32 results about "Loudness compensation" patented technology

The invention relates to the measurement and control of the perceived sound loudness and / or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Audio signal processing relating to the measurement and control of the perceived sound loudness and / or the perceived spectral balance of an audio signal is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. In various embodiments, modification parameters are derived for modifying the audio signal in order to reduce the difference between its specific loudness and a target specific loudness.

An audio loudness compensation system includes a level sensor receiving an audio input signal and operable to estimate a level of the audio input signal over a first predetermined time period, and a level mapper receiving the estimated level and operable to map the estimated level to a raw audio gain in response to a slope setting and an offset setting. The system further includes an attack and decay filter receiving the raw audio gain and operable to smooth out increasing and decreasing changes in the raw audio gain in response to a second and, possibly a third predetermined time period, and a compensation filter receiving the smoothed raw audio gain and operable to modify the audio input signal in response to the smoothed raw audio gain, a center frequency setting and a bandwidth setting, and generate a loudness compensated audio output signal.

An audio correcting apparatus includes a speaker provided on a television apparatus, a microphone provided on a remote controller, an identifying unit which identifies an acoustic characteristic from the speaker to the microphone, and an acoustic characteristic setting unit having the acoustic characteristic. A signal obtained by allowing an audio signal input to the speaker to pass through the acoustic characteristic setting unit, and a signal representing ambient noise are input to an audio-correcting filter and a loudness-compensation-gain calculating unit. Based on both signals, the sound pressure level of sound output from the speaker is corrected so that the sound output from the speaker is clearly heard when reaching the user without being affected by the ambient noise.

Audio signal processing relating to the measurement and control of the perceived sound loudness and / or the perceived spectral balance of an audio signal is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. In various embodiments, modification parameters are derived for modifying the audio signal in order to reduce the difference between its specific loudness and a target specific loudness.

A speech correction apparatus includes a speaker for generating guidance speech; a microphone set at a hearing position; an acoustic-characteristic setting unit for separating ambient noise from the guidance speech at the hearing position; an operating unit; a speech correcting filter for correcting the sound pressure level of the guidance speech generated by the speaker based on the average power of the guidance speech and the average power of the ambient noise which are separated; a loudness-compensating-gain calculating unit; and a speech-head correcting unit for correcting the average power of the guidance speech corresponding to the speech head at the border between a silent state and a speech state of the guidance speech.

An audio compensation system (20) for producing a sound compensated output signal (So). The system comprises a volume control circuit (24) for producing a volume-adjusted signal (Sic) by applying a volume adjustment to an audio signal in response to a volume setting, wherein the sound compensated output signal is responsive to the volume-adjusted signal. The system also comprises circuitry (26, 30, 32) for producing an amplified signal (Sifa) by amplifying a selected bandwith of signals in response to the volume setting. The sound compensated output signal is also responsive to the amplified signal.

Audio signal processing relating to the measurement and control of the perceived sound loudness and / or the perceived spectral balance of an audio signal is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. In various embodiments, modification parameters are derived for modifying the audio signal in order to reduce the difference between its specific loudness and a target specific loudness.

The present invention relates to an auto loudness circuit for performing loudness compensation automatically depending on the signal level. When the signal level decreases, loudness compensation is slowly introduced and as the signal level increases, loudness compensation is quickly removed. To do so, the auto loudness circuit utilizes a filter circuit with the characteristic of a first order bass boost. The filter circuit maintains a corner frequency which is proportional to the inverse of audio level in order to mimic the Fletcher-Munson curves. Because the circuit employs a capacitance-multiplier with a first order resistance capacitance filter, the bass boost is inversely proportional to the signal level. Thus, bass boost is achieved automatically as the program content changes so that the listener is unaware of significant changes in program material as signal levels change either through increase or decrease in volume, crescendo or new material.

The invention provides an equal-loudness contour-based loudness compensation method, which comprises the following steps of: regulating a volume of an input signal, measuring sound pressure levels corresponding to all available volume values, and in each sound pressure level, filtering an input signal by using a preset frequency filter, and multiplying by a weighting coefficient of the preset frequency filter to obtain a signal; and superposing the input signal in each sound pressure level in a plurality of sound pressure levels corresponding to all the available volume values and the signal obtained through filtering and weighting. The method overcomes the defect that when the conventional audio signal is small in volume, the loudness of partial frequency band is too low, and ears of a human body cannot feel the loudness, so that the heard sound is dry and tedious; by the method, the loudness of input signals of partial frequency band is compensated, the sound is loud and full and meets the requirement of people on acoustic comfort; moreover, a sound level meter and the filter and simple computation are only required to realize the loudness compensation of the audio signal, so that the structure is simple and the cost is low.

A decoder for generating an audio output signal having one or more audio output channels is provided, having a receiving interface for receiving an audio input signal having a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated, further having a signal processor for generating the one or more audio output channels of the audio output signal, configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information, and configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. One or more by-pass audio object signals are employed for generating the audio output signal. Moreover, an encoder is provided.

The invention discloses a digital hearing aid loudness compensation method based on frequency compression and movement, which belongs to the technical field of voice signal processing. The digital hearing aid loudness compensation method comprises the following steps of: filtering, and extracting the low-frequency part and the high-frequency part of a signal; compressing and moving the voice high-frequency part according to proportion; simulating a voice signal heard by a patient; designing a four-channel FIR-QMFB (Quadrature Mirror Filter Banks); dividing a frequency range; and carrying out loudness compensation and gain control; and combining voice. The digital hearing aid loudness compensation method is characterized in that the original voice signal is divided into low frequency and high frequency, then, the high-frequency part is compressed and moved to a medium and low frequency range according to proportion, and the overlaid voice signal is subjected to frequency band division and loudness compensation according to a hearing test curve of a hearing impairment patient. Therefore, negative effects brought by carrying out high-gain loudness compensation on the high frequency range in the prior art can be effectively avoided, and the hearing level and the language recognition rate of the patient can be effectively improved.

The invention provides a volume-based personalized loudness compensation method and an earphone equalization system. The invention provides a method including: by playing a burst sound reference signal and a burst sound test signal at each selected volume, adjusting the gain of each burst sound test signal to make its loudness equal to that of the burst sound reference signal, obtaining a loudnessgain curve at each volume, and correcting and interpolating the loudness gain curve by using a universal iso-loudness curve to form a gain continuous amplitude response curve; and generating the earphone equalization curve by amplitude response curve to compensate the loudness of the audio signal. By using the method and the earphone equalization system provided by the invention, the earphone loudness equalization curve under different volume can be obtained, and the equalization curve is obtained by adjusting the loudness gain of the signal by the user, so as to compensate for the problem ofinsensitive frequency band hearing feeling generated by playback audio under different sound pressure levels, and provide better earphone auditory experience for the user.

A loudness-compensating volume control method imposes a desired loudness scaling on an audio signal by processing the audio signal in both the digital and analog domains by receiving a desired loudness scaling, deriving a wideband gain component and one or more other gain components from the desired loudness scaling, applying in the digital domain modifications to the audio signal based on the one or more other gain components to produce a partly-modified audio signal, and applying in the analog domain modifications to the partly-modified audio signal based on the wideband gain component. Additional loudness modifications other than volume control loudness modifications on the audio signal may also be imposed.

A loudness-compensating volume control method imposes a desired loudness scaling on an audio signal by processing the audio signal in both the digital and analog domains by receiving a desired loudness scaling, deriving a wideband gain component and one or more other gain components from the desired loudness scaling, applying in the digital domain modifications to the audio signal based on the one or more other gain components to produce a partly-modified audio signal, and applying in the analog domain modifications to the partly-modified audio signal based on the wideband gain component. Additional loudness modifications other than volume control loudness modifications on the audio signal may also be imposed.

An audio correcting apparatus includes a speaker provided on a television apparatus, a microphone provided on a remote controller, an identifying unit which identifies an acoustic characteristic from the speaker to the microphone, and an acoustic characteristic setting unit having the acoustic characteristic. A signal obtained by allowing an audio signal input to the speaker to pass through the acoustic characteristic setting unit, and a signal representing ambient noise are input to an audio-correcting filter and a loudness-compensation-gain calculating unit. Based on both signals, the sound pressure level of sound output from the speaker is corrected so that the sound output from the speaker is clearly heard when reaching the user without being affected by the ambient noise.

An audio processor for processing an audio signal includes: an audio signal modifier for modifying the audio signal in response to a user input; a loudness controller for determining a loudness compensation gain based on a reference loudness or a reference gain and a modified loudness or a modified gain, where the modified loudness or the modified gain depends on the user input; and a loudness manipulator for manipulating a loudness of a signal using the loudness compensation gain.

The invention discloses a parameter self-adjusting method for a fitting-free hearing aid. The parameter self-adjusting method comprises the following steps: 1, determining 10 groups of parameters [a1, b1, a2, b2, r, s, t] from 1 to 10; 2, according to the hearing loss of the patient at the center frequency CF, respectively calculating the hearing loss of internal / external hair cells corresponding to the 10 groups of parameters, and the maximum gain and compensation coefficient of the normal ear and the affected ear; 3, framing and windowing the input voice xin, and calculating an energy spectrum E(k) of each frame of signal; 4, calculating compensation gains corresponding to the 10 groups of parameters; 5, enabling the compensation gain calculated in the step 4 to act on the signal spectrum, and obtaining 10 groups of compensated voice signals; and 6, calculating the fitness of the 10 groups of voice signals, and regenerating 10 groups of parameters. According to the parameter self-adjustment method for the fitting-free hearing aid, the loudness compensation model is established according to the cochlea hearing loss model, the parameters are adjusted adaptively based on the intelligent algorithm, and the method has the advantages of being good in loudness compensation effect and convenient to deploy.

A decoder for generating an audio output signal having one or more audio output channels is provided. The decoder includes a receiving interface for receiving an audio input signal including a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated. Moreover, the decoder includes a signal processor for generating the one or more audio output channels of the audio output signal. The signal processor is configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information. Furthermore, the signal processor is configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. Moreover, an encoder is provided.

The invention discloses a channel-adaptive digital hearing aid wide dynamic range compression method, which comprises the following steps of: firstly, selecting an asymmetric filter bank decomposition and synthesis algorithm by simulating human auditory characteristics; then, designing the channel number and parameters of a filter bank according to an audiogram of a patient and a psychological acoustic model to obtain a personalized filter bank conforming to the hearing loss of the patient; and finally, performing wide dynamic range compression on input sound signals in channels by using the digital hearing aid of the method. The compression comprises the following specific steps of: carrying out adaptive channel filter bank decomposition on input signals to obtain adaptive channel signals; carrying out loudness compensation on each decomposed channel signal; performing filter bank synthesis on the compensated channel signals to obtain full-band signals; and converting the synthesized full-band signals into sound signals and outputting the sound signals. The calculation complexity of the system is reduced while the performance requirement is met, and the speech intelligibility of the patient is improved.

The invention relates to a sound field expansion method, computer equipment and a computer readable storage medium, and the method comprises the following steps: obtaining at least two sound channel signals, wherein at least two sound channel signals comprise a first sound channel signal and a second sound channel signal; obtaining a first gain signal and a third gain signal corresponding to the first sound channel signal, and a second gain signal corresponding to the second sound channel signal; obtaining a first loudness compensation signal according to the first gain signal and the second gain signal; and obtaining an output signal of the first sound channel signal according to the third gain signal and the first loudness compensation signal. Through loudness compensation, strong soundis removed, weak sound which cannot be perceived by human ears due to the masking effect is amplified and enhanced and is perceived by the human ears, the three-dimensional sound effect is improved, and a real sound field is restored and obtained.

An audio processor for processing an audio signal includes: an audio signal modifier for modifying the audio signal in response to a user input; a loudness controller for determining a loudness compensation gain based on a reference loudness or a reference gain and a modified loudness or a modified gain, where the modified loudness or the modified gain depends on the user input; and a loudness manipulator for manipulating a loudness of a signal using the loudness compensation gain.

The invention relates to artificial intelligence, and provides an identity verification method and device, equipment and a storage medium. The method can obtain to-be-verified audio and environment audio, perform loudness compensation processing on the to-be-verified audio and the environment audio to obtain first verified audio and noise audio, remove the noise audio from the first verified audio to obtain second verified audio, and perform compression processing on the second verified audio to obtain a first audio feature. The similarity between the first audio feature and all preset features are calculated to obtain a first confidence value, if the first confidence value is greater than a first preset threshold and the first confidence value is less than a second preset threshold, the similarity between the second audio feature and all preset features is extracted and calculated to obtain a second confidence value, and if the second confidence value is greater than the second preset threshold, it is determined that the identity verification request passes verification. According to the invention, the accuracy of identity verification can be improved. In addition, the invention also relates to a blockchain technology, and all the preset features can be stored in the blockchain.

The present application relates to a sound field expansion method, computer equipment, and a computer-readable storage medium. The method includes the following steps: acquiring at least two channel signals, wherein the at least two channel signals include a first channel signal and a first channel signal. second channel signal; acquiring a first gain signal and a third gain signal corresponding to the first channel signal, and a second gain signal corresponding to the second channel signal; according to the first gain signal and the obtained obtaining a first loudness compensation signal through the second gain signal; obtaining an output signal of the first channel signal according to the third gain signal and the first loudness compensation signal. Loudness compensation is used to remove strong sounds, and to amplify and enhance weaker sounds that cannot be detected by the human ear due to the masking effect, so that they can be perceived by the human ear, improving the three-dimensional sound effect and restoring the real sound field.

The invention discloses an adaptive multi-channel loudness compensation method for hearing aids and a hearing aid chip, comprising: acquiring an input signal and performing AD conversion to obtain a digital signal, performing frame division to obtain a voice signal; performing filter transformation on all voice signals to obtain multiple Channel signals, and all channel signals are divided into the first channel combination and the second channel combination according to the audible range; each channel signal in the first channel combination is segmented linear gain; each channel in the second channel combination The signals are all subjected to non-linear gain; each channel signal in the first channel combination after gain compensation is synthesized with each channel signal in the second channel combination after gain compensation, and a voice signal after gain compensation is obtained and output. First, multi-channel decomposition is performed on the signal, and then the channels are divided into two groups. The channels with a smaller audible range are combined for piecewise linear gain, while the other channel is combined for non-linear gain, which effectively improves the voice quality after gain compensation. .

The invention relates to the technical field of hearing aids. The invention particularly relates to a frequency shift real-time loudness compensation method based on an equal loudness curve. The method comprises the following steps: outputting N frequency bands of an input signal through a WOLA analysis window, calculating frequency shift parameter inflection point frequency and cut-off frequency,acquiring frequency shift step length according to the selected inflection point frequency and cut-off frequency, acquiring a non-frequency shift signal and a frequency shift signal according to theinflection point frequency, namely the compression ratio, calculating the sound pressure level of each frequency band of the frequency shift signal and the channel index of the original signal frequency in real time through a sound pressure level detection device, making the frequency shift signal pass through a frequency shift device to obtain a frequency-shifted signal and a target channel indexthereof, searching an equal loudness curve array to obtain a signal sound pressure level index so that a compensated frequency shift signal is formed, and synthesizing the compensated frequency shiftsignal and the non-frequency shift signal to form a hearing aid output signal. According to the invention, the frequency shift hearing aid can simply and effectively ensure that a wearer has the perception ability before and after signal frequency shift in any environment.

The invention discloses a hearing aid-oriented adaptive multi-channel loudness compensation method and a hearing aid chip, and the method comprises the steps: obtaining an input signal, carrying out the AD conversion of the input signal, obtaining a digital signal, and carrying out the framing of the digital signal, and obtaining a voice signal; performing filtering transformation on the voice signals to obtain a plurality of channel signals, and dividing all the channel signals into a first channel combination and a second channel combination according to an audible range; performing piecewise linear gain on each channel signal in the first channel combination; performing nonlinear gain on each channel signal in the second channel combination; and synthesizing each channel signal in the first channel combination after gain compensation and each channel signal in the second channel combination after gain compensation to obtain a voice signal after gain compensation, and outputting the voice signal. Firstly, multi-channel decomposition is performed on a signal, then channels are divided into two groups, a channel combination with a smaller audible range is used for piecewise linear gain, and the other channel combination is used for nonlinear gain, so that the voice quality after gain compensation is effectively improved.

The invention discloses a channel-adaptive digital hearing aid wide dynamic range compression method. First, an asymmetric filter bank decomposition and synthesis algorithm is selected by simulating the hearing characteristics of the human ear, and then the filter is designed according to the patient's audiogram and a psychoacoustic model. The number of channels and parameters of the filter group can be used to obtain a personalized filter bank that meets the hearing loss of the patient; finally, the digital hearing aid using the above method performs wide dynamic range compression on the input sound signal sub-channels; the specific steps of compression include: automatically compressing the input signal The adaptive channel filter bank is decomposed to obtain the signal of the adaptive channel; the loudness compensation is performed on each channel signal after decomposition; the filter bank synthesis is performed on the compensated channel signal to obtain a full-band signal; the integrated full-band signal is The signal is converted to an audio signal output. While meeting the performance requirements, the computational complexity of the system is reduced, and the patient's speech intelligibility is improved.