42 results about "Dynamic range compression" patented technology

The invention relates to the technical field of artificial intelligence and digital audio signal processing, and particularly discloses a Karaoke all-in-one machine intelligent tuning method based on AI sound effect optimization, and the method comprises the steps: obtaining an original mixed audio signal sung by a user, and carrying out the preprocessing; processing the preprocessed audio by using a multi-source audio separation model based on a deep neural network, and extracting a human voice track and accompaniment components; extracting time-frequency domain features from the human voice track, and performing quantitative evaluation on the voice tension degree of the user in combination with the trained voice state recognition model; dynamically adjusting an equalizer and a dynamic range compression parameter according to an evaluation result to realize adaptive sound effect optimization; synthesizing the optimized voice and accompaniment and outputting the synthesized voice and accompaniment to monitoring equipment; and finally, establishing a personalized user behavior feature vector by collecting user singing performance and historical tuning data, and performing online fine tuning on a system strategy based on a deep Q network reinforcement learning mechanism to continuously optimize a tuning effect.

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.

The invention relates to the field of audio signal processing, in particular to a multi-channel audio signal hybrid coding method and device and a storage medium, and the method comprises the steps: collecting digital audio signals through a plurality of audio input channels, carrying out the anti-aliasing filtering of the collected digital audio signals, so as to obtain audio signals in a 48kHz / 16bit format, and outputting the audio signals in the 48kHz / 16bit format. An audio signal in a 48kHz / 16bit format is processed through a loudness weighted average method, the audio signal after loudness weighted average is processed through a frequency domain mixing technology to obtain a mixed signal, and dynamic range compression is performed on the mixed signal based on calculation of an auditory masking curve to obtain a compressed signal. And carrying out hybrid coding on the compressed signal to obtain a compressed data stream, and packaging and outputting the compressed data stream. Through dynamic mixing and compression processing, the problem of signal distortion caused by a traditional linear superposition method is effectively solved, reasonable superposition of all paths of signals is ensured through dynamic mixing, and level overload caused by simple addition is avoided.

The invention discloses an event feature combined image defogging method, device and equipment, a medium and a product, and the method comprises the steps: inputting an obtained foggy image into a preset encoder for encoding, and extracting potential feature representation; adding noise to the potential feature representation to obtain a noise representation; acquiring event features of event streams synchronously acquired by the foggy image; fusing the event features into a preset image segmentation network, and performing diffusion reverse denoising iteration on the noise representation to obtain a denoising result; and inputting the de-noising result into a preset decoder for decoding to obtain a defogged image. According to the scheme, the limitation of dynamic range compression in the defogging process can be reduced, and the image defogging effect is improved.

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.

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.

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 invention relates to a system for maintaining reversible dynamic range control information associated with a parametric audio encoder. On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and a multi-channel coding parameter (alpha) from the bitstream, where 1 < = m < n. A pre-processing dynamic range control parameter, i.e., a pre-processing DRC parameter (DRC2), that quantizes the encoder-side dynamic range limitation of the core signal is also derived from the bitstream. The n-channel audio signal is obtained by parameter synthesis according to the multi-channel encoding parameters while eliminating any encoder-side dynamic range limitations based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes the use of compensated post-processing DRC parameters that quantify possible decoder-side dynamic range compression. Preferably, encoder-side range limit cancellation and range compression are performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC preprocessor.

An audio processing unit, comprising a buffer memory (201) configured to store at least one frame of an encoded audio bitstream which includes audio data and a metadata container having a header and one or more metadata payloads after the header, a parser (205) coupled to the buffer memory, and an audio decoder (202) coupled to the parser. The one or more metadata payloads include dynamic range compression (DRC) metadata including DRC control values indicative of a compression profile assumed by an encoder to generate said DRC control values. The audio decoder is configured to perform dynamic range compression on decoded audio data generated by decoding said at least some of the audio data, using the DRC metadata.

A hearing aid includes neural network circuitry configured to implement a neural network trained to separate a speech subsignal and a noise subsignal from an input audio signal, and digital processing circuitry. The digital processing circuitry includes a speech wide dynamic range compression (WDRC) pipeline and a noise WDRC pipeline. The speech WDRC pipeline is configured to apply a set of speech fitting curves to the speech subsignal based at least in part on the level of the speech subsignal. The noise WDRC pipeline is configured to apply a set of noise fitting curves to the noise subsignal based at least in part on the level of the noise subsignal. The set of speech fitting curves is different from the set of noise fitting curves.

The invention relates to an audio signal processing method and device and a storage medium. The audio signal processing method comprises the following steps: converting an audio signal to be processed into a frequency domain to obtain a frequency domain audio signal; performing frequency band division processing on the frequency domain audio signal to obtain frequency domain audio signals of different frequency bands; performing dynamic range compression (DRC) processing on the frequency domain audio signal of each frequency band in the different frequency bands to obtain a DRC-processed frequency domain audio signal of each frequency band; and adding the DRC processed frequency domain audio signal of each frequency band, and performing time domain conversion to obtain a DRC processed time domain audio signal of the audio signal to be processed. A multi-band dynamic range compressor is realized in a frequency domain, point-by-point calculation is avoided, the calculation amount is reduced, and introduction of unnecessary frequency components is avoided.

The invention relates to a microphone array beam forming method, system and device, and the method comprises the steps: obtaining an original sound signal collected by an omnidirectional microphone, and carrying out the synchronous calibration processing of the original sound signal, and obtaining a corresponding preprocessing signal; carrying out differential processing on the pre-processed signal to obtain a plurality of heart-shaped directivity unit signals, and carrying out phase correction and gain balance optimization to obtain optimized heart-shaped unit signals; performing signal covariance matrix and guide vector calculation on the optimized heart-shaped unit signal according to a preset MVDR algorithm to obtain a corresponding optimal weight; performing beam forming processing on the optimal weight and the optimized heart-shaped unit signal to obtain an initial beam signal; performing frequency response equalization and dynamic range compression on the initial beam signal to obtain an enhanced output signal; and performing performance monitoring and parameter adjustment based on the enhanced output signal to obtain a final real-time beam signal. According to the invention, stable operation of the system in different acoustic environments can be ensured.

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 provides a signal processing method of a single-channel hearing aid and the single-channel hearing aid, and the method comprises the steps: obtaining a time-domain signal frame of a single-channel audio signal, and carrying out the time-frequency transformation of the time-domain signal frame, and obtaining the spectrum representation of the time-domain signal frame in a frequency domain; dividing the frequency spectrum into a plurality of virtual frequency sub-bands, and calculating a corresponding dynamic range compression gain for each virtual frequency sub-band; solving the background noise of the frequency spectrum by using a mixed noise power spectral density algorithm to analyze an adaptive noise reduction gain for suppressing the background noise; determining a synthesis mode of a dynamic range compression gain and an adaptive noise reduction gain according to the environmental parameters, and generating a comprehensive gain; and applying the comprehensive gain to the spectrum representation, performing frequency-time inverse transformation on the updated spectrum representation, and reconstructing a target audio signal for output. According to the invention, a multi-channel processing effect can be realized under a single-channel architecture, so that the hearing aid can still provide clear speech signals and comfortable hearing feeling in a complex listening environment.

Aptamer detection techniques with dynamic range compression are described that permit removal of a portion of more abundant aptamers (14) in an aptamer-based assay. In accordance with certain aspects of the present techniques processor-implemented algorithms are employed to facilitate identifying groupings of analytes to be assayed, wherein each grouping will have a different associated dilution factor. Such algorithms may additionally help determine the dilution factors employed for each group. In one aspect a processor-based algorithm may be employed for minimizing or otherwise optimizing total count as part of determining demarcations points (i.e., slice points) between groups. In a further aspect a processor-based algorithm may be employed for minimizing range-error for a set of slice points and dilution factors.

A method performed by an audio source device that is communicatively coupled with an audio output device, the method includes obtaining an audio signal; receiving a volume adjustment; applying the volume adjustment to the audio signal; applying dynamic range compression on the volume adjusted audio signal; and applying an inverse volume adjustment to the compressed audio signal, where the inverse volume adjusted audio signal is to be transmitted to the audio output device.

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.

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

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.

The application discloses a dynamic range compression method for improving image flicker, and proposes a double-threshold judgment method. Compared with a conventional linear method, the double-threshold judgment method increases a flicker threshold, and the continuity between frames can be utilized to make the image mapping curve in the linear mapping process not abrupt. By utilizing the double-threshold judgment method, the application can reduce the sharp change of frame brightness when the scene brightness is abruptly changed, and improve the anti-flicker capability of the picture when the environment is abruptly changed.

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

The invention relates to the technical field of audio control, in particular to a self-adaptive audio control method and system for a vehicle-mounted scene. The method comprises the following steps: acquiring an in-vehicle audio signal stream based on a vehicle-mounted microphone array, performing dynamic range compression and frequency response balance optimization, and constructing an in-vehicle reference audio stream; carrying out multi-position acoustic characteristic analysis according to the in-vehicle reference audio stream, carrying out acoustic transmission path distribution evolution, and constructing a vehicle cabin three-dimensional sound field model; extracting multi-dimensional driving behavior data, performing dynamic driving preference modeling, and constructing a driving behavior portrait; carrying out audio sensitivity calculation according to the driving behavior portrait, carrying out dynamic adaptive adjustment based on a vehicle cabin three-dimensional sound field model, and constructing a dynamic audio adjustment parameter; and performing safety optimization constraint and adaptive tone quality gain on the dynamic audio adjustment parameters, and outputting a final vehicle-mounted audio. The audio parameters are adjusted in real time based on the vehicle scene, and the driving safety and auditory experience are improved.

The present disclosure relates to the field of audio data processing, and proposes an audio signal processing method and device, electronic equipment and storage medium. The method comprises: showing a sound effect mode supported by a target multimedia resource to a user; in response to the user selecting a senior exclusive sound effect, causing the audio of the target multimedia resource to present a special effect corresponding to the senior exclusive sound effect when playing the target multimedia resource; the special effect corresponding to the senior exclusive sound effect comprises at least one of a high-frequency sound enhancement special effect, a human voice enhancement special effect and an audio dynamic range compression special effect, the high-frequency sound enhancement special effect is used to enhance the amplitude of signals belonging to a high-frequency band, the human voice enhancement special effect is used to enhance the amplitude of signals belonging to a human voice band, and the audio dynamic range compression special effect is used to compress the dynamic range of signals. The method supports the senior exclusive sound effect customized for the elderly users, can improve the viewing experience of the elderly users on the multimedia resources, and reduces the adverse effects of the elderly users watching the multimedia resources on the physical health.

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.