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System and method for dynamic range control of an audio signal
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.
Owner:DOLBY LABORATORIES LICENSING CORP +1
METHOD AND DEVICE FOR PROCESSING INFORMATION, FINISH DEVICE AND STORAGE MEDIUM
Owner:BEIJING XIAOMI MOBILE SOFTWARE CO LTD
Audio system with limited array signals
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.
Owner:BOSE CORP
Techniques for automatic equalization of audio
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.
Owner:HARMAN BECKER AUTOMOTIVE SYST GMBH
Audio signal processing method and audio signal processing device
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.
Owner:TYMPHANY HK LTD
Audio automatic equalization technique
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.
Owner:HARMAN BECKER AUTOMOTIVE SYST GMBH
Dynamic Range Compression with Reduced Artifacts
A method for generating output audio for playback by a system or device with limited power processing capabilities, preferably performing dynamic range compression (DRC) on the audio in a manner intended to reduce or prevent undesirable artifacts (e.g., pumping and / or breathing) in the output audio. Some embodiments perform DRC to maximize average loudness during playback (while preventing the loss of quieter elements) and reduce or prevent distortion. Another aspect is a system or device configured to perform embodiments of the method. In some embodiments, if the average loudness of the input audio approaches (or meets or exceeds) a target (e.g., a knee point for DRC, or a signal level near the maximum playback level of the intended playback system), reduced DRC is applied because such input audio is assumed to be already compressed. Otherwise, full DRC is applied to the input audio.
Owner:DOLBY LABORATORIES LICENSING CORP
A high-isolation frequency distribution circuit and distribution method based on common-base amplification
ActiveCN114465587BAmplifier modifications to reduce noise influenceAmplifier input/output impedence modificationLow noiseCommon base
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.
Owner:THE FIFTH RES INST OF TELECOMM SCI & TECH CO LTD
Method for acoustic adjustment using a loudness value and a clipping occurrence rate of a test sound source
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.
Owner:PANASONIC INTELLECTUAL PROPERTY MANAGEMENT CO LTD
Method and device for adapting an audio output to the user of a vehicle, as well as motor vehicle
ActiveDE102018213761B4Gain controlHearing impaired stereophonic signal reproductionEngineeringElectrophonic hearing
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).
Owner:VOLKSWAGEN AG
Signal processing device, control method, program and storage medium
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.
Owner:PIONEER IP
Audio signal processing device, audio signal processing method, and recording medium
An audio signal processing device includes: an obtainer that obtains an audio signal; a first signal processor that generates, based on the audio signal, 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 that adds the first high-frequency signal to the audio signal to generate an expanded audio signal having an expanded high-frequency range. The first signal processor generates the first high-frequency signal by performing: (i) first signal processing including first clipping processing of clipping the audio signal at a first amplitude; and (ii) first high-pass filter processing of passing an audio signal obtained by the first clipping processing through the frequency band higher than the first frequency.
Owner:PANASONIC INTELLECTUAL PROPERTY MANAGEMENT CO LTD
Method and apparatus for improving sound quality of speaker
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.
Owner:HUAWEI TECH CO LTD
A power amplifier control circuit and radio frequency device
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.
Owner:GUANGZHOU STARWAY COMM TECH +1
Compression method for hearing aids
PendingUS20260136141A1Volume compression/expansion in untuned/low-frequency amplifiersSets with customised acoustic characteristicsElectrophonic hearingFit hearing aid
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.
Owner:LINEAR SRL
SYSTEM FOR COMBINATION OF LOUDNESS MEASUREMENTS IN A SINGLE PLAYBACK MODE
Owner:DOLBY INTERNATIONAL AB +1
Limiter system and method for avoiding clipping distortion or increasing maximum sound level in active speakers
To provide a limiter system and method for avoiding clipping distortion or increasing the maximum sound level of an active speaker.SOLUTION: The present disclosure provides a limiter system. The limiter system includes a first equalization filter, a low-pass filter, a first limiter, a high-pass filter, a second limiter, and a mixer. The limiter system provided by the present disclosure further includes a second equalization filter. An audio signal from a signal source first passes through the first equalization filter, the equalized signal is divided into two signals first, one signal is processed by the low-pass filter and the first limiter, the other signal is processed by the high-pass filter and the second limiter, and then the two processed signals enter the mixer to be mixed and outputted. The mixed output signal is subjected to second equalization filtering by the second equalization filter to avoid clipping distortion or to obtain a higher maximum sound level.SELECTED DRAWING: Figure 1
Owner:HARMAN INT IND INC
System and method for dynamic range control of an audio signal
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.
Owner:DOLBY LABORATORIES LICENSING CORP +1
RF power amplifier, chip and communication terminal
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.
Owner:VANCHIP TIANJIN TECH
DYNAMIC RANGE ADJUSTMENT OF SPATIAL AUDIO OBJECTS
Owner:DOLBY LABORATORIES LICENSING CORP
Cochlear implants and corresponding cochlear implant systems
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.
Owner:COCHLEAR LIMITED
Arrangement for aligning an upstream path
Owner:TELESTE NETWORKS OY
Systems and methods for adjusting clarity of an audio output
A method for adjusting the clarity of an audio output in a changing environment, including: receiving a content signal; applying a customized gain to the content signal; and outputting the content signal with the customized gain to at least one speaker for transduction to an acoustic signal, wherein the customized gain is applied on a per frequency bin basis such that frequencies of a lesser magnitude are enhanced with respect to frequencies of a greater magnitude and an intelligibility of the acoustic signal is set approximately at a desired level, wherein the customized gain is determined according to at least one of a gain applied to the content signal, a bandwidth of the content signal, and a content type encoded by the content signal.
Owner:BOSE CORP
Audio Adjustment Method, Device, Equipment and Computer Readable Storage Medium
Owner:SHENZHEN TCL NEW-TECH CO LTD
Volume dependent audio compensation
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.
Owner:AUDIODO AB (PUBL)
Audio processing device and method
PendingUS20250287151A1Signal processingHeadphones for stereophonic communicationTesting MethodsAudio frequency
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.
Owner:SHENZHEN SHOKZ CO LTD
A method and apparatus for improving the sound quality of a speaker
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.
Owner:HUAWEI TECH CO LTD
Speaker distortion correction device and speaker unit
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
Owner:ALPS ALPINE CO LTD
System and method for dynamic range control of an audio signal
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.
Owner:DOLBY LABORATORIES LICENSING CORP +1