23results about "Combination controls" patented technology

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

In various embodiments, the disclosed techniques automatically change the audio equalization levels of audio content. When an audio content is requested for playback, the equalization manager of an audio system extracts the frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines sound pressure level (SPL) metrics of each frequency component. The equalization manager generates a custom equalization setting using gains calculated for each frequency band based on the SPL metrics. The equalization manager then adjusts the audio segment by applying the custom equalization setting to generate a modified segment (e.g., applying the calculated gains to each of the frequency bands). The modified segment is then output as audio to a user. The process then repeats for additional segments of the audio.

An audio signal processing method, and an audio signal processing device for executing the audio signal processing method includes, obtaining an audio signal and compressing the audio signal to generate a first compressed signal; calculating a duration for which a power value of the first compressed signal equals a first power threshold; determining whether the duration exceeds a first time threshold; when the duration exceeds the first time threshold, compressing the first compressed signal to generate a second compressed signal, and amplifying and outputting the second compressed signal.

In various embodiments, the disclosed technology automatically alters audio equalization levels of audio content. When a request to play audio content is made, an equalization manager of an audio system extracts frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines a sound pressure level (SPL) indicator for each frequency component. The equalization manager generates a customized equalization setting based on the SPL indicators using gains computed for each frequency band. The equalization manager then adjusts the audio segment by applying the customized equalization setting to generate a modified segment (e.g., applying the computed gains to each of the frequency bands). The modified audio segment is then output as audio to a user. The process is then repeated for additional segments of the audio.

The application relates to a high-isolation frequency distribution circuit based on common-base amplification, which comprises a gain control amplifier, a distributor, a common-base amplification network and a filter; the gain control amplifier realizes gain control amplification of an input frequency signal and attenuated input to the distributor; the distributor realizes power distribution and isolation of the frequency signal without additional noise, and realizes multi-way output, each way of output being connected with a common-base amplification network; the common-base amplification network improves the reverse isolation characteristic of the frequency signal through the common-base triode current buffer characteristic, realizes high-isolation between multi-way output frequency signals and inputs the frequency signals to the filter; the filter suppresses the out-of-band noise of the frequency signal to reduce the harmonic and the out-of-band noise of the frequency signal. The application can realize amplification, distribution, isolation and filtering of low-noise frequency signals, improves the integration of the output frequency signal of the time-homogeneous equipment, and has the advantages of high isolation, low additional noise and the like.

An arithmetic device executes a method for acoustic adjustment including: setting an initial value of a gain value of a preamplifier based on a difference between energies of the test sound source before and after correction by an equalizer to an input sound; 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 corrected by the equalizer, and determining whether calculation results of the loudness value and the clipping occurrence rate are each less than a predetermined 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 or the clipping occurrence rate is equal to or greater than the predetermined threshold.

A compression method for a hearing aid used by a hearing-impaired person is provided. The hearing aid comprises n frequency channels, each characterized by a linear gain and lower compression thresholds. The method determines compression parameters for each channel, including compression ratios, upper compression thresholds, and corresponding gains. For each channel, the upper compression threshold is set to the hearing-impaired person's annoyance threshold, adjusted by a parameter to convert from dB HL to dB SPL. The gain at the upper compression threshold is set to 0 dB. The compression ratio for each channel is calculated based on the linear gain, lower compression threshold, and annoyance threshold. The method allows for personalized compression settings tailored to the individual's hearing characteristics and comfort levels.

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Presented is a method (100) for real-time processing an audio stream in an audio speaker arrangement comprising at least one audio speaker. The method (100) comprises receiving (110) an audio-stream and a playback volume associated with the audio-stream and determining (120) a playback sound pressure level of the audio speaker based on the playback volume and pre-stored acoustic transfer data. The pre- stored acoustic transfer data is associating the playback volume with the playback sound pressure level of the audio speaker. The method further comprises generating (130) a compensation filter based on amplitude compensation data associated with the determined playback sound pressure level and filtering (140) the audio-stream using the compensation filter and applying a gain factor to the audio stream generating a compensated audio-stream. The gain factor is configured such that a total audio power of the audio stream is preserved regardless of the compensation filter. The compensated audio stream is provided (150) to the audio speaker (21). Additionally, an audio speaker arrangement and a computer program product are presented.

The embodiment of the present application provides a method and device for improving the sound quality of a loudspeaker, relates to the technical field of media, and can improve the low-frequency sound effect of the loudspeaker and improve the sound quality of the loudspeaker. The method comprises the following steps: performing frequency division on an input signal of the loudspeaker to obtain a first low-frequency input signal and a first high-frequency input signal, wherein the input signal is a time-domain signal; performing transient detection on the first low-frequency input signal to determine whether the first low-frequency input signal is a transient signal; if the first low-frequency input signal is a transient signal, performing signal envelope modulation on the first low-frequency input signal to obtain a second low-frequency input signal; and then determining an output signal of the loudspeaker according to the second low-frequency input signal and the first high-frequency input signal. The starting voltage of the second low-frequency input signal is greater than the starting voltage of the first low-frequency input signal, and the loudness of the second low-frequency input signal is greater than the loudness of the first low-frequency input signal.

To provide a speaker distortion correction device and speaker unit for identifying a non-linear parameter of a speaker with a comparatively simple configuration.SOLUTION: A speaker 2 includes a magnetic angle sensor 212 for detecting a change of a vibration system. A non-linear part correction filter 41 corrects an output distortion due to a non-linear parameter of the speaker 2. A transmission function of a linear inverse filter 42 is applied to eliminate the output distortion in a displacement of the vibration system detected with the use of the magnetic angle sensor 212. A control section 1 performs: driving the speaker 2 by a test signal; measuring a response from the displacement of the vibration system detected by using the magnetic angle sensor 212; defining a theoretical value of the non-linear parameter to be theoretically fixed by a design specification as an active non-linear parameter when an error between the response and a theoretical value of a response to be theoretically fixed by the design specification is smaller; and allowing a transmission function to correct an output distortion by the active non-linear parameter to be set in the non-linear part correction filter 41.SELECTED DRAWING: Figure 1

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Disclosed is a suspended audio device with bass enhancement performance, including a low-pass filter, a high-pass filter, an energy controller and a harmonic generator. The low-pass filter is configured to extract a low-frequency signal in an original input signal, and input the low-frequency signal as a fundamental wave of the harmonic generator; the high-pass filter is configured to extract a mid-high frequency signal in the original input signal; the harmonic generator is configured to process the low-frequency signal extracted by the low-pass filter through the NLD algorithm, and generate at least one enhanced harmonic signal in the low-frequency signal; the energy controller is configured to control an overall gain of the at least one enhanced harmonic signal generated by the harmonic generator.

The invention relates to a control unit for a vehicle for specifying a level of an audio signal for a loudspeaker unit (1) in the vehicle, comprising a module (3) for determining a target level for compensating ambient noise, and comprising a sound adjustment unit (5) for generating a desired sound image of the loudspeaker unit (1) by adjusting the audio signal, wherein a signal feedback (7) is provided from the sound adjustment unit (5) to the module (3), which transmits information about a current adjustment of the audio signal by the sound adjustment unit (5) to the module (3), and wherein the module (3) is designed to adjust the target level depending on the information about the current adjustment by the sound adjustment unit (5).

To provide a volume leveler controller and a controlling method.SOLUTION: In one embodiment, a volume leveler controller includes an audio content classifier for identifying a content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner on the basis of the identified content type. The adjusting unit may configured to positively correlate dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.SELECTED DRAWING: Figure 20

The invention relates to a control unit for a vehicle for specifying a level of an audio signal for a loudspeaker unit (1) in the vehicle, comprising a module (3) designed as a VNC module for determining a target level for compensating ambient noise, and comprising a sound adjustment unit (5) for generating a desired sound image of the loudspeaker unit (1) by adjusting the audio signal, wherein a signal feedback (7) is provided from the sound adjustment unit (5) to the module (3), which transmits information about a current adjustment of the audio signal by the sound adjustment unit (5) to the module (3), and wherein the module (3) is designed to adjust the target level depending on the information about the current adjustment by the sound adjustment unit (5).

This specification provides an audio processing device and method. The audio processing device includes a sound-emitting module and an audio processing module, the sound-emitting module includes N sound-emitting units, and each sound-emitting unit outputs an audio signal of a predetermined frequency band during operation; after obtaining initial audio data, the audio processing module performs frequency division processing on the initial audio data to obtain at least M sub-audio data of corresponding M sub-frequency bands, each sub-frequency band corresponds to the predetermined frequency band output by the corresponding sound-emitting unit among the N sound-emitting units, audio adjustment is performed on the corresponding M sub-audio data according to target audio characteristics of the M sub-audio data respectively to obtain M adjusted audio data, and each adjusted audio data is then input to the corresponding sound-emitting unit respectively. The technical solution of the embodiments of this specification can improve the sound quality effect of audio processing of multiple sound-emitting units.

The invention discloses a microwave high-power synthesis method based on four-path signal shunt adjustment and temperature compensation, which comprises the following steps of: 1, generating four paths of pre-amplified signals according to a low-power microwave signal; step 2, performing phase detection and amplitude detection on the four paths of pre-amplified signal branches, and performing temperature compensation and overshoot adjustment within a threshold value to generate phase and amplitude coordinated adjustment parameters; step 3, according to the multi-point temperature data of the key node of the synthetic link, calculating a temperature change quantity and comparing the temperature change quantity with a preset temperature change threshold value to generate a corrected phase amplitude cooperative adjustment parameter; step 4, performing power superposition on the corrected phase amplitude coordinated adjustment parameters in combination with a reverse power suppression mechanism to generate combined power data; and step 5, comparing the combined power data with a preset maximum power threshold value, and extracting amplitude fluctuation and phase deviation data based on a comparison result to generate an optimized trigger signal until the combined power data reaches the preset maximum power threshold value.

An audio system capable of applying equal-loudness correction that takes into account a speaker sensitivity characterization. The audio system includes a speaker, a microphone, and a processor coupled the microphone and the speaker. The processor is configured to calculate the speaker sensitivity characterization and apply equal-loudness correction that takes into account the speaker sensitivity characterization.