WO2010000042A1 - Amplification de gain linéaire pour des sons d’intensité moyenne à élevée dans un processeur de son compressif - Google Patents

Amplification de gain linéaire pour des sons d’intensité moyenne à élevée dans un processeur de son compressif Download PDF

Info

Publication number
WO2010000042A1
WO2010000042A1 PCT/AU2009/000870 AU2009000870W WO2010000042A1 WO 2010000042 A1 WO2010000042 A1 WO 2010000042A1 AU 2009000870 W AU2009000870 W AU 2009000870W WO 2010000042 A1 WO2010000042 A1 WO 2010000042A1
Authority
WO
WIPO (PCT)
Prior art keywords
signal
gain value
channel
sound
output
Prior art date
Application number
PCT/AU2009/000870
Other languages
English (en)
Inventor
Peter Blamey
Original Assignee
Peter Blamey
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Priority claimed from AU2008903428A external-priority patent/AU2008903428A0/en
Application filed by Peter Blamey filed Critical Peter Blamey
Publication of WO2010000042A1 publication Critical patent/WO2010000042A1/fr

Links

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • 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
    • H04R25/43—Electronic input selection or mixing based on input signal analysis, e.g. mixing or selection between microphone and telecoil or between microphones with different directivity characteristics
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/06—Transformation of speech into a non-audible representation, e.g. speech visualisation or speech processing for tactile aids
    • G10L2021/065—Aids for the handicapped in understanding
    • 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
    • H04R2225/41—Detection or adaptation of hearing aid parameters or programs to listening situation, e.g. pub, forest
    • 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
    • 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

Definitions

  • the present invention relates to sound processing devices in which an acoustic sound input or an electric or digital representation of an acoustic sound input is processed and converted to an acoustic sound output, and in particular relates to the processing of sound in the mid to high intensity part of the output sound range to improve speech intelligibility, sound quality and naturalness of the sound.
  • Sound processing devices of this kind are often used in hearing aids, assistive listening devices (ALD), and consumer audio devices such as radios, television sets, CD players, MP3 players, stereo systems, headsets, telephones, and mobile phone handsets.
  • ALD assistive listening devices
  • consumer audio devices such as radios, television sets, CD players, MP3 players, stereo systems, headsets, telephones, and mobile phone handsets.
  • the Global Medical Device Nomenclature Agency (GMDNS) definition of an ALD is an amplifying device, other than a hearing aid, for use by a hard of hearing person.
  • the sound output level In the field of consumer audio devices, it is common for the sound output level to be controlled by means of a volume control and for the spectral shape to be controlled by a bass or treble adjustment or a multi-channel graphic equalizer. These controls may be adjusted according to the listener's preferences, the input signal to the device, and the ambient sound in the listener's environment. By adjusting these controls, the listener is effectively optimizing the sound output from the device to fall within their preferred listening range in each frequency band taking into account their preferences, their hearing abilities, the sound they are listening to, and the other sounds in the environment.
  • AGCo automatic gain control
  • AVC automatic volume control
  • AGC and AVC do not automatically take into account the listener's preferences or hearing abilities.
  • HA hearing aids
  • CI cochlear implants
  • ALD assistive listening devices
  • AGC In conventional devices for people with normal or impaired hearing, AGC is typically described by an input-output function (for example, see Figure 3), or the relationship between the input level and the gain in the case of AGCi (see Figure 4), or the relationship between the output level and the gain in the case of AGCo (see Figure 5).
  • Figures 3, 4, and 5 are equivalent descriptions of the behaviour of an example AGC device, although the AGCo and AGCi versions would achieve this result with different implementations and methods.
  • the high compression regions are characterized by a gain that decreases without limit as the input intensity increases in the case of AGCi, or as the output intensity increases in the case of AGCo.
  • the maximum gain is the gain that applies in the linear region at low to mid intensity levels.
  • non-linear hearing aid fitting prescriptions such as the NAL-NLl prescription, the FIG6 prescription, the IHAFF prescription, and the DSL i/o prescription (see for example, figure 9.12 of Dillon, H., Hearing aids, Boomerang Press, 2001)
  • a maximum gain is prescribed for a linear region at low intensities and compression is applied at mid-to-high intensities.
  • the conventional rationale for this type of compression is that people with impaired hearing have a steeper than normal relationship between loudness and sound level referred to as recruitment (Fletcher & Munson, 1937).
  • recruitment Fletcher & Munson, 1937
  • the volume control typically controls the maximum gain that can be applied to the signal.
  • the maximum gain is controlled by means of a sensitivity control, and the maximum output level is controlled by means of a volume control.
  • Compression is also used in AVC systems to ensure that the signal remains in the range between the ambient noise level and the level at which the signal will become uncomfortably loud. As the ambient noise level rises, this target intensity range becomes narrower, and higher compression ratios are required. Compression is usually applied to the high intensity parts of the output signal to keep them below the discomfort threshold of hearing.
  • the normal human auditory system includes some elements that may be thought of as biological AGC systems.
  • the outer hair cells in the Organ of Corti of the cochlea provide additional gain to the motion of the basilar membrane at low intensities. As the input sound intensity increases, the additional gain provided by the outer hair cells decreases. This is sometimes called "cochlear compression".
  • the input output curve for the cochlear compression has its shallowest slope in the middle of the intensity range (see Figure 6). It has been reported that complete loss of outer hair cells can account for up to 60 dB of hearing loss.
  • the stapedius muscle and stapedial reflex can act to reduce the intensity of sound transmitted through the mechanical linkage of the middle ear when the sensation becomes too loud.
  • This action of the stapedius muscle is analogous to AGCo, and the action of the outer hair cells is analogous to AGCi.
  • the vocalization-induced stapedius reflex is analogous to a voice activated change in the input level (a form of AGCi).
  • the stapedius muscle and stapedial reflex are still functional in most people with impaired hearing.
  • AGC and AVC use compression in the mid-to-high intensity part of the input and output ranges, in order to compensate for hearing loss or rising ambient noise levels. Compression inevitably introduces distortion into the output signal.
  • the level of the distortion products is proportional to the rate of change of the gain of the device and the input level of the signal.
  • the application of high compression ratios with fast acting compression in the high intensity part of the output range is likely to generate audible distortion products that degrade the quality of the sound.
  • the compression will reduce the intensity difference between the speech and the noise, resulting in a decrease in intelligibility of the speech.
  • the compression at mid-to-high levels in the HA or ALD is attempting to compensate for a reduction in the outer hair cell compression which naturally occurs at low-to-mid intensity levels.
  • the naturally occurring compression that occurs through the action of the stapedius muscle will operate in addition to the effect of the AGC and/or AVC.
  • the device is quite sophisticated, it will be unable to differentiate between the wearer's own vocalization and other speakers' voices so that the effect of the vocalization-induced stapedial reflex and the device compression together will be to artificially reduce the apparent level of the device wearer's own voice relative to other speakers.
  • compression schemes are capable of implementing a linear input-output function throughout their whole range of operation by making the compression ratio 1:1 or by setting the "low kneepoint" to a high input level, however this does not achieve the requirement for the application of compression in the low-to-mid input level range.
  • a few hearing aid compression schemes are capable of implementing compression for low input intensities and a linear or approximately linear input- output function for high input intensities.
  • Goldstein used a model of cochlear compression which tended towards linearity at high input intensities
  • Armstrong, Sykes, and Csermak used a combination of compression and expansion at high intensities
  • Salmi and Scheller summed the outputs of a conventional compression scheme and a linear amplifier.
  • ADRO® Adaptive Dynamic Range Optimization
  • ADRO is an adaptive linear amplification scheme that slowly adjusts gain to compensate for long- term variations in the output signal dynamic range.
  • the peaks in the output signal represented by the 90 th percentile of the output level distribution for example, are adjusted to fall at or below a "comfort target” and the troughs in the output signal, represented by the 30 th percentile of the output level distribution for example, are adjusted to remain at or above an "audibility target".
  • These two adjustments are referred to as the "comfort rule” and the "audibility rule” respectively.
  • ADRO also use a "hearing protection rule” that applies fast infinite compression at high output levels to prevent the sound becoming uncomfortably loud and potentially damaging the listener's hearing, and a “background noise rule” that imposes a maximum gain to prevent soft background sounds from being over-amplified.
  • a “hearing protection rule” that applies fast infinite compression at high output levels to prevent the sound becoming uncomfortably loud and potentially damaging the listener's hearing
  • background noise rule that imposes a maximum gain to prevent soft background sounds from being over-amplified.
  • ADRO has been shown to produce more natural sound quality than schemes that use compression in the mid-to-high intensity range, and is preferred over compression by approximately 75% of hearing aid users.
  • ADRO is reported to make loud sounds too soft in some circumstances. This typically occurs when a loud sound is followed by a soft sound and the slow-acting ADRO rules may take ten seconds or more to increase the gain back to an appropriate level for the second softer sound.
  • the present invention provides a method for setting and implementing linear gain for mid-to-high intensity signals in a sound processor, the method comprising: providing amplification of the signal with gain under the control of a control module; constraining the gain of the amplifier to remain at or above a minimum gain value for mid-to- high intensity signals; setting, storing, and adjusting the minimum gain value; and calculating one or more appropriate values for the minimum gain value.
  • the present invention provides a sound processing device with linear gain for mid-to-high intensity signals, the sound processing device comprising: an amplifier with gain under the control of a control module; a control module that constrains the gain of the amplifier to remain at or above a minimum gain value for mid-to-high intensity signals; a means for setting, storing, and adjusting the minimum gain value; and a means for calculating one or more appropriate values for the minimum gain value.
  • the present invention provides a computer program product comprising computer program code means to make a computer execute a sound processing procedure with linear gain for mid-to-high intensity signals, the computer program product comprising: computer program means for providing amplification of the signal with gain under the control of a control module; computer program means for constraining the gain of the amplifier to remain at or above a minimum gain value for mid-to-high intensity signals; computer program means for setting, storing, and adjusting the minimum gain value; and computer program means for calculating one or more appropriate values for the minimum gain value.
  • the amplifier and control module preferably comprise a conventional AGCi, AGCo, or ADRO sound amplifier with gain control, modified to impose a minimum gain constraint in the mid-to-high intensity region.
  • the appropriate values for the minimum gain value are preferably calculated on the basis of extraneous factors comprising the hearing loss of the listener, the preferences of the listener, the ambient noise level in the vicinity of the sound processor, the setting of the volume control on the device, and the setting of the sensitivity control of the device.
  • the minimum gain constraint may be applied in at least one channel of an in-line signal processor. In another preferred embodiment, the minimum gain constraint may be applied in at least one channel of an off-line signal processor.
  • Figure 1 illustrates a block diagram for a sound processor with off-line processing elements that control an adaptive processor on the signal path.
  • Figure 2 illustrates a block diagram for a sound processor with in-line parallel signal processing paths.
  • Figure 3 shows an illustrative example of a typical input output function of a sound processor.
  • Figure 4 shows an illustrative example of a typical input gain function of an input controlled automatic gain control (AGCi) in a sound processor.
  • Figure 5 shows an illustrative example of a typical output gain function of an output controlled automatic gain control (AGCo) in a sound processor.
  • Figure 6 shows a theoretical example of the input output function of the outer hair cells in a normal human ear.
  • Figure 7 illustrates a generalized channel processor for an embodiment of the present invention with AGCi and minimum gain constraint.
  • Figure 8 illustrates a generalized channel processor for an embodiment of the present invention with AGCo and minimum gain constraint.
  • Figure 9 shows an illustrative example of a typical input output function of a sound processor with the minimum gain constraint applied in the mid-to-high intensity region.
  • Figure 10 shows an illustrative example of a typical input gain function of an input controlled automatic gain control (AGCi) in a sound processor with the minimum gain constraint applied in the mid-to-high intensity region.
  • AGCi input controlled automatic gain control
  • Figure 1 1 shows an illustrative example of a typical output gain function of an output controlled automatic gain control (AGCo) in a sound processor with the minimum gain constraint applied in the mid-to-high intensity region.
  • AGCo output controlled automatic gain control
  • Figure 12 illustrates a generalized channel processor for an embodiment of the present invention with ADRO and minimum gain constraint.
  • Figure 1 illustrates a system for sound signal processing.
  • One or more input signals 101 are passed to a channel separator 102.
  • the input signals are usually provided by one or more microphones or by signals transmitted from a remote microphone as in a telephone for exampl Iei or from a signal store as in an MP3 player for example.
  • the channel separator 102 provides meaiia i ⁇ i optionally separating the input signal(s) into parallel channels for further processing by one or more channel processors 103.
  • the channel separator 102 comprises a bank of bandpass filters and each channel 103 processes the output of one bandpass filter.
  • the channel separator 102 splits the signal into one or more slowly varying parts (the background noise) and one or more rapidly varying parts (the desired signals).
  • Each channel processor 103 processes its input signal to produce one or more control signals 105 that are passed to the adaptive processor 106.
  • the adaptive processor 106 processes the input signals 101 to form one or more output signals 106 from the sound processor.
  • the output signals 106 are usually converted to sound or transmitted to a remote location where they are converted to sound by a loudspeaker or similar transducer.
  • the channel processors 103 may optionally receive or send control signals 104 from or to other channel processors 103.
  • Each channel processor 103 in the embodiment illustrated by Figure 1 may be followed by a minimum gain constraint module 108.
  • Figure 2 illustrates another system for sound signal processing.
  • One or more input signals 101 are passed to a channel separator 102.
  • the input signals are usually provided by one or more microphones or by signals transmitted from a remote microphone as in a telephone for example or from a signal store as in an MP3 player for example.
  • the channel separator 102 provides means for optionally separating the input signal(s) into parallel signal channels for further processing by one or more channel processors 203.
  • the channel separator 102 comprises a bank of bandpass filters and each channel 203 processes the output of one bandpass filter.
  • the channel separator 102 splits the signal into one or more slowly varying parts (the background noise) and one or more rapidly varying parts (the desired signals).
  • Each channel processor 203 processes its input signal to produce an output signal that is passed to the channel combiner 205.
  • the channel combiner 205 sums the output signals of the channel processors 203 to form one or more output signals 107 from the sound processor.
  • the output signals 107 are usually converted to sound or transmitted to a remote location where they are converted to sound by a loudspeaker or similar transducer.
  • the channel processors 203 may optionally receive or send control signals 204 from or to other channel processors 203.
  • Each channel processor 203 in the embodiment illustrated by Figure 2 may comprise an example of the present invention with other signal processing means.
  • Figure 7 shows an example architecture for a channel processor 103 or 203 including the present invention and an input-based automatic gain control (AGCi).
  • the input signal 701 to the channel processor comes from the channel separator 102.
  • the amplitude estimator 703 estimates the amplitude of the input signal 701 and passes the amplitude estimate to the AGCi module 704 and the output limiter 706.
  • the AGCi module 704 calculates an initial gain value using an input-gain function as shown, for example in the low-to-mid intensity region of Figure 10.
  • the initial gain value is passed to the minimum gain constraint module 705 which constrains the gain value to be greater or equal to the minimum gain value stored in the minimum gain constraint module.
  • the constrained gain value is passed to the output limiter 706 which constrains the product of the amplitude estimate and the gain value to be less than or equal to a maximum output limit stored in the output limiter 706.
  • the output limit constraint is achieved by reducing the gain value 707 if required.
  • the gain value 707 is passed from the channel processor 103 to the adaptive processor 106 as one of the control signals 105.
  • the gain value 707 multiplies the output of the optional unity gain processing module 702 using multiplier 708 and the result of the multiplication is the output 709 of the channel processor 203 which is passed to the channel combiner 201.
  • Figure 8 shows an example architecture for a channel processor 103 or 203 including the present invention and an output-based automatic gain control (AGCo).
  • the input signal 701 to the channel processor comes from the channel separator 102.
  • the input signal 701 is optionally processed by the unity gain processing unit 702 and the output of the unity processing unit is multiplied by the gain value 707.
  • the output amplitude estimator 803 estimates the amplitude of the output signal 709 and passes the amplitude estimate to the AGCo module 804 and the output limiter 706.
  • the AGCo module 804 calculates an initial gain value using an output-gain function as shown, for example in the low-to-mid intensity region of Figure 1 1.
  • the initial gain value is passed to the minimum gain constraint module 705 which constrains the gain value to be greater or equal to the minimum gain value stored in the minimum gain constraint module.
  • the constrained gain value is passed to the output limiter 706.
  • the output limiter 706 compares the amplitude estimate from the amplitude estimator 803 with the maximum output level stored in the output limiter. If the output amplitude estimate is greater than the output limit, the gain value 707 is reduced. If the output amplitude estimate is less than the output limit, then the gain value 707 is replaced by the lesser of the constrained gain value from the minimum gain constraint module 705 and the gain value 707 multiplied by the ratio of the maximum output limit and the output amplitude estimate.
  • the gain value 707 is passed to the adaptive processor 106 as one of the control signals 105.
  • the gain value 707 multiplies the output of the optional unity gain processing module 702 using multiplier 708 and the result of the multiplication is the output 709 of the channel processor 203 which is passed to the channel combiner 201.
  • the input-output function for the channel processors 103 and 203 described above will resemble the input-output function illustrated by Figure 9.
  • the advantages of these embodiments of the present invention include the similarity of the input-output function of the channel processor with the normal input-output function for normal hearing illustrated by Figure 6; the linearity of the input- output function in the mid-to-high intensity range which provides greater naturalness and sound quality; the capacity to set a suitable value for the minimum gain without requiring a change in the AGCi, AGCo or maximum output limit settings; the capacity to use high compression ratios and faster time constants than would otherwise be possible in the low-to-mid intensity region where the distortion artefacts generated by the compression will be less audible than if compression was applied in the mid-to-high intensity range; and the stapedius reflex for loud sounds and the vocalization- induced stapedius reflex will have their normal effects.
  • Non-linear fitting prescriptions usually prescribe a gain for soft sounds and a gain for loud sounds, or parameters such as kneepoints and compression ratios that allow calculation of gains for arbitrary input levels.
  • the minimum gain value for the invention should be approximately equal to the gain prescribed for loud sounds, and the low-to-mid intensity parts of the input-output function should be as normally prescribed.
  • the compression region should be just above the hearing threshold, and the compression ratio should be higher than is normally prescribed to allow for a reasonably large linear region in the mid-to-high intensity part of the input-output function.
  • the limiting region of the input-output function should be as normally prescribed.
  • the low-to-mid intensity part, the mid-to-high intensity part, and the limiting region of the input-output function are completely specified by parameters that are independent of one another and can be used in fine tuning the hearing aid to correct problems for soft sounds, moderately loud sounds, and very loud sounds respectively without causing new problems in other regions.
  • the compression region In the case of a consumer audio device, the compression region should be placed just above the ambient noise level in order to make soft sounds more audible against the background noise.
  • the minimum gain value should be sufficiently large to achieve an acceptable signal-to-noise ratio for the average level of the signal.
  • a manual volume control or an automatic volume control may be suitable for adapting the minimum gain value for the present invention.
  • Alternative methods may be used to adapt the parameters of the input-output function, one example of which being the technique set out in PCT/AU2004/001691, the content of which is incorporated herein by reference.
  • a further embodiment of the present invention comprises an ADRO channel processor as set out in US Patent No. 6,731,767, the content of which is incorporated herein by reference.
  • Figure 12 illustrates and ADRO channel processor comprising and ADRO processor according to PCT/AU2004/001691 and a minimum gain constraint module 1207 according to the present invention.
  • the ADRO channel processor illustrated by Figure 12 can be used in either an off-line or in-line architecture as illustrated by Figures 1 and 2 respectively.
  • the input signal 1201 to the ADRO channel processor comes from the channel separator 102.
  • the input signal 1201 is multiplied by the gain value 1210 by multiplier 1202.
  • the amplitude estimator 1203 estimates the amplitude of the output signal 1211 and passes the amplitude estimate to the percentile estimator 1204 and the output limiter 1209.
  • the percentile estimator 1204 calculates a high percentile value, for example the 90 th percentile which is the amplitude value that is exceeded 10% of the time, and a low percentile, for example the 30 th percentile which is the amplitude value that is exceeded 70% of the time.
  • the two percentile values are passed to the gain adjuster 1205.
  • the gain adjuster 1205 decrements the interim gain value 1208, otherwise if the low percentile is below the "audibility target vale” stored in the gain adjuster, the gain adjuster increments the interim gain value by a small amount.
  • These steps are called the “comfort rule” and the “audibility rule” of ADRO.
  • the interim gain value is compared with a maximum gain value stored in the maximum gain constraint module 1206. If the interim gain is greater than the maximum gain value, the interim gain is set equal to the maximum gain value. This step is called the "background noise rule" of ADRO.
  • the interim gain value is compared with a minimum gain value stored in the minimum gain constraint module 1207.
  • the interim gain is set equal to the minimum gain value.
  • the interim gain value is passed to the output limiter 1209.
  • the output limiter 1209 compares the amplitude estimate from the amplitude estimator 1203 with the maximum output level stored in the output limiter. If the output amplitude estimate is greater than the output limit, the gain value 1210 is reduced by a factor equal to the maximum output level divided by the amplitude estimate. If the output amplitude estimate is less than the output limit, then the gain value 1210 is replaced by the lesser of the interim gain value 1208 and the gain value 707 multiplied by the ratio of the maximum output limit and the output amplitude estimate. This step is called the "hearing protection rule" of ADRO.
  • the gain value 1210 is passed to the adaptive processor 106 as one of the control signals 105.
  • the output signal 1211 of the channel processor 203 is passed to the channel combiner 201.
  • ADRO is an adaptive linear system with a very slow adaptation rate. This means that ADRO already encompasses most of the advantages that stem from linearity in the mid-to-high intensity range.
  • One advantage of introducing the new ADRO rule is that the minimum gain can be used to prevent ADRO from becoming too soft in the presence of loud noise with a narrow intensity range. This problem can arise for some ADRO hearing aid users under some circumstances.
  • Another advantage is that the minimum gain and the maximum gain concepts are already familiar to audiologists who fit compression hearing aids, as the gain for soft sounds and the gain for moderately loud sounds. By using these parameters, conventional compression fitting prescriptions can also be used for ADRO.
  • the optimal minimum gain value may be larger than 0 dB.
  • Notable examples include hearing aid fittings for people with a conductive or mixed hearing loss.
  • the minimum gain should be set equal to or less than the air-bone gap in the listener's audiogram.

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Acoustics & Sound (AREA)
  • Health & Medical Sciences (AREA)
  • Physics & Mathematics (AREA)
  • Otolaryngology (AREA)
  • Neurosurgery (AREA)
  • General Health & Medical Sciences (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Multimedia (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

Dans un mode de réalisation, un appareil pour traiter le son comprend un moyen pour analyser un signal sonore en un nombre de canaux de signal qui peuvent être classés comme des canaux de signal souhaités et des canaux de signal de bruit et en outre divisés en canaux de signal spécifique de la fréquence; un moyen pour appliquer un gain variable à chaque canal de signal indépendamment; et un moyen pour appliquer un gain minimal à chaque canal de signal de façon à accomplir une amplification linéaire de signaux d'intensité moyenne à élevée dans chaque canal de signal. Les nombreux canaux de signal ajusté en gain sont ensuite combinés afin de générer un seul signal sonore de sortie. L'application du gain minimal de façon différentielle dans chaque canal de signal améliore la qualité sonore pour des sons intenses et améliore le rapport signal sur bruit dans le signal de sortie combiné. L’appareil peut être implémenté dans un mode de réalisation matériel dédié ou par un logiciel fonctionnant sur un microprocesseur.
PCT/AU2009/000870 2008-07-04 2009-07-05 Amplification de gain linéaire pour des sons d’intensité moyenne à élevée dans un processeur de son compressif WO2010000042A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
AU2008903428A AU2008903428A0 (en) 2008-07-04 Linear gain amplification for mid-to-high intensity sounds in a compressive sound processor
AU2008903428 2008-07-04

Publications (1)

Publication Number Publication Date
WO2010000042A1 true WO2010000042A1 (fr) 2010-01-07

Family

ID=41465425

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/AU2009/000870 WO2010000042A1 (fr) 2008-07-04 2009-07-05 Amplification de gain linéaire pour des sons d’intensité moyenne à élevée dans un processeur de son compressif

Country Status (1)

Country Link
WO (1) WO2010000042A1 (fr)

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2454944A1 (fr) 2010-11-18 2012-05-23 Industries FAC, S.L. Moule pour aliments
ITTO20120879A1 (it) * 2012-10-09 2014-04-10 Inst Rundfunktechnik Gmbh Verfahren zum messen des lautstaerkeumfangs eines audiosignals, messeinrichtung zum durchfuehren des verfahrens, verfahren zum regeln bzw. steuern des lautstaerkeumfangs eines audiosignals und regel- bzw. steuereinrichtung zum durchfuehren des regel-
ITTO20121011A1 (it) * 2012-11-20 2014-05-21 Inst Rundfunktechnik Gmbh Verfahren zum messen des lautstaekeumfangs eines audiosignals, messeinrichtung zum durchfuehren des verfahrens, verfahren zum regeln bzw. steuern des lautstaerkeumfangs eines audiosignals und regel- bzw. steuereinrichtung zum durchfuhren des regel- b
WO2014057442A3 (fr) * 2012-10-09 2014-11-27 Institut für Rundfunktechnik GmbH Procédé de mesure de la plage de sonie de signal audio, appareil de mesure pour mise en œuvre dudit procédé, procédé de commande de la plage de sonie de signal audio et appareil de commande pour mise en œuvre dudit procédé de commande

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5832097A (en) * 1995-09-19 1998-11-03 Gennum Corporation Multi-channel synchronous companding system
US5903655A (en) * 1996-10-23 1999-05-11 Telex Communications, Inc. Compression systems for hearing aids
US20030112088A1 (en) * 1999-11-29 2003-06-19 Bizjak Karl L. Compander architecture and methods
US6731767B1 (en) * 1999-02-05 2004-05-04 The University Of Melbourne Adaptive dynamic range of optimization sound processor
WO2006058361A1 (fr) * 2004-12-04 2006-06-08 Dynamic Hearing Pty Ltd Procede et appareil utilisant des parametres de traitement sonore adaptatifs

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5832097A (en) * 1995-09-19 1998-11-03 Gennum Corporation Multi-channel synchronous companding system
US5903655A (en) * 1996-10-23 1999-05-11 Telex Communications, Inc. Compression systems for hearing aids
US6731767B1 (en) * 1999-02-05 2004-05-04 The University Of Melbourne Adaptive dynamic range of optimization sound processor
US20030112088A1 (en) * 1999-11-29 2003-06-19 Bizjak Karl L. Compander architecture and methods
WO2006058361A1 (fr) * 2004-12-04 2006-06-08 Dynamic Hearing Pty Ltd Procede et appareil utilisant des parametres de traitement sonore adaptatifs

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2454944A1 (fr) 2010-11-18 2012-05-23 Industries FAC, S.L. Moule pour aliments
ITTO20120879A1 (it) * 2012-10-09 2014-04-10 Inst Rundfunktechnik Gmbh Verfahren zum messen des lautstaerkeumfangs eines audiosignals, messeinrichtung zum durchfuehren des verfahrens, verfahren zum regeln bzw. steuern des lautstaerkeumfangs eines audiosignals und regel- bzw. steuereinrichtung zum durchfuehren des regel-
WO2014057442A3 (fr) * 2012-10-09 2014-11-27 Institut für Rundfunktechnik GmbH Procédé de mesure de la plage de sonie de signal audio, appareil de mesure pour mise en œuvre dudit procédé, procédé de commande de la plage de sonie de signal audio et appareil de commande pour mise en œuvre dudit procédé de commande
ITTO20121011A1 (it) * 2012-11-20 2014-05-21 Inst Rundfunktechnik Gmbh Verfahren zum messen des lautstaekeumfangs eines audiosignals, messeinrichtung zum durchfuehren des verfahrens, verfahren zum regeln bzw. steuern des lautstaerkeumfangs eines audiosignals und regel- bzw. steuereinrichtung zum durchfuhren des regel- b

Similar Documents

Publication Publication Date Title
AU2008361614B2 (en) Method for sound processing in a hearing aid and a hearing aid
US20080069385A1 (en) Amplifier and Method of Amplification
US10375484B2 (en) Hearing aid having level and frequency-dependent gain
CN101641967A (zh) 用于依赖于声音环境分类缓解耳鸣的声音富集
US9736583B2 (en) Audio processing compression system using level-dependent channels
US20190191253A1 (en) Audio systems, devices, and methods
WO2010000042A1 (fr) Amplification de gain linéaire pour des sons d’intensité moyenne à élevée dans un processeur de son compressif
US9408001B2 (en) Separate inner and outer hair cell loss compensation
US11445307B2 (en) Personal communication device as a hearing aid with real-time interactive user interface
US20210227340A1 (en) Compensating Hidden Hearing Losses by Attenuating High Sound Pressure Levels
US20070019833A1 (en) Hearing device and method for setting an amplification characteristic
US20050175199A1 (en) Method to operate a hearing device and a hearing device
Levitt Compression amplification
Bruce Advances in the Design of Hearing Aids
WO2010015027A1 (fr) Dispositif de traitement du son pour audition fluctuante
Blamey et al. Adaptive dynamic range optimization for hearing aids
de Vries et al. An integrated approach to hearing aid algorithm design for enhancement of audibility, intelligibility and comfort
Flynn Maximizing speech understanding and listening comfort in noise

Legal Events

Date Code Title Description
121 Ep: the epo has been informed by wipo that ep was designated in this application

Ref document number: 09771859

Country of ref document: EP

Kind code of ref document: A1

NENP Non-entry into the national phase

Ref country code: DE

122 Ep: pct application non-entry in european phase

Ref document number: 09771859

Country of ref document: EP

Kind code of ref document: A1