15 results about "Dynamic range compression" patented technology

The invention provides a compression method and device for soft information in a signal demodulation process and electronic equipment. The method comprises the following steps: acquiring linear soft information output by demapping in a signal demodulation process, wherein the linear soft information has a preset dynamic range; the linear soft information is subjected to dynamic range compression processing through a nonlinear quantization algorithm, nonlinear soft information is obtained, and the number of quantization levels allocated to low-amplitude soft information by the nonlinear quantization algorithm is larger than the number of quantization levels allocated to high-amplitude soft information by the nonlinear quantization algorithm; and outputting the non-linear soft information to a decoder, so that the decoder restores an original hard bit stream transmitted by a signal transmitting end based on a confidence coefficient characteristic in the non-linear soft information, and the bit width of the non-linear soft information is adaptive to the bit width constraint of the decoder on the input information. According to the invention, information loss in a signal demodulation process can be reduced.

The invention discloses a dynamic range compression method and device for magnetic flux leakage signal visualization, and the method comprises the steps: obtaining an original magnetic flux leakage signal data matrix obtained through part of collection data, and obtaining a gradient signal matrix after preprocessing and gradient enhancement; obtaining an initial value of a controllable percentile parameter p; calculating a dynamic threshold group based on the current parameter p and the statistical distribution of the gradient signal matrix; carrying out nonlinear compression transformation on the gradient signal matrix by adopting an S-shaped nonlinear compression function and taking a dynamic threshold group as a scale factor to obtain a normalized feature matrix; visual mapping is carried out in combination with the dynamic threshold group, and a pseudo-color image is output; and when the preset requirement is not met, adjusting the parameter p, processing again until the preset requirement is met, and applying the parameter p to all the collected data to obtain a final pseudo-color image. According to the method, the problem of visual range imbalance caused by the great dynamic range of the magnetic flux leakage signal is solved, the parameter adjustment robustness and efficiency can be improved, and slight defect detail reservation and serious defect signal tolerance are both considered.

A system (400) for providing an audio profile for streaming video, includes an audio profile subsystem (433) that receives audio mix data and synchronizes audio data in an audio profile. A streaming service video streaming server (420) receives video data, uncompressed audio data synchronized with audio tracks, and the audio profile. An audio profile server (430) is configured to receive the audio profile and dynamic range compression descriptors to produce audio profile metadata. An audio reproduction system (410) receives audio profile metadata and listening environment parameters to adjust audio parameters based on the audio rendering system characteristics and a listening environment of the audio rendering system. The audio parameters include one or more of equalization, a filter, a rendering delay, and volume.

Various implementations include hearing assist devices and systems for processing audio signals. In particular implementations, a process includes receiving an input signal via a microphone; performing a sound pressure level (SPL) shift that decreases a gain of the input signal to generate a gain reduced audio signal; amplifying the gain reduced audio signal using dynamic range compression to generate an amplified audio signal; generating a noise reduced amplified signal using active noise reduction that simultaneously processes the input signal; and outputting the noise reduced amplified signal to an electrodynamic transducer.

The application discloses an audio signal processing method and system, a sound box and a storage medium, and relates to the technical field of the sound box. The method comprises the following steps: acquiring a transient audio signal; performing delay processing on the transient audio signal to obtain a delayed audio signal; when the transient value of the delayed audio signal exceeds a preset threshold, performing automatic gain processing on the delayed audio signal to obtain a gain audio signal; performing dynamic range compression on the gain audio signal to obtain a first steady-state signal; and playing the first steady-state signal through the sound box. The application solves the problem that the sound box in the prior art is distorted at a large volume, achieves the purpose of playing the signal after processing the received signal of the sound box, and avoids signal distortion.

Dynamic range control (DRC) cannot be simply applied to higher-order ambisonics (HOA) based signals. [Solution] A method for performing DRC on an HOA signal includes converting the HOA signal to the spatial domain, analyzing the converted HOA signal, and obtaining a gain factor usable for dynamic compression from the results of the analysis. The gain factor can be transmitted together with the HOA signal. When DRC is applied, the HOA signal is converted to the spatial domain, the gain factor is extracted, and in the spatial domain, it is multiplied with the converted HOA signal to obtain a gain-compensated converted HOA signal. The gain-compensated converted HOA signal is converted back to the original HOA domain to obtain a gain-compensated HOA signal.

An equine audio system for modulating sound input to a horse comprises anatomically-shaped earbuds with speakers configured to fit securely in the horse's ears. A digital audio processor receives an input audio signal, applies signal processing to generate an output audio signal optimized for an equine auditory system, and transmits the output audio signal to the speakers. The signal processing includes noise cancellation, frequency equalization to emphasize frequencies within the equine hearing range, and dynamic range compression. The earbuds attenuate ambient noise and may include acoustically optimized sound tubes. The system may further include a wireless receiver, a remote control device, an accelerometer for motion-based audio adjustment, and sudden loud noise detection and gradual attenuation capabilities.

The invention relates to the technical field of electro-acoustic conversion and signal processing, and discloses a bone conduction signal adaptive enhancement system of an acoustic electro-mechanical sensor, which comprises a signal acquisition unit for synchronously acquiring voltage and current signals of a transducer driving loop; the impedance analysis unit is used for removing a static impedance component based on an equivalent circuit model and extracting a motional impedance component representing mechanical load characteristics; the self-adaptive equalization unit is used for determining damping characteristics according to the motional impedance component and adjusting the gain of the filter; and the displacement protection unit is used for estimating the displacement of the vibrating diaphragm of the transducer in real time according to the motional impedance component and preferentially executing dynamic range compression when the displacement is close to a physical limit. The problem of mechanical overload edge beating caused by gain compensation when bone conduction equipment is in a loose coupling low-damping state is solved, and the mechanical safety of a system is ensured while the sound quality is improved.

To provide an audio compression method and device for suppressing distortion of an audio signal during reproduction and improving reproduction quality. [Solution] The audio compression method includes the steps of obtaining a first audio signal to be processed (an original audio signal to be reproduced), filtering the first audio signal with a frequency spectrum compensation filter based on a predetermined target frequency spectrum related to the distortion band of the audio reproduction device to obtain a second audio signal, processing the second audio signal with a predetermined dynamic range compressor (DRC) to obtain a compression coefficient, and outputting a compressed signal based on the compression coefficient.

According to the display equipment and the sound effect setting method provided by some embodiments of the invention, the method can respond to the selection instruction of the installation mode, and the target mode indicated by the selection instruction is analyzed. Wherein the selection instruction is generated based on the startup navigation interface or the menu interface. And loading a sound effect file corresponding to the target mode, setting sound parameters of the audio processor according to the sound effect file, and reading equalizer parameters corresponding to the target mode. Wherein the sound effect file comprises sound effect parameters. And writing the equalizer parameter into the power amplifier, so that the audio output device outputs sound corresponding to the audio signal based on the target mode. Wherein the power amplifier is provided with a corresponding dynamic range compression parameter. According to the method, based on the installation mode of the display equipment, different modes are switched to set the sound effect, sound loss of the display equipment caused by different installation modes can be reduced, and then the sound quality effect of sound output by the display equipment is improved.

The application relates to the technical field of electroacoustic processing, and discloses a microphone anti-howling system. The system continuously collects an audio input stream containing time-domain waveform data and background acoustic parameters through an audio capture module; a dynamic range compression module processes the audio stream to generate a standardized audio frame sequence; a feature extraction module extracts sound pressure space-time variation features including a sound pressure gradient field, a sound energy density distribution and a phase coherence atlas from the sequence; a howling identification network in a howling identification module outputs a howling occurrence probability and hotspot area coordinates based on the features; a probability correction module performs real-time correction on the probability according to environmental acoustic parameters; and an intervention scheme construction module generates a targeted acoustic intervention scheme according to the corrected probability and the hotspot area coordinates. The application realizes accurate identification and effective suppression of the howling phenomenon, and significantly improves the stability and sound quality performance of an audio system.

This invention relates to the field of audio control technology, and more particularly to an adaptive audio control method and system for in-vehicle scenarios. The method includes the following steps: acquiring in-vehicle audio signal streams based on an in-vehicle microphone array, performing dynamic range compression and frequency response balance optimization to construct an in-vehicle reference audio stream; analyzing the acoustic characteristics at multiple locations based on the in-vehicle reference audio stream, and performing acoustic transmission path distribution evolution to construct a three-dimensional sound field model of the vehicle cabin; extracting multi-dimensional driving behavior data, performing dynamic driving preference modeling, and constructing a driving behavior profile; calculating audio sensitivity based on the driving behavior profile, and dynamically adaptively adjusting it based on the three-dimensional sound field model of the vehicle cabin to construct dynamic audio adjustment parameters; applying safety optimization constraints and adaptive sound quality gain to the dynamic audio adjustment parameters, and outputting the final in-vehicle audio. This invention adjusts audio parameters in real time based on the vehicle scenario, improving driving safety and auditory experience.

The invention provides a hearing aid and a 4C audio processing system and method thereof. The system comprises a radio module, a Fourier transform module, a memory pool, a feedback elimination module, a noise reduction module, a dynamic range compression module, a steady-state coefficient estimator, an inverse Fourier transform module and an output module. The microphone module receives sound information, the Fourier transform module calculates time sequence characteristic information of the sound information, the memory pool stores the time sequence characteristic information, and the steady-state coefficient estimator reads the time sequence characteristic information of the memory pool and calculates a steady-state coefficient; the feedback elimination module reads time sequence characteristic information and a steady-state coefficient to calculate a feedback output signal, the noise reduction module calculates a noise reduction signal, and the dynamic range compression module outputs a processing signal according to the noise reduction signal; and the inverse Fourier transform module inversely transforms the processing signal to obtain an output signal, and the output module outputs a final output signal according to the output signal. The calculation efficiency is improved, the modules are coordinated, and the output sound is more balanced.

The present application is applicable to the technical field of audio processing. Provided are an audio compression method and apparatus. The method comprises: acquiring a first audio signal to be processed; on the basis of a preset target spectrum, using a spectrum compensation filter to perform filtering processing on the first audio signal, so as to obtain a second audio signal, wherein the target spectrum is related to a distortion frequency band of an audio playing device; processing the second audio signal by means of a preset dynamic range compressor (DRC), so as to obtain a compression coefficient; and on the basis of the compression coefficient, outputting a compressed signal. The method can expand the compression range without significantly reducing the gain of high-volume frequency bands, thereby reducing the occurrences of distortion during audio signal playing, and improving playing quality.

To solve such a problem that dynamic range control (DRC) cannot be applied simply to a signal of high order ambisonics (HOA) based signal.SOLUTION: A method for executing DRC of a HOA signal includes to convert the HOA signal into space region, to analyze the converted HOA signal, and to obtain a gain factor usable for dynamic compression from the analysis results. The gain factor can be transmitted together with the HOA signal. When the DRC is applied, the HOA signal is converted into the space region, the gain factor is extracted, and multiplied by the converted HOA signal in the space region, thus obtaining a converted HOA signal subjected to gain compensation. The converted HOA signal subjected to gain compensation is converted into the original HOA region, thus obtaining a HOA signal subjected to gain compensation.SELECTED DRAWING: Figure 2