether the listener position is within the at least one critical distance and implementing a noise compensation method for the ambient noise based, at least in part, on an estimate of whether the listener position is within the critical distance.

Distortion sensing, distortion prevention, and distortion aware bass enhancement are disclosed. Systems and methods that provide distortion sensing, prevention, and / or distortion aware bass enhancement in an audio system can be implemented in a variety of applications. Sensing circuitry can generate statistics based on a received input signal for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide an input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to a bass enhancement circuit to operatively provide a parameter to the bass enhancement circuit. The bass enhancement circuit can provide a bass enhanced signal for generation of the acoustic output based on the parameter. Various combinations of bass enhancement circuits and multi-notch filters using statistics from the sensing circuitry can be implemented to provide an enhanced acoustic output.

Systems and methods for controlling loudness of an electro-acoustic transducer may include receiving transfer function data elements representing a transfer function between (a) an electrical input and (b) an output sound pressure level (SPL) of the electro-acoustic transducer, applying the transfer function to the incoming electrical signal to obtain an expected SPL signal representing an expected SPL of the electro-acoustic transducer in response to the incoming electrical signal, identifying at least one acoustic fundamental in the expected SPL signal, generating at least one compensation electrical signal corresponding to an acoustic harmonic of the identified at least one acoustic fundamental, and controlling loudness of the electro-acoustic transducer based at least in part on the at least one compensation electrical signal.

The present application is applicable to the technical field of audio processing, and provides an audio compression method and an audio compression apparatus. The method includes: obtaining a first audio signal to be processed; performing, by using a spectrum compensation filter, filtering processing on the first audio signal based on a preset target spectrum to obtain a second audio signal, where the target spectrum is correlated with a distorted frequency band of the audio playing device; using a preset dynamic range compressor (DRC) to process the second audio signal to obtain a compression coefficient; and outputting a compressed signal according to the compression coefficient. The method may expand a compression range, reduce an occurrence rate of distortion when playing the audio signal and improve a playing quality of the audio signal, without significantly reducing the gain of the loud sound frequency band.

A digital power amplifier to drive an RFID antenna with an antenna signal of a substantial sinusoidal output current. This digital power amplifier includes a switching element built to switch at least one of the 2*W driver blocks from a contributing mode into a none-contributing mode, in which none-contributing mode the driver block does not contribute with an increment of charge to the output current to adjust the amplitude and / or the waveform of the output signal.

A hearing device, e.g. a hearing aid, comprises a) a forward path comprising al) an input transducer for providing an electric input signal representing sound in the environment of the hearing device, the input transducer providing an input gain Gi, a2) a signal processor for processing said electric input signal or a signal based thereon and providing a processed signal, the signal processor comprising a compressor for determining a frequency and level dependent desired compressor gain GP to compensate for a hearing impairment of the user, and to provide a resulting compressor gain G'P, a3) an output transducer for providing output stimuli perceivable as sound for the user based on said processed signal, the output transducer providing an output gain, Go. A resulting forward path gain G' is defined in a logarithmic representation as GI+G'P+GO. The hearing device further comprises b1) a loop gain estimator for continuously estimating a current loop gain ΔL(n), configured to provide a loop gain estimate within a predefined number of feedback loop delays after a feedback buildup has started, wherein the loop gain estimate is calculated as the current level of a signal of the forward path at time index n minus the level of the same signal one feedback loop earlier, and b2) a loop gain controller for dynamically controlling said resulting forward path gain G' in dependence of said estimate of said current loop gain ΔL(n). The feedback loop is represented by the electric forward path of the hearing device from the input transducer to the output transducer and an acoustic feedback path from the output transducer to the input transducer, the feedback path exhibiting a feedback gain H. A resulting loop gain, LG', is determined as a sum of the resulting forward path gain G' and the feedback gain H when given in a logarithmic representation. The loop gain controller is configured to provide that the resulting loop gain is limited to stay below a predefined value. A method of operating a hearing device is further disclosed.

A system and method for computerised audio processing having an input 11 and output 27. An audio signal is received and frequency bands are designated through a splitter or filter 12 into low / bass 14 and high 13 frequencies, with the bass band including a fundamental frequency of the audio signal. Harmonics of the fundamental frequency are then generated 17 using integer multiples. The system / method will dynamically measure the input and adapt energy 19 of the generated harmonics 29 based on the input. The harmonics at the dynamically adapted energy are then combined with a raw or processed input signal 23 to produce a bass-enhanced output. The method may comprise dynamically computing audibility of distortion based on measured input and adapt energy of the harmonics accordingly. The method may perform comparison of energy in the bass frequency to energy in the higher frequency and decrease generated harmonic energy accordingly, or do the same using a determined threshold.

A speaker has a magnetic angle sensor that detects the displacement of the vibration system of the speaker. A non-linearity compensation filter compensates for output distortion due to a non-linear parameter of the speaker. A transfer function of a linear inverse filter is applied so that output distortion is eliminated from the displacement, detected by the magnetic angle sensor, of the vibration system. A controller drives the speaker by using a test signal, and measures a response from the detected displacement of the vibration system. If error between the measured response and the theoretical value, theoretically determined from the design specifications, of the response is small, the controller uses the theoretical value, theoretically determined from the design specifications, of the non-linear parameter as an active non-linear parameter and sets, in the non-linearity compensation filter, a transfer function that compensates for distortion in an output due to the active non-linear parameter.

Noise compensation method comprising: (a) receiving a content stream including content audio data; (b) receiving first microphone signals from a first device; (c) detecting ambient noise from a noise source location in or near the audio environment; (d) causing a first wireless signal to be transmitted from the first device to a second device, the first wireless signal including instructions for the second device to record an audio segment; (e) receiving a second wireless signal from the second device; (f) determining a content stream audio segment time interval for a content stream audio segment; (g) receiving a third wireless signal from the second device, including a recorded audio segment captured via a second device microphone; (h) determining a second device ambient noise signal at the second device location; and (i) implementing a noise compensation method for the content audio data based, at least in part, on the second device ambient noise signal.

Systems and methods to provide personalized audio streaming and rendering include receiving a cinematic audio track for selected content and a maximum accessible audio track for the content, and adjustably combining the cinematic audio track and accessible audio track to provide an improved dialogue audio track which allows a user / listener to hear the dialogue over an environmental noise floor, the combining being provided by a cross-fade renderer adjusted manually by the user / listener or automatically adjusted based on a measured noise floor, and optionally providing a personalized equalizer for hearing impairments. It also allows a content provider to create the maximum accessible audio track for a given content separate from the user / listener, which may also use a X-Fade renderer to verify quality of the accessible audio track.

This document describes a dynamic range control unit (210) configured to apply a dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to: downsample a subband signal (212) derived from the audio signal (211) to provide a downsampled subband signal (321); determine a DRC gain (329) based on the downsampled subband signal (321); and apply the DRC gain (329) to the subband signal (212) to provide a compressed subband signal (213) of a compressed audio signal (214).

A method for performing loudness normalization, the method being performed by an audio signal processing device, includes the steps of receiving an audio signal; receiving information about an integrated loudness of the audio signal; receiving information about a target loudness of the audio signal; correcting the integrated loudness based on one or more processing steps to obtain a corrected integrated loudness; and normalizing the audio signal based on the corrected integrated loudness and the target loudness to obtain a normalized audio signal.

An audio processor including a filter configured to separate in the input audio signal a bass frequency band including a fundamental frequency of the input audio signal. A harmonics generator is configured to generate audio harmonics of the fundamental frequency. A measurement module is configured to measure an audio feature of the input audio signal and output accordingly a measured audio feature. A gain adjustment module is configured to input the measured audio feature of the input audio signal and determine at least in part therefrom, an adapted energy of the audio harmonics generated by the harmonics generator. A combiner is configured to combine the audio harmonics at the adapted energy with the input audio signal or a processed version of the input audio signal, to produce a bass-enhanced output audio signal at the audio output.

A DCP server for use with a cinema processor for playing high quality live streaming content from a venue, such as a concert, and distributing it for playing in a movie theater is disclosed. The DCP server automatically adjusts the volume and the equalization outputs of the speakers in the movie theater to the industry standard and controls the volume setting of the cinema processor.

The application provides a debugging method of a multi-port power amplifier, comprising the following steps: S1, closing all HPA branches; S2, selecting two channels in the MPA for testing, one channel as a measurement input port to receive an uplink input sweep signal, and the other channel corresponding to an output port as a measurement output port to transmit a downlink isolated output signal, the downlink isolated output signal is received by a spectrum analyzer, and the level of the downlink received signal is recorded, then all HPA branches are opened one by one for testing, and the level of the downlink received signal is recorded; S3, opening two HPA branches with a theoretical phase difference of 180 degrees on the output port to transmit an uplink sweep signal, so that the two HPA branches work in a linear region, keeping the phase shifter gear of one HPA branch unchanged, observing the output signal level, adjusting the phase shifter gear of the other HPA branch to make the output signal level lowest, and recording the phase shifter gears of the two HPA branches, and then adjusting the phase shifter gears of all HPA branches one by one.