EP1841284A1 - Appareil auditif pour l'enregistrement de données audio codées, méthode d'opération et procédé de fabrication du même - Google Patents

Appareil auditif pour l'enregistrement de données audio codées, méthode d'opération et procédé de fabrication du même Download PDF

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Publication number
EP1841284A1
EP1841284A1 EP06405133A EP06405133A EP1841284A1 EP 1841284 A1 EP1841284 A1 EP 1841284A1 EP 06405133 A EP06405133 A EP 06405133A EP 06405133 A EP06405133 A EP 06405133A EP 1841284 A1 EP1841284 A1 EP 1841284A1
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Prior art keywords
audio
signal
hearing instrument
user
coded
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EP06405133A
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German (de)
English (en)
Inventor
Andrea Brueckner
Lukas Florian Erni
Franziska Barbara Pfister
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Sonova Holding AG
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Phonak AG
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Priority to EP06405133A priority Critical patent/EP1841284A1/fr
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    • 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
    • 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
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • 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
    • 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
    • H04R25/70—Adaptation of deaf aid to hearing loss, e.g. initial electronic fitting
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments

Definitions

  • the invention relates to the field of hearing instruments. It relates to a method for operating a hearing instrument having audio feedback capability, a hearing instrument having audio feedback capability, and a method for manufacturing a hearing instrument having audio feedback capability as described in the preamble of the corresponding independent claims.
  • hearing instrument or “hearing device”, as understood here, denotes on the one hand hearing aid devices that are therapeutic devices improving the hearing ability of individuals, primarily according to diagnostic results. Such hearing aid devices may be for instance Outside-The-Ear hearing aid devices or In-The-Ear hearing aid devices or cochlear implants. On the other hand, the term also stands for hearing protection devices and for any other devices which may improve the hearing of individuals with normal hearing, e.g. in specific acoustical situations as in a very noisy environment or in concert halls, or which may even be used in context with remote communication or with audio listening, for instance as provided by headphones.
  • a hearing instrument for example uses a real-time live audio processor for processing a picked-up audio signal and providing the processed signal immediately to the user.
  • the hearing devices as addressed by the present invention are so-called active hearing devices which comprise at the input side at least one acoustical to electrical converter, such as a microphone, at the output side at least one electrical to mechanical converter, such as a loudspeaker, and which further comprise a signal processing unit for processing signals according to the output signals of the acoustical to electrical converter and for generating output signals to the electrical input of the electrical to mechanical output converter.
  • the signal processing circuit may be an analog, digital or hybrid analog-digital circuit, and may be implemented with discrete electronic components, integrated circuits, or a combination of both.
  • a hearing instrument thus is configured to be worn by a user and comprises an input means for picking up an audio signal, a processing unit for amplifying and/or filtering the audio signal, thereby generating a processed audio signal, and an electromechanical converter for converting the processed audio signal and outputting it to the user.
  • These audio signals are the "ordinary” audio signals that are amplified and filtered or otherwise processed, and provided "live" to the user, that is, immediately, without being stored, according to the hearing instrument's purpose of improving the users hearing ability.
  • User feedback in a hearing aid currently consists of a beep or similar acoustic signal delivered to the user via the hearing aid receiver.
  • W0 01/30127 A2 describes a system where the audio feedback in a hearing instrument is user-definable. Different acknowledgement messages can be selected by means of exchangeable memory chips, rewriteable memory, or through communication with an external device. No specific details of storing and playback means are given.
  • EP 0557 847 B1 describes a mechanism for producing user feedback indentifying the program to which a hearing instrument is set. This preferably is done by representing the number of the program by a number of synthetically generated beep signals. As an alternative, "speech generation" is mentioned, but no further description of means for speech generation is given.
  • US 6,839,446 B2 describes a hearing instrument in which an audio signal that has been processed by the hearing instrument can be replayed, typically in response to a user input.
  • the sound signal is stored in an analog "bucket-brigade” circuit, or in a digital storage implementing a circular buffer.
  • the method for operating a hearing instrument having audio feedback capability comprises the steps of
  • the method further comprises the steps of
  • the method further comprises the steps of, in the course of fitting the hearing instrument to a particular user,
  • fitting denotes the process of determining at least one audiological parameter from at least one aural response obtained from a user of the hearing instrument, and programming or configuring the hearing instrument in accordance with or based on said audiological parameter. In this manner, parameters influencing the audio and audiological performance of the hearing instrument are adjusted and thereby tailored or fitted to the end user.
  • the fitting process determines and/or adjusts program parameters embodied in said software, be it in the form of program code instructions, algorithmic parameters or in the form of data processed by the program.
  • the method further comprises the step of, when coding the input audio signal, taking into account a hearing loss characteristic of a user. This adapts the information needed to represent signals according to the user's shifted perception levels in different frequency bands.
  • the storage requirements for the messages can thus be varied in accordance with the hearing loss. Only the information that can actually be perceived by the user is stored.
  • the algorithms for implementing this type of compression including psychoacoustic masking etc. are known, but commonly are implemented with a standard hearing curve as a reference. In the present case, they are implemented with the actual impaired hearing curve of the respective user.
  • the method further comprises the step of, prior to processing the decompressed audio message signal by the processing unit, performing a compensating operation on the decompressed audio message, which compensating operation at least partially compensates for an operation performed by the subsequent processing.
  • the compensation operation is performed prior to compressing and storing the audio message, for a plurality of different compensation operations.
  • the same audio message is stored in different variants, each variant corresponding to one of different operations performed by the subsequent processing, or to other characteristics of the transmission of the audio signal to the user.
  • Different HI programs provide different transfer functions due to different acoustic input conditions. These conditions do not apply for internally generated sound.
  • the compensation operation typically is an equalisation filter, having a frequency dependent gain, in or after the audio message decoder.
  • the variations in subsequent processing may be caused not only by differing hearing programs being selected, but also on differing characteristics affecting the transmission path of the audio message to the user's eardrum, e.g. by differing transfer functions caused by D/A-conversion and/or varying speaker and acoustic coupling characteristics.
  • the acoustic coupling through the ear canal is estimated (given the type of hearing instrument, vent size, etc.) or measured, and the audio messages are compensated or selected accordingly.
  • the method further comprises the steps of
  • the coded audio data is a transformed signal generated by an Extended Lapped Transform (ELT) of an audio message signal, in particular by a Modified Discrete Cosine Transform (MDCT) of an audio message signal, and comprising the step of computing coefficients of the transformed signal by applying said transform to the audio message signal.
  • EHT Extended Lapped Transform
  • MDCT Modified Discrete Cosine Transform
  • a high degree of data compression is achieved by lossy compression, where information is deliberately lost to reduce the amount of data.
  • lossy coders not only try to eliminate redundancy, but also irrelevance. Irrelevance is the part of the information in the signal that is (ideally) not perceptible by the human ear.
  • the quantization process introduces the loss of information. Since only a finite number of bits are available to represent a number with (theoretically) infinite precision, the number is rounded to the nearest quantization level. The error between the quantized value and the actual value is called the quantization error or noise and can be assumed to be a white noise process.
  • Perceptual audio coders such as MP3 attempt to hide the quantization noise under the human perception threshold.
  • a preferred solution presented here does not include such a perceptual shaping of the quantization noise. Instead, it attempts to minimize the overall quantization noise in a mathematical sense. This is not as efficient as a perceptual scheme but is computationally less expensive.
  • audio coders with increased coding efficiency may be used, e.g:
  • the method further comprises the step of, when decoding the coded audio data, extracting side information from the coded audio data, which side information represents normalization factors for the coefficients of the transformed signals. Normalizing the coefficients increases the coding accuracy and/or efficiency when coding the coefficients, but requires that the normalization coefficients be transmitted along with the transform coefficients.
  • the method further comprises the step of, when decoding the coded audio data, decoding the side information by means of a predictor-based coding scheme.
  • a predictor-based coding scheme This implies that the side information was encoded by a predictor based encoder. Coding the side information in this manner further reduces the number of bits to be stored.
  • the method further comprises the step of determining the decoded normalization factors by taking the inverse logarithm of the decoded side information. This implies that not the normalization coefficients themselves were encoded as the side information, but rather a logarithm of the normalization coefficients. It appears that this improves the coding efficiency even more.
  • the method further comprises the steps of
  • This use of a double buffer allows to synchronise the operation of the first processor - typically the main microprocessor or controller of the hearing instrument - with the operation of the second processor - typically a digital signal processor (DSP) that does the actual signal processing.
  • DSP digital signal processor
  • the method further comprises the steps of
  • the method further comprises the step of, prior to outputting an audio message signal, outputting an alert signal for indicating the beginning of an audio message signal.
  • the method further comprises the step of generating a combined audio message signal by concatenating a sequence of separately coded and stored audio message signals.
  • This allows to assemble a message from a sequence of elementary "building blocks", which may be e.g. phrases, words, syllables, triphones, biphones, phonemes.
  • the building blocks are stored, and for each message, the list of building blocks making up the message is stored.
  • the intonation and stress or, in general, prosody parameters of the audio message are modulated.
  • This modulation may take place when recording the message, fitting the hearing instrument, and/or when reconstructing and playing back the audio message.
  • Voice Messages may be modulated either by applying filtering techniques to pre-recorded samples or storing different instances of the same sentence, but spoken differently. Different Messages are preferably given different intonation to enhance the intended meaning. For example, a message alerting the user of low battery may be increasingly stressed if the user ignores it.
  • the speech messages may be adapted to the user's mood.
  • the mood may for example be detected by the frequency of the user switching the controls: Switching the UI controls often in the last few minutes may be interpreted to indicate that the user is irritated. Accordingly, speech messages may be made to sound more soothing. Speech Messages may also be adapted to the current acoustical situation, e.g. quiet or loud surroundings, enhancing certain frequency bands in loud surroundings.
  • the principles for adapting prosody parameters are known in the literature.
  • the audio signals may be spatialized using binaural filtering or standard multichannel techniques. Different messages could be located at different positions, depending on the meaning, or which hearing aid it is coming from. A binaurally spatialized message may be more comfortable and natural to the listener.
  • the decompressed audio signal is output to the user by means of the electromechanical converter of the hearing instrument.
  • the decompressed audio signal is output to the user by means of a converter of a further device, the further device being separate from the hearing instrument, and the method comprising the step of transmitting the decompressed audio signal from the hearing instrument to the further device
  • the hearing instrument having audio feedback capability comprises
  • the hearing instrument comprises a coder for coding an input audio signal picked up by the input means, thereby generating a compressed audio message signal, and for storing the compressed audio message signal as coded audio data in the storage element.
  • the hearing instrument comprises data processing means configured to perform the method steps described above.
  • the data processing means is programmable.
  • the method for manufacturing a hearing instrument having audio feedback capability comprises first the steps of assembling into a compact unit, an input means for picking up an audio signal, a processing unit for amplifying and/or filtering the audio signal, thereby generating a processed audio signal, and an electromechanical converter for converting the processed audio signal and outputting it to the user.
  • the method then comprises the further steps of providing, as elements of the hearing instrument,
  • FIG. 1 schematically shows a structure of a hearing instrument 100.
  • the elements of the hearing instrument 100 are arranged in a housing 101.
  • the housing 101 is shaped to be arranged behind or inside a user's ear.
  • the hearing instrument 100 comprises input means such as a microphone 1 or a telephone coil 1' or a wireless receiver (not shown).
  • Signals from the input means 1, 1' are pre-amplified in analog form and selected by a selector switch 2, converted to a digital representation by an analog to digital converter 3, and processed by a digital signal processor (DSP) 4.
  • DSP digital signal processor
  • signals from different input means 1, 1' are both amplified, combined, and provided to the DSP, or combined by the DSP.
  • the DSP 4, the selector 2 and further elements of the hearing instrument 100 are controlled (dotted lines) by a microprocessor 8.
  • the microprocessor 8 is arranged to retrieve coded audio data from a data store 9 and to forward them to the DSP 4 by means of a double buffer 7.
  • User input may be provided to the microprocessor 8 by means of user controls 102 such as switches or toggle switches, or by wireless remote control (not shown).
  • the processed audio signal generated by the DSP DSP 4 is passed to a digital to analog converter 5, amplified and output to the user by means of a speaker 6. This outputting may alternatively also be implemented in a separate device.
  • FIG. 2 schematically shows a block diagram 10 for decoding an audio message signal.
  • the functionality represented by this block diagram 10 is implemented by the elements of the hearing instrument 100.
  • retrieval block 11 coded audio data is retrieved from the data store 9 and provided as a stream of data blocks (STR) to a processing unit such as the DSP 4 embodying a decoding block or function 12.
  • the decoding function 12 is e.g. realized in a dedicated time slot of the DSP's task allocation schedule.
  • the data stream in demultiplexer block 13 (DEMULT) is separated into data and side information.
  • dequantization block 14 DEQT
  • the decoding step typically involves a look-up table associating codewords with output values and implicitly realizes a nonlinear scaling of the signal.
  • side info dequantization block 15 SDEQT
  • the side information is decoded.
  • the side info dequantization block 15 also performs a decoding of the side information, e.g. by means of a predictive decoder, as explained later on.
  • denormalization block 16 In denormalization block 16 (DENORM), the decoded data is denormalized in accordance with the side information, resulting in transform coefficients representing the audio message signal.
  • IELT inverse transform block 17
  • the time sequence of audio data points is recreated from the transform coefficients. This preferably is done by means of the inverse of the Extended Lapping Transform (ELT) explained in detail further below.
  • ELT Extended Lapping Transform
  • upsampling block 18 the audio signal is upsampled, and in output block 19 (AO) the upsampled audio signal is provided for further processing, typically to the DA converter 5 of the hearing instrument 100 or the external device.
  • FIG. 3 schematically shows a block diagram 20 for coding an audio message signal.
  • the functionality represented by this block diagram 10 is implemented by the elements of the hearing instrument 100, or by a separate data processing unit such as, a personal computer, audiology workstation etc.
  • Audio input block 21 AI
  • Windowing block 22 ELT
  • STDEV standard deviation calculation block 23
  • NVM normalization block 24
  • These standard deviation values constitute the side information.
  • the actual values of the transform coefficients are scaled, in normalization block 24, in accordance with this standard deviation.
  • the scaling is done with the standard deviation values obtained by first quantizing the side information in side info quantization block 25 (SQT) and the dequantizing it again in side info dequantization block 26 (SDEQT). This ensures that the standard deviation values used in normalization block 24 are exactly the same as those used in denormalization block 16 when decoding.
  • the side info quantization block 25 also performs a coding of the side information, e.g. by means of a predictive encoder, as explained later on.
  • quantization block 27 the normalized coefficients are quantized. This quantization step ultimately causes the data compression.
  • MULT multiplexing block 28
  • the quantized coefficients are interleaved with the side info, generating a data stream (STR) output in block 29 to a storage or a transmission channel.
  • the main functional block of the system is the ELT which implements the time-frequency transform.
  • the purpose of the transform is to decorrelate the samples in the signal.
  • the decorrelated samples will have a lower variance than the original samples and can therefore be encoded with less bits for the same signal to noise ratio (SNR). This reduction is called the coding gain and will be discussed in more detail further below.
  • SNR signal to noise ratio
  • the coder described here uses principles taken from Audio Coding schemes often referred to as Transform Coders. These include the popular MP3, AAC or ATRAC Audio Coders. Unlike advanced coding schemes mentioned, the ELT as presented here does not use perceptual models for quantization noise masking, as these are costly to implement on hardware currently available.
  • the audio message coder and encoder run on a sampling frequency of 10 kHz.
  • the output is then upsampled to the sample frequency of 20 kHz as used in the remaining hearing instrument 100.
  • Figure 4 schematically shows a format of the coded audio data generated by multiplexer 28 and disassembled by demultiplexer 13.
  • a stored or transmitted data stream consists of a sequence of frames 30, each frame comprising one block of side info 31 and a sequence 32 of e.g. eight data blocks 33, 33', 33".
  • the coded data is stored in a non-volatile memory data store 9 of the hearing instrument 100 and transferred to the DSP 4 by the microprocessor 8 or controller.
  • a suitable mechanism for passing the data to the DSP 4 is required. As mentioned in the context of Figure 1, this data passing is achieved by means of a double buffer 7.
  • Figure 5 schematically shows a communication flow when retrieving coded audio data and passing it to the DSP 4 through the double buffer 7. Since there is no common clock, operations are synchronized by the DSP 4 sending an interrupt request IRQ to the microprocessor 8, denoted as ⁇ P.
  • the routine associated with the interrupt request IRQ has sufficient priority to fetch the next block of coded data (step 51, GET) and write it (step 52, WR 1) to a first buffer B1 of the double buffer 7 in the course of a common cycle time of e.g. 25 ms.
  • the DSP 4 reads the coded data previously stored in the second buffer B2 (step 53, RD 2), decodes it (step 54, PROC), merges it with the ordinary audio signal and passes the merged signal to the DA converter 5 (step 55, OUTP). Then the DSP 4 issues a further IRQ, causing the microprocessor 8 to fetch the next block of coded data and write it to the second buffer B2 (step 56, WR 2), while the DSP 4 reads from the first buffer B1 (step 57, RD 1).
  • This double buffering mechanism is implemented in separate threads or time frames, once for retrieving the data blocks 32, 32', 32" and once (used less often) for retrieving the side info blocks 31.
  • the Extended Lapping Transform (ELT) as mentioned previously serves to reduce the correlation between samples.
  • ELT Extended Lapping Transform
  • the basic principles are commonly known, the following is a summary of the forward transform.
  • the inverse transform is analogous to the forward transform.
  • the ELT decomposes the signal into a set of basis functions.
  • the resulting transform coefficients have a lower variance than the original samples.
  • ⁇ 2 f is the variance of the transform coefficients and ⁇ 2 t the variance of the time-domain samples.
  • DCT Discrete Cosine Transform
  • n is the block length and i is the coefficient index.
  • the DCT can be applied blockwise to a signal with a rectangular window and reconstruction can be achieved by the inverse transform.
  • the rectangular window however introduces blocking artefacts which are audible in the reconstructed signal. By using an overlapping window these artefacts can be reduced and the coding gain increased.
  • the ELT is therefore usually used in signal compression applications.
  • This transform can be implemented through the DCT and uses an overlapping transform window while maintaining critical sampling. Increasing the transform length with an overlapping window would normally result in an oversampling of the signal which is clearly undesirable in data compression.
  • the performance of the transform can be further increased by using a window that tapers to zero towards the edges.
  • the parameter ⁇ is between 0 and 1 and is set to 0.5 in this case.
  • the length n of the transform is 32 to allow the use of a particular FFT Coprocessor to calculate the transform.
  • N is 128 and so is M.
  • Figure 6 schematically shows a predictor-based coder implemented as part of the side info quantization block 25.
  • the logarithm base 2 64 of the standard deviation is taken and a prediction algorithm is applied.
  • the predictor decorrelates samples in a sequence, thereby reducing the variance.
  • the scheme used here is a simple first-order closed-loop predictor comprising an adder 64, a time delay 65 and a gain 66 corresponding to the prediction coefficient. Its output is subtracted from the input signal x(n) by a difference operator 62 and the difference is quantized by quantizer 63.
  • the output of the quantizer 63 is input to the adder 64 of the predictor. Equation 11 shows the optimal result of the prediction algorithm.
  • ⁇ y 2 1 - ⁇ 2 ⁇ ⁇ x 2
  • ⁇ 2 y is the variance of the output
  • ⁇ 2 x the variance of the input
  • ⁇ the prediction coefficient in this case 0.98.
  • Figure 7 schematically shows the corresponding predictor-based decoder implemented as part of the dequantization block 15. It comprises the inverse predictor with adder 67, delay 68 and gain 69, which is the same as in the encoder. The prediction is performed with the signal after it has been quantized and dequantized again. This ensures that the value at the output of the inverse quantizer is the same in encoder and decoder, as is shown in equation 12.
  • e n x n - a x ⁇ ⁇ n - 1
  • the values x ⁇ ( n ) at the output of the predictor have a probability density function that approaches a Gaussian distribution, i.e. they approach a white noise sequence.
  • the side information can therefore be quantized with Gaussian quantizers.
  • the combination of log function and prediction allows the side information to be transmitted with 3 bits only, leaving more bandwidth for the Data.
  • Figure 8 schematically shows a block diagram conceptually illustrating, in terms of signal flow, the compensation of at least part of the subsequent processing.
  • the ordinary audio signal flow path passes from input device 1, 1' over selector 2 and A/D-Converter 3 into the main processing block 84.
  • the processed signal to be output is fed from the main processing block 84 to the D/A-Converter 5, an amplifier and to the speaker 6.
  • the main processing block may be regarded as comprising a first processing operation 85 and a second processing operation 86 (where "first" and "second” do not necessarily imply a particular sequence of these operations).
  • the first processing operation 85 (F) typically is a generic processing operation corresponding to the hearing program chosen.
  • the second processing operation 86 (G) typically is a user specific adaptation and usually is much more complex than the first processing operation.
  • the audio message signal retrieved from the store 9, after decoding in decoding block 12, is passed through an inverse function block 87 and added to the main signal flow path by adder 88 before the main processing block 84.
  • the inverse function block 87 implements at least approximately the inverse (F -1 ) of the first processing operation 85 (F) in order to reduce or minimize the effect of the first processing operation 85 on the audio message signal.
  • the function of the inverse function block 87 is changed in accordance with the hearing program functions embodied in the first processing operation 85.
  • the inverse function block 87 is in reality implemented on the DSP 4 under control of the microprocessor 8 as are the other processing functions.

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  • Engineering & Computer Science (AREA)
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  • Acoustics & Sound (AREA)
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  • Physics & Mathematics (AREA)
  • Otolaryngology (AREA)
  • Neurosurgery (AREA)
  • General Health & Medical Sciences (AREA)
  • Computational Linguistics (AREA)
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  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
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Cited By (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE102007014132A1 (de) * 2007-03-23 2008-09-25 Siemens Audiologische Technik Gmbh Prozessorsystem mit direkt verschalteten Ports
DE102008063207A1 (de) 2008-12-29 2010-07-08 Siemens Medical Instruments Pte. Ltd. Verfahren zum Betreiben einer Hörvorrichtung mit Sprachsynthese und Hörvorrichtung
WO2010094335A1 (fr) * 2009-02-20 2010-08-26 Widex A/S Système d'enregistrement de message sonore pour prothèse auditive
GB2470279A (en) * 2009-05-12 2010-11-17 Sonitus Medical Inc Mouth wearable digital audio player
EP2082779A3 (fr) * 2008-01-22 2014-02-19 Cochlear Limited Adaptation contrôlée par le porteur d'une prothèse auditive
CN107180639A (zh) * 2013-04-29 2017-09-19 杜比国际公司 对更高阶高保真度立体声响复制表示进行压缩和解压缩的方法和装置
US10412512B2 (en) 2006-05-30 2019-09-10 Soundmed, Llc Methods and apparatus for processing audio signals
US10484805B2 (en) 2009-10-02 2019-11-19 Soundmed, Llc Intraoral appliance for sound transmission via bone conduction

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US10735874B2 (en) 2006-05-30 2020-08-04 Soundmed, Llc Methods and apparatus for processing audio signals
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US10477330B2 (en) 2006-05-30 2019-11-12 Soundmed, Llc Methods and apparatus for transmitting vibrations
US10412512B2 (en) 2006-05-30 2019-09-10 Soundmed, Llc Methods and apparatus for processing audio signals
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US8660278B2 (en) 2007-08-27 2014-02-25 Sonitus Medical, Inc. Headset systems and methods
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WO2010094335A1 (fr) * 2009-02-20 2010-08-26 Widex A/S Système d'enregistrement de message sonore pour prothèse auditive
GB2470279A (en) * 2009-05-12 2010-11-17 Sonitus Medical Inc Mouth wearable digital audio player
US10484805B2 (en) 2009-10-02 2019-11-19 Soundmed, Llc Intraoral appliance for sound transmission via bone conduction
CN107180639A (zh) * 2013-04-29 2017-09-19 杜比国际公司 对更高阶高保真度立体声响复制表示进行压缩和解压缩的方法和装置
CN107180639B (zh) * 2013-04-29 2021-01-05 杜比国际公司 对更高阶高保真度立体声响复制表示进行压缩和解压缩的方法和装置

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