46results 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.

An audio system, including at least one approximating circuit, the at least one approximating circuit configured to apply an approximated gain to the input signal, the least one approximating circuit outputting an approximated signal; a limiter configured to reduce a magnitude of the input signal by a gain determined by an amount the magnitude of the approximated filtered signal exceeds a predetermined threshold, the limiter outputting a limited input signal; a filter being respectively configured to receive the limited input signal and to respectively apply a filter gain, the filters outputting a plurality of filtered signals, wherein the approximated gain is substantially similar to the filter gain of one of the plurality of equalization filters; and a speaker array comprising a plurality of speakers, each speaker configured to receive a respective filtered signal of the plurality of filtered signals and to transduce the filtered signal to an audio signal.

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.

Method for adapting an audio output to users of a vehicle, wherein the vehicle is designed with separate audio zones for different seating positions, in which - for different users of the vehicle, an individual set of parameters describing the frequency-dependent hearing ability of the respective user is determined (2); and - by means of the respective individual parameter set in the vehicle for multiple users, depending on the seating position, a suitable correction of the audio output signal is made by a frequency response and dynamic range adjustment for different frequency ranges (4).

A converter 2 includes a time window cut-out block 21, a fast Fourier transform (FFT) block 22, an attenuation amount limitation block 23, a quantization noise attenuation block 24, an overtone generation block 25, an inverse fast Fourier transform (IFFT) block 26, and a time window resynthesis block 27. The attenuation amount limitation block 23 determines the maximum attenuation amount of quantization noise to be attenuated in the quantization noise attenuation block 24 based on the magnitude of a signal level of sound data supplied from the time window cut-out block 21. The quantization noise attenuation block 24 adjusts amplitude in a frequency domain based on the maximum attenuation amount determined by the attenuation amount limitation block 23, to attenuate the quantization noise.

A method for improving sound quality of a speaker includes: performing frequency division on an input signal of the speaker to obtain a first low-frequency input signal and a first high-frequency input signal, where 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; determining an output signal of the speaker based on the second low-frequency input signal and the first high-frequency input signal, where a startup voltage of the second low-frequency input signal is higher than a startup voltage of the first low-frequency input signal, and loudness of the second low-frequency input signal is greater than loudness of the first low-frequency input signal.

The present application discloses a power amplifier control circuit and a radio frequency device, which are applied to the field of amplifier control technology to solve the problem in the prior art that the circuit structure and control logic of the power amplifier control circuit for realizing power control and gain control are relatively complex. Specifically: the mode switching module is used to determine the current control mode according to the mode selection signal. In the gain control mode, the first voltage is used as the attenuation voltage. In the power control mode, the power control module determines the second voltage that meets the power requirement according to the first voltage. The mode switching module uses the second voltage as the attenuation voltage. The attenuation module is used to adjust the attenuation amount of the radio frequency signal according to the attenuation voltage. The power amplifier module performs power amplification on the adjusted radio frequency signal. In this way, the mode switching module determines the working mode according to the mode selection signal and determines the attenuation voltage for power control or gain control, thereby simplifying the circuit structure and control logic.

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.

The present invention discloses an RF power amplifier, a chip, and a communication terminal. The RF power amplifier includes a power amplification circuit, an output matching circuit, a power detection circuit, and a bias comparison circuit. The output power of the main signal path is detected by the power detection circuit, and an equivalent voltage proportional to the output power is obtained and input to the bias comparison circuit. The value of the equivalent voltage is adjusted by the bias comparison circuit and compared with a control voltage to provide a bias voltage and / or collector voltage to the power amplification circuit. This forms a closed-loop circuit, allowing the RF power amplifier to maintain stable operation of gain and output power under different power levels.

The present application discloses a cochlear implant and a corresponding cochlear implant system, wherein the cochlear implant includes: at least one input transducer for capturing incoming sound and for generating an electrical audio signal in a frequency band representing the incoming sound; a sound processor configured to analyze and process the electrical audio signal; a transmitter for transmitting the processed electrical audio signal; a receiver / stimulator that receives the processed electrical audio signal from the transmitter and converts the processed electrical audio signal into electrical pulses; an electrode array suitable for implantation in the cochlea, comprising a plurality of electrodes for stimulating the cochlear nerve with the electrical pulses; and a control unit configured to control the distribution of the electrical pulses to the plurality of electrodes; wherein the control unit is configured to distribute the electrical pulses to the plurality of electrodes by applying one of a plurality of different encoding schemes, and wherein the encoding scheme applied is selected according to characteristics of the incoming sound.

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.

An audio processing device and method is provided. 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 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.

The embodiments of this specification provide a signal processing circuit, method, equalizer, and communication device for an equalizer, which are applied to the field of wireless communication technology. The circuit includes a power processing module that separates the radio frequency signal to obtain a first input signal and a second input signal, and outputs the first input signal and the second input signal to a delay module; the delay module adjusts the signal delay information between the first input signal and the second input signal, and outputs the adjusted first input signal and the adjusted second input signal to a phase shift module; the phase shift module performs phase shift processing on the adjusted first input signal and the adjusted second input signal according to the interactive frequency point, and outputs the phase-shifted first input signal and the phase-shifted second input signal to the power processing module; the power processing module synthesizes the phase-shifted first input signal and the phase-shifted second input signal to obtain an RF signal after amplitude adjustment and equalization. This achieves adjustable amplitude equalization of the RF signal.

An audio signal processing device (100) includes: an obtainer (110) that obtains an audio signal (AS1); a first signal processor (120a) that generates, based on the audio signal (AS1), a first high-frequency signal including a high-frequency component in a frequency band higher than a first frequency higher than a human hearing range; and an adder (130) that adds the first high-frequency signal to the audio signal (AS1) to generate an expanded audio signal (AS1) having an expanded high-frequency range. The first signal processor (120a) generates the first high-frequency signal by performing: (i) first signal processing including first clipping processing of clipping the audio signal (AS1) at a first amplitude; and (ii) first high-pass filter processing of passing an audio signal (AS3) obtained by the first clipping processing through the frequency band higher than the first frequency.

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.

To provide an audio track analysis technique for supporting personalization of an audio system.SOLUTION: According to embodiments of the present invention, a system for and a technique for allowing personalization of an audio system are disclosed. This technique has a step of specifying audio personal settings for an audio category, a step of specifying one or more audio properties of an audio track, a step of selecting a first part of a representative audio track of the audio category based on the one or more audio properties, a step of reproducing the first part of the audio track for a user, and a step of adjusting the personal settings for the user based on a user input upon reproduction of the first part of the audio track.SELECTED DRAWING: Figure 3

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 method and a system for performing dynamic range compression (DRC) on audio in a manner intended to reduce or prevent undesirable artifacts in the output audio, such as pumping and breathing.SOLUTION: A DRC system performs DRC to maximize the average loudness and reduce or prevent distortion during playback. When the average loudness of the input audio approaches, meets, or exceeds a target, knee point for DRC or a signal level near the maximum playback level of the intended playback system, then it is assumed that such input audio is already compressed and therefore applies reduced DRC or full DRC to the input audio.SELECTED DRAWING: Figure 2

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