62results 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.

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 USE connection between analog processor and digital computer processor.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

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.

Methods for performing dynamic range compression (DRC) on audio in a manner intended to produce output audio for playback by systems or devices with limited power handling capabilities and preferably also to reduce or prevent undesirable artifacts (e.g., pumping and / or breathing) in the output audio. Some embodiments perform the DRC so as to maximize average loudness (while preventing loss of quieter elements) during playback, and also to reduce or prevent distortion. Other aspects are systems or devices configured to perform embodiments of the method. In some embodiments, reduced DRC is applied when average loudness of the input audio approaches (or matches or exceeds) a target (e.g., a knee point for DRC, or a signal level near to a maximum playback level of the intended playback system), since such input audio is assumed to have already been compressed, and otherwise applying full DRC to the input audio.

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.

A dynamic range control unit (210) configured to apply 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), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).

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.

Some noise compensation methods involve: receiving a microphone signal 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, the critical distance being a distance from the noise source location at which the directly propagated sound pressure equals the diffuse-field sound pressure. Some examples involve estimating whether the listener position is within the at least one critical distance, and performing the noise compensation method for the ambient noise based at least in part on the estimation of whether the listener position is within the critical distance.

The invention is related to signal processing methods for processing audio signals. Especially the invention concerns signal processing methods for adjusting the tone of music signals. The invention provides a signal processing method that provides an adaptive tonality control. In the method a shape dependent filter magnitude response is calculated, a level dependent filter magnitude response is calculated, a result response is calculated based on the shape and level dependent filter magnitude responses, and an output signal is calculated by convolving an input signal with the result response. In particular, the calculation of the shape dependent filter magnitude response comprises at least the steps of - calculating (110) a magnitude spectrum of a signal, - filtering (120) the resulting magnitude spectrum using a first smoothing filter, - applying (130) a predetermined weighting curve to the spectrum, - detrending (140) said weighted spectrum by fitting a curve to said weighted spectrum and subtracting said fitted curve from said weighted spectrum, and - inverting (150) the resulting spectrum to form the shape dependent filter magnitude response.

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.

An arithmetic device including at least a preamplifier and an equalizer executes a method for acoustic adjustment including: setting an initial value of a gain value of the preamplifier based on a difference between energy, to an input test sound source, of the test sound source before correction by the equalizer and energy, to the test sound source, of the test sound source after correction by the equalizer; amplifying a volume of the test sound source by the preamplifier using the initial value, calculating a loudness value and a clipping occurrence rate of the test sound source in a case where a frequency characteristic of the test sound source is corrected by the equalizer, and determining whether a calculation result of the loudness value is less than a first threshold and whether a calculation result of the clipping occurrence rate is less than a second threshold; and reducing the initial value of the gain value by a specified value in a case where the calculation result of the loudness value is equal to or greater than the first threshold or the calculation result of the clipping occurrence rate is equal to or greater than the second threshold.

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.

A computing device is configured to (i) receive an indication of an operational limit of a given playback device, where the given playback device is included in a group of at least two playback devices that are configured for synchronous playback of audio content, (ii) receive a first audio signal for playback by the group of at least two playback devices, (iii) based on the operational limit of the given playback device, apply a filter to the first audio signal to generate a second audio signal, and (iv) transmit the second audio signal to at least the given playback device for synchronous playback by the group of at least two playback devices.

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.