CN113132882A - Multi-dynamic-range companding method and system - Google Patents
Multi-dynamic-range companding method and system Download PDFInfo
- Publication number
- CN113132882A CN113132882A CN202110414005.9A CN202110414005A CN113132882A CN 113132882 A CN113132882 A CN 113132882A CN 202110414005 A CN202110414005 A CN 202110414005A CN 113132882 A CN113132882 A CN 113132882A
- Authority
- CN
- China
- Prior art keywords
- signal
- dynamic range
- channel
- gain function
- ith
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
Images
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
- H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
- H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
- H04R25/30—Monitoring or testing of hearing aids, e.g. functioning, settings, battery power
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
- H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
Landscapes
- Health & Medical Sciences (AREA)
- General Health & Medical Sciences (AREA)
- Otolaryngology (AREA)
- Neurosurgery (AREA)
- Physics & Mathematics (AREA)
- Engineering & Computer Science (AREA)
- Acoustics & Sound (AREA)
- Signal Processing (AREA)
- Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
- Circuit For Audible Band Transducer (AREA)
Abstract
The invention provides a multi-dynamic range companding method for use in a hearing aid, the hearing aid comprising a feedforward microphone positioned on a side of the hearing aid remote from an ear canal, the method comprising: acquiring an input signal provided by the feedforward microphone; analyzing signal characteristics of the input signal, the signal characteristics including a signal-to-noise ratio and/or a signal-to-mixing ratio; selecting a target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio; and according to the energy of the input signal and the target gain function, carrying out dynamic range compression on the input signal to obtain a target signal. In the present invention, the noise or reverberation can be prevented from being excessively amplified by selecting a target gain function for dynamic range compression by a signal-to-noise ratio and/or a signal-to-mixing ratio.
Description
Technical Field
The present invention relates to the field of speech processing, and in particular, to a method, a system, a computer device, and a computer-readable storage medium for multi-dynamic-range companding.
Background
The long-term exposure to the environment with large sound pressure level easily causes mental stress and hearing loss of people. The frequency components of the sound heard by human ears are mainly 20hz to 20ihz, and different hearing domains exist according to different frequency points. As the age increases, the range of audible frequency components gradually narrows, and the auditory range narrows. That is, as the age increases or the hearing is impaired, the auditory field gradually rises. For this reason, hearing aids are being developed and used.
A hearing aid, which can boost a weak signal into the hearing range of a hearing-impaired person, is shown in fig. 1.
The present inventors have appreciated that the hearing aid generally amplifies signals in two ways:
(1) the signals of all frequencies are uniformly amplified by the analog element, so that the signals of a hearing-impaired person in a specific frequency band can be amplified to be within the hearing range, but some useless signals can be equally amplified.
(2) Performing multi-dynamic range companding by a digital processor: the signals in different frequency bands are processed (amplified) differently. The amplification in each channel depends on the magnitude of the energy gain of the signal in its respective frequency band. The multi-dynamic range companding described above tends to amplify unwanted signals such as background noise equally with the desired signal.
Therefore, when reverberation or noise is large, the sound signal after the dynamic range compression process may seriously affect the hearing experience.
Disclosure of Invention
It is an object of the present invention to provide a multi-dynamic range companding method, system, computer device and computer readable storage medium that solves or partially solves the above mentioned problems.
An aspect of an embodiment of the present invention provides a multi-dynamic range companding method for use in a hearing aid, the hearing aid comprising a feedforward microphone positioned on a side of the hearing aid remote from an ear canal, the method comprising:
acquiring an input signal provided by the feedforward microphone;
analyzing signal characteristics of the input signal, the signal characteristics including a signal-to-noise ratio and/or a signal-to-mixing ratio;
selecting a target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio;
and according to the energy of the input signal and the target gain function, carrying out dynamic range compression on the input signal to obtain a target signal.
Optionally, the signal characteristics of the input signal include N subband signal characteristics, where N is an integer > 1; the step of analyzing the input signal characteristic of the input signal comprises:
decomposing the input signal into N first subband signals S corresponding to N channels11、S12、…S1N(ii) a And
analyzing the N first sub-band signals respectively to obtain the N sub-band signal characteristics;
wherein each sub-band signal characteristic comprises a sub-band signal-to-noise ratio and/or a sub-band signal-to-noise ratio of the respective first sub-band signal.
Optionally, the target gain function includes N gain functions corresponding to the N channels, and the N channels are in one-to-one correspondence with the N gain functions; the step of selecting the target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio comprises:
comparing the first threshold with the ith sub-band signal-to-noise ratio corresponding to the ith channel, wherein i is more than or equal to 1 and less than or equal to N, and is an integer;
if the ith subband signal-to-noise ratio is larger than the first threshold, selecting a first gain function of a plurality of gain functions as an ith gain function corresponding to the i channels, wherein the gain functions are preset;
if the ith subband signal-to-noise ratio is not greater than the first threshold, taking a second gain function of the plurality of gain functions as the ith gain function; the slope of the first gain function is greater than the slope of the second gain function.
Optionally, the step of performing dynamic range compression on the input signal according to the energy of the input signal and the target gain function to obtain a target signal includes:
generating N target subband signals from the N first subband signals, comprising: performing dynamic range compression on an ith first sub-band signal corresponding to the ith channel according to the ith gain function and an ith sub-band energy of the ith channel to generate an ith target sub-band signal corresponding to the ith channel; and
and synthesizing the N target subband signals to obtain the target signal.
Optionally, the hearing aid further comprises a VAD element proximate to the ear canal of the wearer when the hearing aid is worn; the method further comprises the following steps:
obtaining a feedback signal provided by the VAD element;
decomposing the feedback signal into N second subband signals S corresponding to the N channels21、S22、…S2NWherein the second subband signal S2iCorresponding first subband signal S1iCorresponding to the same channel; and
judging whether each channel comprises the own sound signal of the wearer or not according to each second sub-band signal; and
adjusting the gain function of each channel according to the judgment result of each channel; each channel corresponds to a judgment result, the judgment result is a first judgment result or a second judgment result, the first judgment result is used for indicating that the corresponding channel comprises the sound signal of the wearer and indicating that the slope of the gain function of the corresponding channel is reduced, and the second judgment result is used for indicating that the corresponding channel does not comprise the sound signal of the wearer.
Optionally, the method further includes:
judging whether each channel comprises an impact signal or not;
if one of the channels includes the impact signal, the gain function of that channel is adjusted.
Optionally, the method further includes:
judging whether the input signal comprises a wind noise signal;
and if the input signal comprises the wind noise signal, adjusting the gain function of each channel lower than a preset frequency point.
An aspect of an embodiment of the present invention further provides a multi-dynamic range companding system, for use in a hearing aid, the hearing aid including a feedforward microphone positioned on a side of the hearing aid remote from an ear canal, the system comprising:
the acquisition module is used for acquiring an input signal provided by the feedforward microphone;
an analysis module for analyzing signal characteristics of the input signal, the signal characteristics including a signal-to-noise ratio and/or a signal-to-mixing ratio;
a selection module, configured to select a target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio;
and the processing module is used for compressing the dynamic range of the input signal according to the energy of the input signal and the target gain function to obtain a target signal.
An aspect of the embodiments of the present invention further provides a computer device, which includes a memory, a processor, and a computer program stored on the memory and executable on the processor, and the processor implements the steps of the multi-dynamic range companding method as described above when executing the computer program.
An aspect of the embodiments of the present invention further provides a computer-readable storage medium, which includes a memory, a processor, and a computer program stored on the memory and executable on the processor, and the processor executes the computer program to implement the steps of the multi-dynamic range companding method as described above.
The multi-dynamic range companding method, the device and the computer readable storage medium provided by the embodiment of the invention can prevent noise or reverberation from being excessively amplified by selecting the target gain function of dynamic range compression through the signal-to-noise ratio and/or the signal-to-mixing ratio.
Drawings
Fig. 1 schematically shows a auditory field range diagram;
fig. 2 schematically shows a schematic view of the construction of a hearing aid;
FIG. 3 is a flow chart of dynamic range compression based on signal-to-noise ratio;
FIG. 4 is a graph of a gain function for dynamic range compression;
FIG. 5 is a flow diagram of dynamic range compression based on signal-to-mixture ratio;
FIG. 6 is a flow chart of dynamic range compression based on signal-to-noise ratio and VAD;
FIG. 7 is a flow chart of dynamic range compression based on signal-to-noise ratio and VAD;
FIG. 8 is a flow chart that schematically illustrates a method for multi-dynamic range companding, in accordance with a first embodiment of the present invention;
FIG. 9 is a sub-flowchart of step S802 in FIG. 8;
FIG. 10 is a sub-flowchart of step S804 in FIG. 8;
FIG. 11 is a sub-flowchart of step S806 in FIG. 8;
FIG. 12 is a diagram schematically illustrating additional steps of a multi-dynamic range companding method according to a first embodiment of the present invention;
FIG. 13 is a diagram schematically illustrating another additional step of the multi-dynamic range companding method according to the first embodiment of the present invention;
FIG. 14 is a diagram schematically illustrating another additional step of the multi-dynamic range companding method according to the first embodiment of the present invention;
FIG. 15 is another flow chart diagram of a multi-dynamic range companding method;
FIG. 16 schematically illustrates a block diagram of a multi-dynamic range companding system according to a second embodiment of the present invention; and
fig. 17 is a schematic diagram illustrating a hardware architecture of a computer device suitable for implementing the multi-dynamic range companding method according to a third embodiment of the present invention.
Detailed Description
In order to make the objects, technical solutions and advantages of the present invention more apparent, the present invention is described in further detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
It should be noted that the descriptions relating to "first", "second", etc. in the embodiments of the present invention are only for descriptive purposes and are not to be construed as indicating or implying relative importance or implicitly indicating the number of technical features indicated. Thus, a feature defined as "first" or "second" may explicitly or implicitly include at least one such feature. In addition, technical solutions between various embodiments may be combined with each other, but must be realized by a person skilled in the art, and when the technical solutions are contradictory or cannot be realized, such a combination should not be considered to exist, and is not within the protection scope of the present invention.
Interpretation of terms to which the invention relates:
VAD (Voice activity detection), also called Voice endpoint detection, is used to detect whether a Voice signal exists.
The signal-to-noise ratio (SNR) is used to compare the strength of the desired signal to the strength of the background noise, which is the ratio of the signal power to the noise power, expressed in decibels (dB). A ratio of greater than 1:1 (above 0dB) indicates that more signal is required than noise. It should be noted that the signal-to-noise ratio in the present invention is used to describe an electronic signal.
The signal-to-noise ratio (SRR) is the ratio of the direct signal to the reflected signal. Wherein the direct signal is a signal that is directly transmitted to the feedforward microphone. The reflected signal is a signal that is reflected and ultimately picked up by the feedforward microphone.
Dynamic Range Compression (WDRC) for adjusting an input signal to within the Dynamic Range of the wearer's hearing. For example, the sound intensity suitable for the wearer to hear is around 90dB SPL (sound pressure level). If the energy strength of the input signal is 50dB SPL, a 35dB gain is obtained after WDRC is enabled. If the input signal is 65dB SPL, a gain of 25dB is obtained after starting the WDRC. If the input signal is 80dB SPL, 15dB gain is obtained after WDRC is enabled. It can be seen that the magnitude of the gain is gradually decreasing as the intensity of the acoustic input increases. That is, with WDRC, it is possible to compress the input range of 30dB (80 dB-50 dB) and the like to within the range of 10dB (in this example, the maximum output is 95dB SPL and the minimum output is 85dB SPL).
In the description of the present invention, it should be understood that the numerical references before the steps do not identify the order of performing the steps, but merely serve to facilitate the description of the present invention and to distinguish each step, and thus should not be construed as limiting the present invention.
Fig. 2 schematically shows an environmental application diagram of the multi-dynamic range companding method according to an embodiment of the present invention. The multi-dynamic range companding method may be implemented in a hearing aid 2.
The hearing aid 2 comprises a housing containing a feedforward microphone 21, a VAD element 22 (optional), a processor 23, a speaker 24.
A feed forward microphone 21 is located on the side of the hearing aid 2 remote from the ear canal (outside the housing) for acquiring the ambient signal of the wearer.
A VAD element 22 located on the other side of the housing, i.e. the VAD element 22 is located in or next to the ear canal of the wearer when the hearing aid is worn. The VAD element 22 may be used to obtain the feedback signal. The VAD element 22 may be a shock signal sensitive element, such as a bone conduction microphone or an acceleration sensor, among others. Since speech is conducted through the skull, the VAD element 22 readily detects whether the wearer is speaking.
A processor 23, electrically coupled to the feedforward microphone 21, the VAD element 22 and the speaker 24, is configured to process an input signal provided by the feedforward microphone 21, such as Wide Dynamic Range Compression (WDRC), beamforming, etc. In the present embodiment, the processor 23 may be a DSP (Digital Signal Processing) chip or the like.
And a speaker 24 for receiving the target signal processed by the processor 23 and outputting the target signal to the ear canal 4.
A silicone sleeve 25 for at least partial insertion into the ear canal 4 when the hearing aid 2 is worn. The silicone sleeve 25 may to some extent block the entry of sound around the wearer into the ear canal 4. Of course, the material of the silicone sleeve 25 may be replaced.
The present invention is directed to providing a multi-dynamic range companding scheme to enhance the hearing experience of a wearer.
(1) The input signal provided by the feedforward microphone 21 may include a noise signal with a large energy intensity. When the desired signal is amplified by the WDRC, the noise signal is also amplified, thereby affecting the hearing experience of the wearer.
Therefore, the present invention proposes a plurality of gain functions for selection.
And when the energy intensity of the noise signal in the input signal is larger, reducing the amplification ratio of the input signal. The method comprises the following specific steps:
when the SNR is more than 5dB, a first gain function is adopted;
when the SNR is less than or equal to 5dB, a second gain function is adopted;
the slope of the function curve corresponding to the first gain function is greater than the slope of the function curve corresponding to the second gain function.
As shown in fig. 3, a further scheme is as follows:
step S300: decomposing the input signal into N first subband signals S corresponding to N channels11、S12、…S1N。
Step S302: respectively analyzing the N first sub-band signals to obtain the N sub-band signal-to-noise ratios
Step S304: according to the signal-to-noise ratio of each sub-band, the N first sub-band signals are respectively subjected to dynamic range compression operation to obtain N target sub-band signals Sout。
By way of example, the step S304 may include steps S304A-S304D, wherein: step S304A: comparing a first threshold (e.g. 5dB) with an ith subband signal-to-noise ratio corresponding to an ith channel, wherein i is more than or equal to 1 and less than or equal to N, and is an integer; if the ith subband signal-to-noise ratio is greater than the first threshold, go to step S304B, otherwise go to step S304C. Step S304B: a first gain function of a plurality of gain functions is selected as an ith gain function corresponding to the i channels. Step S304C: taking a second gain function of the plurality of gain functions as the ith gain function; the slope of the first gain function is greater than the slope of the second gain function. Step S304D: generating N target subband signals from the N first subband signals: and according to the ith gain function and the ith sub-band energy of the ith channel, performing dynamic range compression on the ith first sub-band signal corresponding to the ith channel to generate an ith target sub-band signal corresponding to the ith channel.
One specific example is provided below: the input signal provided by the feedforward microphone 21 is divided into N frequency bands (channels) from the frequency content using a multi-channel filter with divided frequency bands. The energy S of the first sub-band signal in each channel is measuredin(i) (i.e., energy intensity, in sound pressure level dB SPL). And obtaining an ith target subband signal of the ith channel after the ith subband signal is compressed in a dynamic range according to the ith first subband signal of the ith channel and the corresponding compression rule (gain function) of the human ear hearing curve. As shown in fig. 4, which shows a function curve of the gain function corresponding to the ith channel. In FIG. 4, Sin(i) Representing the energy, S, of the ith first subband signalout(i) And representing the energy of the ith target subband signal obtained by compressing the dynamic range of the ith first subband signal. The function curve is the slope, i.e. the compression ratio (r: 1). The example implements differentiated dynamic range compression for each channel, and user experience is better.
Step S306: and synthesizing the N target subband signals to obtain the target signal.
(2) The input signal provided by the feedforward microphone 21 may include a reverberation signal with a large energy intensity. When the desired signal is amplified by the WDRC, the reverberant signal is also amplified, thereby affecting the hearing experience of the wearer.
Thus, the present invention proposes a plurality of gain functions.
And when the energy intensity of the reverberation signal in the input signal is larger, reducing the amplification ratio of the input signal. The method comprises the following specific steps:
when the SRR is more than 5dB, a first gain function is adopted;
when the SRR is less than or equal to 5dB, a second gain function is adopted;
the slope of the function curve corresponding to the first gain function is greater than the slope of the function curve corresponding to the second gain function.
As shown in fig. 5, the hearing aid 2 may replace the signal-to-noise ratio with the detected signal-to-noise ratio, and determine the gain function of each channel by comparing the first threshold with the ith subband signal-to-noise ratio corresponding to the ith channel, and other steps are not repeated.
(3) The input signal provided by the feedforward microphone 21 may comprise the wearer's own speech signal. When the required signal is amplified by the WDRC, the wearer's own voice signal is also amplified, thereby affecting the wearer's hearing experience.
Therefore, as shown in fig. 6 or fig. 7, the present invention also provides the following solutions:
the hearing aid 2 can detect whether the wearer is speaking by means of the VAD component 22. When it is detected that the wearer is speaking, it is necessary to prevent the wearer's own voice signal from being excessively amplified when performing dynamic range compression.
(4) The input signal provided by the feedforward microphone 21 may comprise an impulse signal. When the desired signal is amplified by the WDRC, the shock signal is also amplified, thereby affecting the hearing experience of the wearer and even impairing hearing.
Therefore, the present invention also provides the following solutions:
the hearing aid 2 may detect whether the input signal comprises a shock signal and, if the input signal comprises a shock signal, adjust the amplification strategy for dynamic range compression, e.g. reduce the amplification of the full band dynamic range compression in a shorter time.
In the embodiment, whether the input signal is the impact signal or not can be detected through the time domain signal energy, the calculation process is simple, the calculation resource consumption is low, the reaction speed is high, and the better hearing experience of a wearer is guaranteed.
The detection scheme of the impact signal may be as follows.
The hearing aid 2 may further be provided with a reference microphone positioned at a side remote from the ear canal 2 for collecting ambient signals. The sensitivity of the reference microphone is greater than the sensitivity of the feedforward microphone.
For a reference microphone with higher sensitivity: the dynamic range is reduced and the amplitude of the signal sent to the processor 23 is larger. When the impulse signal is included in the signal, it causes the signal acquired by the processor 23 to be saturated.
Due to the saturation phenomenon, the amplitude of the impulse signal obtained by the processor 23 from the reference microphone 22 of higher sensitivity is limited. That is, saturation of the signal energy of the impact signal obtained by the processor 23 from the higher sensitivity reference microphone 22 may occur. Thus, the time domain energy difference between the shock signal obtained by the processor 23 from the higher sensitivity reference microphone 22 and the shock signal obtained from the lower sensitivity feedforward microphone 21 is reduced (e.g., less than a particular threshold). For example, the feedforward microphone 21 employs a normal sensitivity microphone (e.g., -38dBV in sensitivity), and the reference microphone 22 employs an ultra-high sensitivity microphone (e.g., -23dBV in sensitivity). Thus, when the time domain energy difference between the impulse signal from the higher sensitivity reference microphone 22 and the impulse signal from the lower sensitivity feedforward microphone 21 is less than 15dB, it is an indication that an impulse signal may be present in the input signal, and the lower the time domain energy difference, the greater the probability of an impulse signal being present.
Based on the above analysis, it can be determined whether there is an impact signal based on the energy ratio.
Firstly, acquiring an input signal through a feedforward microphone 21, and calculating first time domain signal energy of the input signal;
acquiring a reference signal through a reference microphone 22, and calculating second time domain signal energy of the reference signal;
comparing the time domain energy difference between the time domain signal energy of the input signal and the time domain signal energy of the reference signal, and judging whether an impact signal exists in the surrounding environment of the wearer or not according to the time domain energy difference.
And if the time domain energy difference is larger than a preset energy difference threshold value (15dB), judging that an impact signal exists.
Further, the present inventors found that:
since the frequency range covered by the impact signal is the full frequency band, the voice is mainly focused on 300-.
Therefore, a weighting value can be increased for discrimination of a channel portion of 4ihz or more.
And the weighted value of the low-frequency part is lower, the scheme can effectively resist the interference of voice, so that the judgment of the impact signal is more robust.
In order to further improve the accuracy of the judgment on the impact signal, whether the impact signal exists can be judged by the following steps:
the method comprises the following steps: performing multi-band filtering on an input signal or a reference signal respectively to obtain N first sub-band signals corresponding to N channels and N second sub-band signals corresponding to the N channels, wherein the N first sub-band signals are obtained according to the input signal, and the N second sub-band signals are obtained according to the reference signal;
step two: calculating the time domain energy difference between the sub-band average energy of the first sub-band signal in the ith channel and the sub-band instantaneous peak energy of the first sub-band signal and the time domain energy difference between the sub-band average energy of the second sub-band signal in the ith channel and the sub-band instantaneous peak energy of the second sub-band signal to obtain the ith judgment result corresponding to the ith channel, thereby obtaining N judgment results corresponding to N channels, wherein i is a natural number, and i is more than or equal to 1 and less than or equal to N;
step three: respectively configuring a weight value for each channel, wherein the channel with the frequency point higher than 4ihz is configured with a higher weight value, and the channel with the frequency point lower than 4ihz is configured with a lower weight value;
step four: and comprehensively judging whether an impact signal exists according to each judgment result in the N judgment results and the corresponding weight value.
For example, when there is an impact signal, the corresponding discrimination result is 1; when no impact signal exists, the corresponding judgment result is-1; the channel with the frequency point higher than 4ihz has the weight value of 0.5, and the channel with the frequency point lower than 4ihz has the weight value of 0.2. Wherein, the influence of the judgment result of each channel in the comprehensive judgment is as follows: and judging the weight value of the channel.
The above-mentioned integrated weighted value can be compressed by sigma function, etc. to obtain probability value between 0-1.
The higher the probability value, the greater the likelihood of the presence of an impulse signal, and the greater the degree to which the input signal is suppressed in the time domain.
The lower the probability value, the less the degree of suppression.
(5) The input signal provided by the feedforward microphone 21 may comprise a wind noise signal. When the desired signal is amplified by the WDRC, the wind noise signal is also amplified, thereby affecting the hearing experience of the wearer.
Therefore, the present invention also provides the following solutions:
the hearing aid 2 may detect whether the input signal comprises a wind noise signal. When it is detected that the input signal includes the wind noise signal, it is necessary to reduce the amplification of the low-frequency range dynamic range compression when performing the dynamic range compression.
The above-mentioned schemes (1) to (5) can be freely combined to obtain different combination schemes as required.
As an example, scheme (1) and scheme (2) are combined to jointly determine the gain function adopted by the ith channel according to the subband signal-to-noise ratio and the subband signal-to-noise ratio in the ith channel, thereby further optimizing the user experience.
As an example, the scheme (1) and the scheme (3) are combined to determine the gain function adopted by the ith channel according to the sub-band signal-to-noise ratio in the ith channel, and when it is determined that the ith channel contains the wearer's own voice signal, the gain function obtained by the scheme (1) is further adjusted, for example, the compression ratio (i.e., the slope) is further reduced.
A number of embodiments are provided below, and the various embodiments provided below may be used to implement the multi-dynamic range companding method described above. For ease of understanding, the following description will exemplarily describe the hearing aid 2 as the execution body.
Example one
Fig. 8 is a flow chart schematically illustrating a multi-dynamic range companding method according to a first embodiment of the present invention.
As shown in fig. 8, the multi-dynamic range companding method may include steps S800 to S806, wherein:
in step S800, an input signal provided by the feedforward microphone 21 is acquired.
The input signal is collected in the environment around the wearer and comprises a voice signal, a noise signal and the like.
The voice signal may include the voice signal of another person, or may include the wearer's own voice signal.
The noise signals include various background noise signals, impact signals such as car horns, reverberation signals, wind noise signals, and the like.
Step S802, analyzing signal characteristics of the input signal, where the signal characteristics include a signal-to-noise ratio and/or a signal-to-mixing ratio.
In this embodiment, the signal characteristics include a signal-to-noise ratio and a signal-to-mixing ratio. In some further embodiments, the signal features may further include: whether it includes an impact signal, whether it includes the wearer's own voice signal, whether it includes a wind noise signal, etc. In combination with these signal characteristics, may be used to select and adjust the gain function in the WDRC.
Step S804, selecting the target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio.
The hearing aid 2 is provided with a number of preset gain functions for selection.
The larger the signal-to-noise ratio is, the smaller the noise signal in the input signal is;
the smaller the signal-to-noise ratio, the larger the noise signal in the input signal.
When the noise signal is large, if the noise signal is excessively amplified, the user hearing experience may be affected.
Thus, the target gain function may be determined based on the signal-to-noise ratio:
when the signal-to-noise ratio is small, a gain function with a more gentle amplification curve is set as the target gain function, and the noise is prevented from being excessively amplified.
Step S806, according to the energy of the input signal and the target gain function, performing dynamic range compression on the input signal to obtain a target signal.
The multi-dynamic range companding method provided by the embodiment selects the target gain function of dynamic range compression through the signal-to-noise ratio and/or the signal-to-mixing ratio. When the signal-to-noise ratio or the signal-to-noise ratio is small (i.e., the time-out signal or the reverberation signal is large), a target gain function with a more gradual amplification curve can be selected to prevent the noise or the reverberation from being excessively amplified.
In an exemplary embodiment, the signal characteristic of the input signal comprises N subband signal characteristics, N being an integer > 1. As shown in fig. 9, the step S802 may include steps S900 to S902, wherein: step S900, decomposing the input signal into N first subband signals S corresponding to N channels11、S12、…S1N(ii) a Step S902, analyzing the N first subband signals respectively to obtain the N subband signal features; wherein each sub-band signal characteristic comprises a sub-band signal-to-noise ratio and/or a sub-band signal-to-noise ratio of the respective first sub-band signal. By obtaining the sub-band signal characteristics of each first sub-band signal, more targeted processing can be achieved.
In an exemplary embodiment, the target gain function includes N gain functions corresponding to the N channels, and the N channels are in one-to-one correspondence with the N gain functions. As shown in fig. 10, the step S804 may include steps S1000 to S1004, in which: step S1000, comparing a first threshold value with an ith sub-band signal-to-noise ratio corresponding to an ith channel, wherein i is more than or equal to 1 and less than or equal to N, and i is an integer; step S1002, if the ith subband signal-to-noise ratio is greater than the first threshold, selecting a first gain function of a plurality of gain functions as an ith gain function corresponding to the i channels, where the plurality of gain functions are preset; step S1004, if the ith subband signal-to-noise ratio is not greater than the first threshold, taking a second gain function of the plurality of gain functions as the ith gain function; the slope of the first gain function is greater than the slope of the second gain function. When the signal-to-noise ratio or the signal-to-noise ratio of a certain channel is small (that is, the energy intensity of the noise signal or the reverberation signal is large), a gain function with a more gradual amplification curve can be selected for the channel, so that the difference processing is performed on the first subband signal in the channel.
In an exemplary embodiment, the energy of the input signal includes N subband energies corresponding to the N channels. As shown in fig. 11, the step S804 may include steps S1100 to S1102, in which: step S1100, generating N target subband signals according to the N first subband signals, includes: performing dynamic range compression on an ith first sub-band signal corresponding to the ith channel according to the ith gain function and an ith sub-band energy of the ith channel to generate an ith target sub-band signal corresponding to the ith channel; and step S1102, synthesizing the N target subband signals to obtain the target signal. The hearing aid 2 may further determine whether the amplitude of the target signal is greater than a predetermined threshold. And if the amplitude of the target signal is greater than the preset threshold value, carrying out amplitude limiting processing on the target signal to prevent signal distortion.
In an exemplary embodiment, the hearing aid further comprises a VAD element, the VAD element being proximate to the ear canal of the wearer when the hearing aid is worn. As shown in FIG. 12, the method may further include steps S1200 to 1204, wherein: step S1200, acquiring a feedback signal provided by the VAD element; step S1202 of decomposing the feedback signal into N second subband signals S corresponding to the N channels21、S22、…S2NWherein the second subband signal S2iCorresponding first subband signal S1iCorresponding to the same channel; step S1204, according to each second sub-band signal, judging whether each channel includes the sub-band signalThe wearer's own voice signal; step S1206, adjusting the gain function of each channel according to the judgment result of each channel; each channel corresponds to a judgment result, the judgment result is a first judgment result or a second judgment result, the first judgment result is used for indicating that the corresponding channel comprises the sound signal of the wearer and indicating that the slope of the gain function of the corresponding channel is reduced, and the second judgment result is used for indicating that the corresponding channel does not comprise the sound signal of the wearer. For example, when it is determined that the channel corresponding to the ith second sub-band signal includes the sound signal of the wearer, the gain function of the channel is adjusted, for example, the slope of the channel is adjusted to be smaller, so as to prevent the sound signal of the wearer from being excessively amplified, thereby improving the hearing experience of the user.
In an exemplary embodiment, as shown in FIG. 13, the method may further include steps S1300-1302, wherein: step 1300, judging whether each channel comprises an impact signal; in step S1302, if one of the channels includes the impact signal, the gain function of the channel is adjusted. For example, when it is determined that a channel includes an impulse signal, the gain function of the channel may be adjusted. For example, the slope of this channel is adjusted down to prevent over-amplification of the impulse signal within this channel, thereby enhancing the user's listening experience.
In an exemplary embodiment, as shown in FIG. 14, the method may further include steps S1400 to 1402, wherein: step S1400, judging whether the input signal comprises a wind noise signal; step S1402, if the input signal includes the wind noise signal, adjusting a gain function of each channel lower than a preset frequency point. For example, when it is determined that a wind noise signal is included in a channel, the gain function of the channel may be adjusted. For example, the slope of this channel is adjusted down to prevent over-amplification of the wind noise signal within this channel, thereby enhancing the user listening experience.
It should be noted that, when it is detected that a certain channel includes an impact signal, a wind noise signal, and a sound signal of the wearer, the gain function corresponding to the channel may be adjusted together according to the weight value of the signal characteristic.
For ease of understanding, as shown in fig. 15, one specific example is provided below.
Step S1, acquiring an input signal, which includes the signal of the feedforward microphone 21;
step S2, analyzing the input signal by a filter bank to obtain a plurality of first subband signals corresponding to a plurality of channels;
step S3, determining the gain function of the dynamic range compression of each channel;
the gain function for dynamic range compression of each channel, the influencing factors may include the following:
first, signal-to-noise ratio;
second, signal-to-mixture ratio;
thirdly, impact noise;
fourth, the wearer's own voice signal;
fifthly, wind noise;
taking channel a as an example:
(1) the hearing aid 2 may determine a gain function Y based on the signal-to-noise ratio of the channel a;
(2) the gain of the input signal is finely adjusted on the basis of the gain function Y, based on the root signal-to-noise ratio, the impulse noise, the wearer's own voice signal and the wind noise, so as to further optimize the dynamic range compression effect corresponding to the channel a.
Step S4: performing corresponding diffusion processing on the first sub-band signals in each channel according to the gain function of each channel;
step S5: synthesizing each target subband signal obtained based on the diffusion in step S4 to obtain a target signal;
step S6: and outputting the target signal.
Example two
As shown in fig. 16, fig. 16 schematically shows a block diagram of a multi-dynamic range companding system 1600 according to a second embodiment of the invention. The multi-dynamic range companding system 1600 is used in a hearing aid comprising a feed forward microphone and a VAD element, said VAD element being located in the ear canal of the wearer when the hearing aid is worn. The system may be partitioned into one or more program modules, which are stored in a storage medium and executed by one or more processors to implement embodiments of the invention. The program modules referred to in the embodiments of the present invention refer to a series of computer program instruction segments that can perform specific functions, and the following description will specifically describe the functions of the program modules in the embodiments.
As shown in fig. 16, the multi-dynamic range companding system 1600 can include an acquisition module 1610, an analysis module 1620, a selection module 1630, and a processing module 1640. Wherein:
an obtaining module 1610 configured to obtain an input signal provided by the feedforward microphone;
an analysis module 1620 configured to analyze a signal characteristic of the input signal, where the signal characteristic includes a signal-to-noise ratio and/or a signal-to-mixing ratio;
a selecting module 1630, configured to select a target gain function for the dynamic range compression according to the snr and/or the snr;
the processing module 1640 is configured to perform dynamic range compression on the input signal according to the energy of the input signal and the target gain function to obtain a target signal.
Optionally, the signal characteristics of the input signal include N subband signal characteristics, where N is an integer > 1; the analysis module 1620, further configured to:
decomposing the input signal into N first subband signals S corresponding to N channels11、S12、…S1N(ii) a And
analyzing the N first sub-band signals respectively to obtain the N sub-band signal characteristics;
wherein each sub-band signal characteristic comprises a sub-band signal-to-noise ratio and/or a sub-band signal-to-noise ratio of the respective first sub-band signal.
Optionally, the target gain function includes N gain functions corresponding to the N channels, and the N channels are in one-to-one correspondence with the N gain functions; the selecting module 1630 is further configured to:
comparing the first threshold with the ith sub-band signal-to-noise ratio corresponding to the ith channel, wherein i is more than or equal to 1 and less than or equal to N, and is an integer;
if the ith subband signal-to-noise ratio is larger than the first threshold, selecting a first gain function of a plurality of gain functions as an ith gain function corresponding to the i channels, wherein the gain functions are preset;
if the ith subband signal-to-noise ratio is not greater than the first threshold, taking a second gain function of the plurality of gain functions as the ith gain function; the slope of the first gain function is greater than the slope of the second gain function.
Optionally, the energy of the input signal comprises N subband energies corresponding to the N channels; the processing module 1640 is further configured to:
generating N target subband signals from the N first subband signals, comprising: performing dynamic range compression on an ith first sub-band signal corresponding to the ith channel according to the ith gain function and an ith sub-band energy of the ith channel to generate an ith target sub-band signal corresponding to the ith channel; and
and synthesizing the N target subband signals to obtain the target signal.
Optionally, the hearing aid further comprises a VAD element proximate to the ear canal of the wearer when the hearing aid is worn; the system 1600 further includes a first adjustment module to:
obtaining a feedback signal provided by the VAD element;
decomposing the feedback signal into N second subband signals S corresponding to the N channels21、S22、…S2NWherein the second subband signal S2iCorresponding first subband signal S1iCorresponding to the same channel; and
judging whether each channel comprises the own sound signal of the wearer or not according to each second sub-band signal; and
adjusting the gain function of each channel according to the judgment result of each channel; each channel corresponds to a judgment result, the judgment result is a first judgment result or a second judgment result, the first judgment result is used for indicating that the corresponding channel comprises the sound signal of the wearer and indicating that the slope of the gain function of the corresponding channel is reduced, and the second judgment result is used for indicating that the corresponding channel does not comprise the sound signal of the wearer.
Optionally, the system 1600 further includes a first adjusting module, and the second adjusting module is configured to:
judging whether each channel comprises an impact signal or not;
if one of the channels includes the impact signal, the gain function of that channel is adjusted.
Optionally, the system 1600 further includes a first adjusting module, and the second adjusting module is configured to:
judging whether the input signal comprises a wind noise signal;
and if the input signal comprises the wind noise signal, adjusting the gain function of each channel lower than a preset frequency point.
EXAMPLE III
As shown in fig. 17, fig. 17 schematically shows a hardware architecture diagram of a computer device 1700 suitable for implementing the multi-dynamic range companding method according to the third embodiment of the present invention. In the present embodiment, the computer device 1700 is a device capable of automatically performing numerical calculation and/or information processing in accordance with a preset or stored instruction, and may be, for example, a hearing aid, a hearing assistance device having a hearing aid function, or the like. As shown in fig. 17, computer device 1700 includes at least, but is not limited to: the memory 1710, processor 1720, and network interface 1730 may be communicatively linked to each other via a system bus. Wherein:
the memory 1710 includes at least one type of computer-readable storage medium including flash memory, a hard disk, a multimedia card, a card-type memory (e.g., SD or DX memory, etc.), a random access memory (RAN), a static random access memory (SRAN), a read-only memory (RON), an electrically erasable programmable read-only memory (EEPRON), a programmable read-only memory (PRON), a magnetic memory, a magnetic disk, an optical disk, and so forth. In some embodiments, memory 1710 may be an internal storage module of the computer device 1700, such as a hard disk or memory of the computer device 1700. In other embodiments, the memory 1710 may also be an external storage device of the computer device 1700, such as a plug-in hard disk provided on the computer device 1700, a smart memory Card (SNC), a Secure Digital (SD) Card, a Flash memory Card (Flash Card), and so on. Of course, memory 1710 may also include both internal and external memory modules of computer device 1700. In this embodiment, the memory 1710 is generally used for storing an operating system and various types of application software installed in the computer apparatus 1700, such as program codes of the multi-dynamic range companding method. In addition, the memory 1710 can also be used to temporarily store various types of data that have been output or are to be output.
Processor 1720 may be a Central Processing Unit (CPU), controller, microcontroller, microprocessor, or other data Processing chip in some embodiments. The processor 1720 generally serves to control overall operation of the computer device 1700, such as performing control and processing related to data interaction or communication with the computer device 1700. In this embodiment, processor 1720 is configured to execute program code stored in memory 1710 or to process data.
Network interface 1730 may include a wireless network interface or a wired network interface, and network interface 1730 is typically used to establish communication links between computer device 1700 and other computer devices. For example, the network interface 1730 is used to connect the computer device 1700 to an external terminal via a network, to establish a data transmission channel and a communication link between the computer device 1700 and the external terminal, and the like. The network may be a wireless or wired network such as an Intranet (Internet), the Internet (Internet), a Global system of mobile communication (GSN), Wideband Code Division multiple Access (WCDNA), a 4G network, a 5G network, Bluetooth (Bluetooth), Wi-Fi, and the like. A
It should be noted that FIG. 17 only shows a computer device having components 1710 and 1730, but it should be understood that not all of the shown components are required to be implemented, and that more or fewer components may be implemented instead.
In this embodiment, the multi-dynamic range companding method stored in the memory 1710 can be further divided into one or more program modules and executed by one or more processors (in this embodiment, the processor 1720) to implement the embodiments of the present invention.
Example four
The present invention also provides a computer-readable storage medium having stored thereon a computer program which, when executed by a processor, implements the steps of the multi-dynamic range companding method in embodiments.
In this embodiment, the computer-readable storage medium includes a flash memory, a hard disk, a multimedia card, a card type memory (e.g., SD or DX memory, etc.), a random access memory (RAN), a static random access memory (SRAN), a read only memory (RON), an electrically erasable programmable read only memory (EEPRON), a programmable read only memory (PRON), a magnetic memory, a magnetic disk, an optical disk, and the like. In some embodiments, the computer readable storage medium may be an internal storage unit of the computer device, such as a hard disk or a memory of the computer device. In other embodiments, the computer readable storage medium may also be an external storage device of the computer device, such as a plug-in hard disk equipped on the computer device, a smart memory Card (SNart new Card, SNC for short), a Secure Digital (SD) Card, a Flash memory Card (Flash Card), and so on. Of course, the computer-readable storage medium may also include both internal and external storage devices of the computer device. In this embodiment, the computer-readable storage medium is generally used for storing an operating system and various types of application software installed in the computer device, for example, the program code of the multi-dynamic-range companding method in the embodiment, and the like. Further, the computer-readable storage medium may also be used to temporarily store various types of data that have been output or are to be output.
It will be apparent to those skilled in the art that the modules or steps of the embodiments of the invention described above may be implemented by a general purpose computing device, they may be centralized on a single computing device or distributed across a network of multiple computing devices, and alternatively, they may be implemented by program code executable by a computing device, such that they may be stored in a storage device and executed by a computing device, and in some cases, the steps shown or described may be performed in an order different than that described herein, or they may be separately fabricated into individual integrated circuit modules, or multiple ones of them may be fabricated into a single integrated circuit module. Thus, embodiments of the invention are not limited to any specific combination of hardware and software.
The above description is only a preferred embodiment of the present invention, and not intended to limit the scope of the present invention, and all modifications of equivalent structures and equivalent processes, which are made by using the contents of the present specification and the accompanying drawings, or directly or indirectly applied to other related technical fields, are included in the scope of the present invention.
Claims (10)
1. A multi-dynamic range companding method for use in a hearing aid, said hearing aid comprising a feedforward microphone located on a side of said hearing aid remote from an ear canal; the method comprises the following steps:
acquiring an input signal provided by the feedforward microphone;
analyzing signal characteristics of the input signal, the signal characteristics including a signal-to-noise ratio and/or a signal-to-mixing ratio;
selecting a target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio;
and according to the energy of the input signal and the target gain function, carrying out dynamic range compression on the input signal to obtain a target signal.
2. The multi-dynamic range companding method according to claim 1, wherein the signal characteristics of the input signal comprise N subband signal characteristics, N being an integer > 1; the step of analyzing the input signal characteristic of the input signal comprises:
decomposing the input signal into N first subband signals S corresponding to N channels11、S12、…S1N(ii) a And
analyzing the N first sub-band signals respectively to obtain the N sub-band signal characteristics;
wherein each sub-band signal characteristic comprises a sub-band signal-to-noise ratio and/or a sub-band signal-to-noise ratio of the respective first sub-band signal.
3. The multi-dynamic range companding method according to claim 2, wherein the target gain function comprises N gain functions corresponding to the N channels, the N channels being in one-to-one correspondence with the N gain functions; the step of selecting the target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio comprises:
comparing the first threshold with the ith sub-band signal-to-noise ratio corresponding to the ith channel, wherein i is more than or equal to 1 and less than or equal to N, and is an integer;
if the ith subband signal-to-noise ratio is larger than the first threshold, selecting a first gain function of a plurality of gain functions as an ith gain function corresponding to the i channels, wherein the gain functions are preset;
if the ith subband signal-to-noise ratio is not greater than the first threshold, taking a second gain function of the plurality of gain functions as the ith gain function; the slope of the first gain function is greater than the slope of the second gain function.
4. The multi-dynamic range companding method according to claim 3, wherein the energy of the input signal comprises N subband energies corresponding to the N channels, and the step of performing dynamic range compression on the input signal according to the energy of the input signal and the target gain function to obtain the target signal comprises:
generating N target subband signals from the N first subband signals, comprising: performing dynamic range compression on an ith first sub-band signal corresponding to the ith channel according to the ith gain function and an ith sub-band energy of the ith channel to generate an ith target sub-band signal corresponding to the ith channel; and
and synthesizing the N target subband signals to obtain the target signal.
5. The multi-dynamic range companding method according to claim 4, wherein the hearing aid further comprises a VAD element, which is proximate to the ear canal of the wearer when the hearing aid is worn; the method further comprises the following steps:
obtaining a feedback signal provided by the VAD element;
decomposing the feedback signal into N second subband signals S corresponding to the N channels21、S22、…S2NWherein the second subband signal S2iCorresponding first subband signal S1iCorresponding to the same channel; and
judging whether each channel comprises the own sound signal of the wearer or not according to each second sub-band signal; and
adjusting the gain function of each channel according to the judgment result of each channel; each channel corresponds to a judgment result, the judgment result is a first judgment result or a second judgment result, the first judgment result is used for indicating that the corresponding channel comprises the sound signal of the wearer and indicating that the slope of the gain function of the corresponding channel is reduced, and the second judgment result is used for indicating that the corresponding channel does not comprise the sound signal of the wearer.
6. The multi-dynamic range companding method according to claim 4, further comprising:
judging whether each channel comprises an impact signal or not;
if one of the channels includes the impact signal, the gain function of that channel is adjusted.
7. The multi-dynamic range companding method according to claim 4, further comprising:
judging whether the input signal comprises a wind noise signal;
and if the input signal comprises the wind noise signal, adjusting the gain function of each channel lower than a preset frequency point.
8. A multi-dynamic range companding system for use in a hearing aid, said hearing aid comprising a feedforward microphone positioned on a side of said hearing aid remote from an ear canal, said system comprising:
the acquisition module is used for acquiring an input signal provided by the feedforward microphone;
an analysis module for analyzing signal characteristics of the input signal, the signal characteristics including a signal-to-noise ratio and/or a signal-to-mixing ratio;
a selection module, configured to select a target gain function of the dynamic range compression according to the signal-to-noise ratio and/or the signal-to-mixing ratio;
and the processing module is used for compressing the dynamic range of the input signal according to the energy of the input signal and the target gain function to obtain a target signal.
9. A computer device comprising a memory, a processor and a computer program stored on the memory and executable on the processor, characterized in that the processor, when executing the computer program, is adapted to carry out the steps of the multi-dynamic range companding method according to any of claims 1-7.
10. A computer-readable storage medium storing a computer program executable by at least one processor to cause the at least one processor to perform the steps of the multi-dynamic range companding method according to any one of claims 1-7.
Priority Applications (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN202110414005.9A CN113132882B (en) | 2021-04-16 | 2021-04-16 | Multi-dynamic-range companding method and system |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN202110414005.9A CN113132882B (en) | 2021-04-16 | 2021-04-16 | Multi-dynamic-range companding method and system |
Publications (2)
Publication Number | Publication Date |
---|---|
CN113132882A true CN113132882A (en) | 2021-07-16 |
CN113132882B CN113132882B (en) | 2022-10-28 |
Family
ID=76776984
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
CN202110414005.9A Active CN113132882B (en) | 2021-04-16 | 2021-04-16 | Multi-dynamic-range companding method and system |
Country Status (1)
Country | Link |
---|---|
CN (1) | CN113132882B (en) |
Citations (10)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4811404A (en) * | 1987-10-01 | 1989-03-07 | Motorola, Inc. | Noise suppression system |
EP1469703A2 (en) * | 2004-04-30 | 2004-10-20 | Phonak Ag | Method of processing an acoustical signal and a hearing instrument |
US20050244023A1 (en) * | 2004-04-30 | 2005-11-03 | Phonak Ag | Method of processing an acoustic signal, and a hearing instrument |
CN101593522A (en) * | 2009-07-08 | 2009-12-02 | 清华大学 | A kind of full frequency domain digital hearing aid method and apparatus |
CN104168024A (en) * | 2013-05-15 | 2014-11-26 | 英飞凌科技股份有限公司 | Apparatus and method for controlling an amplification gain of an amplifier |
CN206077668U (en) * | 2016-09-06 | 2017-04-05 | 佛山科学技术学院 | A kind of Audio Processing control system for possessing feedback suppression and decrease of noise functions |
CN108200523A (en) * | 2016-11-24 | 2018-06-22 | 奥迪康有限公司 | Include the hearing devices of self voice detector |
CN108235211A (en) * | 2016-12-22 | 2018-06-29 | 奥迪康有限公司 | Hearing devices and its operation method including dynamic compression amplification system |
CN109804559A (en) * | 2016-09-28 | 2019-05-24 | 诺基亚技术有限公司 | Gain control in spatial audio systems |
US20200027467A1 (en) * | 2018-07-20 | 2020-01-23 | Mimi Hearing Technologies GmbH | Systems and methods for encoding an audio signal using custom psychoacoustic models |
-
2021
- 2021-04-16 CN CN202110414005.9A patent/CN113132882B/en active Active
Patent Citations (10)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4811404A (en) * | 1987-10-01 | 1989-03-07 | Motorola, Inc. | Noise suppression system |
EP1469703A2 (en) * | 2004-04-30 | 2004-10-20 | Phonak Ag | Method of processing an acoustical signal and a hearing instrument |
US20050244023A1 (en) * | 2004-04-30 | 2005-11-03 | Phonak Ag | Method of processing an acoustic signal, and a hearing instrument |
CN101593522A (en) * | 2009-07-08 | 2009-12-02 | 清华大学 | A kind of full frequency domain digital hearing aid method and apparatus |
CN104168024A (en) * | 2013-05-15 | 2014-11-26 | 英飞凌科技股份有限公司 | Apparatus and method for controlling an amplification gain of an amplifier |
CN206077668U (en) * | 2016-09-06 | 2017-04-05 | 佛山科学技术学院 | A kind of Audio Processing control system for possessing feedback suppression and decrease of noise functions |
CN109804559A (en) * | 2016-09-28 | 2019-05-24 | 诺基亚技术有限公司 | Gain control in spatial audio systems |
CN108200523A (en) * | 2016-11-24 | 2018-06-22 | 奥迪康有限公司 | Include the hearing devices of self voice detector |
CN108235211A (en) * | 2016-12-22 | 2018-06-29 | 奥迪康有限公司 | Hearing devices and its operation method including dynamic compression amplification system |
US20200027467A1 (en) * | 2018-07-20 | 2020-01-23 | Mimi Hearing Technologies GmbH | Systems and methods for encoding an audio signal using custom psychoacoustic models |
Also Published As
Publication number | Publication date |
---|---|
CN113132882B (en) | 2022-10-28 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
US9305568B2 (en) | Active acoustic filter with socially determined location-based filter characteristics | |
CN101208742B (en) | Adapted audio response | |
US8290190B2 (en) | Method for sound processing in a hearing aid and a hearing aid | |
US10631105B2 (en) | Hearing aid system and a method of operating a hearing aid system | |
CN112954115B (en) | Volume adjusting method and device, electronic equipment and storage medium | |
US10499167B2 (en) | Method of reducing noise in an audio processing device | |
CN107454537B (en) | Hearing device comprising a filter bank and an onset detector | |
CN110840462A (en) | Hearing-aid method for human ears based on earphone, computer readable storage medium and Bluetooth earphone | |
US11445307B2 (en) | Personal communication device as a hearing aid with real-time interactive user interface | |
CN113132880B (en) | Impact noise suppression method and system based on dual-microphone architecture | |
CN113132882B (en) | Multi-dynamic-range companding method and system | |
JP2008522511A (en) | Method and apparatus for adaptive speech processing parameters | |
US9992583B2 (en) | Hearing aid system and a method of operating a hearing aid system | |
EP3704870B1 (en) | Method for processing an acoustic speech input signal and audio processing device | |
KR102403996B1 (en) | Channel area type of hearing aid, fitting method using channel area type, and digital hearing aid fitting thereof | |
NL2021071B1 (en) | Method for processing an audio signal for a hearing aid | |
Ngo et al. | An integrated approach for noise reduction and dynamic range compression in hearing aids | |
JPH0956000A (en) | Hearing aid | |
CN117896646A (en) | Noise reduction method and device, electronic equipment and storage medium | |
CN116778949A (en) | Personalized loudness compensation method, device, computer equipment and storage medium |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
PB01 | Publication | ||
PB01 | Publication | ||
SE01 | Entry into force of request for substantive examination | ||
SE01 | Entry into force of request for substantive examination | ||
GR01 | Patent grant | ||
GR01 | Patent grant |