WO2010004473A1 - Amélioration audio - Google Patents

Amélioration audio Download PDF

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Publication number
WO2010004473A1
WO2010004473A1 PCT/IB2009/052818 IB2009052818W WO2010004473A1 WO 2010004473 A1 WO2010004473 A1 WO 2010004473A1 IB 2009052818 W IB2009052818 W IB 2009052818W WO 2010004473 A1 WO2010004473 A1 WO 2010004473A1
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WO
WIPO (PCT)
Prior art keywords
audio signal
audio
inter
signal
channel parameters
Prior art date
Application number
PCT/IB2009/052818
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English (en)
Inventor
Sriram Srinivasan
Cornelis P. Janse
Armin G. Kohlrausch
Dirk J. Breebaart
Steven L. J. D. E. Van De Par
Nicolle H. Van Schijndel
Valery S. Kot
Original Assignee
Koninklijke Philips Electronics N.V.
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
Application filed by Koninklijke Philips Electronics N.V. filed Critical Koninklijke Philips Electronics N.V.
Publication of WO2010004473A1 publication Critical patent/WO2010004473A1/fr

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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
    • H04R25/55—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using an external connection, either wireless or wired
    • H04R25/552—Binaural
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/03—Application of parametric coding in stereophonic audio systems

Definitions

  • the invention relates to an audio enhancement method.
  • the invention also relates to an audio enhancement device, a hearing aid, a headset, and a computer program.
  • noise reduction algorithms currently used in hearing aids are not designed to preserve localization cues as discussed in J.G. Desloge, W.M. Rabinowitz, and P.M. Zurek, "Microphone-Array Hearing Aids with Binaural Output - Part I: Fixed- Processing Systems", IEEE Trans. Speech Audio Processing, vol. 5, no. 6, pp. 529-542, Nov. 1997.
  • a first audio signal is extracted from a first input signal
  • a second audio signal is extracted from a second input signal.
  • Inter-channel parameters are extracted from at least a part of the first audio signal and at least a part of the second audio signal.
  • the first audio signal and the second audio signal are processed into processed audio signals.
  • the inter-channel parameters are re-instated into the processed audio signals resulting in a first output audio signal and a second output audio signal.
  • extracting the first audio signal from at least one first input signal and the second audio signal from at least one second input signal comprises beam- forming. This is especially relevant for hearing aids each with multiple microphones. It allows a beam former applied to the multiple microphones on one of the hearing aids to generate a noise-reduced signal. In other words it improves an audio quality.
  • the inter-channel parameters are extracted for a noise component comprised in the first audio signal and the second audio signal.
  • the noise component of the first audio signal and the noise component of the second audio signal are suppressed.
  • imperfections in an audio processing e.g., short filter length, reverberation, imperfect estimates of source statistics
  • there is often a residual noise left after the audio processing e.g., short filter length, reverberation, imperfect estimates of source statistics
  • the residual noise in different channels may no longer have the same inter-channel parameters as before.
  • extracting the first audio signal from at least one first input signal and the second audio signal from at least one second input signal comprises noise cancellation.
  • noise cancellation allows cancelling localized noise sources. This, in situations when speech and noise sources are simultaneously active, improves listener comfort and/or speech intelligibility.
  • the inter-channel parameters comprise one of, e.g., inter-aural differences in time (ITDs), inter-aural differences in level (ILDs), inter-aural coherence and inter-aural correlation.
  • ITDs and ILDs allow to spatially localize a physical sound source. They are often used as the localization cues.
  • Inter-aural coherence and inter- aural correlation describe the perceived width of sound sources in a scene. The auditory scene can be preserved or modified as desired by manipulating these cues.
  • processing the first audio signal and the second audio signal into the processed audio signals comprises one of linear and non- linear processing.
  • Examples of such processing are noise cancellation, feedback cancellation, and loudness compensation, etc.
  • Non-linear processing is often required for loudness compensation. In some cases hearing impaired users can perceive loud sounds well but not soft sounds. In such cases a linear gain is not suited.
  • Non-linear processing can however distort the inter-channel parameters. By extracting inter-channel parameters prior to non-linear processing and reinstating afterwards, the parameters are not affected by the processing.
  • processing the first audio signal and the second audio signal into the processed audio signals comprises a gain control.
  • Automatic gain control is a common feature in many devices, e.g., hearing aids where soft sounds are given a high gain and loud sounds are given a low gain.
  • the gain control can be non-linear and thus can affect the inter-channel parameters. The invention allows preservation of these parameters.
  • processing the first audio signal and the second audio signal into the processed audio signals comprises a noise suppression. In situations when there is a background noise present, this provides improved listener comfort and/or improved intelligibility. In a further embodiment, processing the first audio signal and the second audio signal into the processed audio signals comprises an echo cancellation. In situations where there is background noise, this provides improved listener comfort and/or improved intelligibility, e.g., during communication with a remote talker.
  • processing the first audio signal and the second audio signal into the processed audio signals comprises a dynamic range reduction.
  • Dynamic range reduction is an important feature in hearing aids to compensate for hearing loss, e.g., soft sounds can be amplified without amplifying loud sounds.
  • processing the first audio signal and the second audio signal into the processed audio signals comprises: parametric stereo encoding converting the first audio signal and the second audio signal into a mono down mix, and processing of the mono down mix into a processed audio signal.
  • the processing is thus applied to a single mono down mix derived from the first audio signal and the second audio signal.
  • the original inter-channel parameters are kept as separate parameters and are re-instated after the processing such that the resulting two-channel signal contains the original inter-channel parameters.
  • using the mono down mix lowers the data rate between the right hearing aid and the left hearing aid that otherwise is needed when PCM data, thus non- compressed audio, needs to be exchanged between the two hearing aids.
  • re-instating the inter-channel parameters into the processed audio signal resulting in a first output audio signal and a second output audio signal comprises parametric stereo decoding. This allows proper re-instating of the inter- channel parameters into the first output signal and the second output signal.
  • the audio enhancement method further comprises modifying the inter-channel parameters preceding re-instating the inter-channel parameters into the processed audio signals resulting in the first output audio signal and the second output audio signal.
  • This may allow improved intelligibility, e.g., when speech and noise are simultaneously active.
  • the noise inter-channel parameters may be modified such that the perceived spatial distance between the speech and noise sources is increased, thereby resulting in improved intelligibility.
  • modifying the inter-channel parameters compensates cue changes introduced during processing the first audio signal and the second audio signal into the processed audio signals. This results in preservation of the auditory scene after processing.
  • the processing comprises the noise cancellation
  • the inter-channel parameters corresponding to the noise may be modified. Compensating for these changes preserves the auditory scene.
  • modifying the inter-channel parameters comprises adapting to hearing characteristics of a user. This is especially important for listeners who suffer since a long time from an asymmetric hearing loss. Instead of re-instating the original inter-channel parameters, these inter-channel parameters are adjusted to the hearing characteristics of the two ears (this information available beforehand) such that the inter- channel parameters lead to the localization percept to which these users have been adapted as a consequence of their long-lasting hearing impairment.
  • modifying the inter-channel parameters is controlled by a user. This may allow improved intelligibility when e.g. speech and noise are simultaneously active. The user can then modify noise the inter-channel parameters such that the perceived spatial distance between the speech and noise sources is increased, thereby resulting in improved intelligibility.
  • an audio enhancement device According to another aspect of the invention there is provided an audio enhancement device. It should be appreciated that the features, advantages, comments etc described above are equally applicable to this aspect of the invention.
  • the invention further provides a hearing aid comprising the audio enhancement device according to the invention.
  • the invention further provides a computer program product enabling a programmable device to perform the method according to the invention.
  • Fig. 1 shows a flow chart of an enhancement method according to some embodiments of the invention
  • Fig. 2 shows an example architecture of an audio enhancement device according to some embodiments of the invention
  • Fig. 3 shows an example architecture of the audio enhancement device wherein extracting a first audio signal from at least one first input signal and a second audio signal from at least one second input signal comprises noise cancellation;
  • Fig. 4 shows an example architecture of the audio enhancement device wherein processing the first audio signal and the second audio signal comprises parametric stereo encoding
  • Fig. 5 shows an example architecture of the audio enhancement device according to some of the embodiments in which means for modifying the inter-channel parameters compensates cue changes introduced during processing the first audio signal and the second audio signal into at most two processed audio signals.
  • hearing- support device such as two channels/microphones/speakers, in which a processing is performed independently on different channels of a stereo or multichannel signal.
  • Fig. 1 shows a flow chart of an enhancement method according to some embodiments of the invention.
  • a step 101 comprises receiving at least one first input audio signal from at least one first audio source, and at least one second input audio signal from at least one second audio source.
  • a step 102 comprises extracting a first audio signal from at least one first input signal and a second audio signal from at least one second input signal.
  • a step 103 comprises extracting inter-channel parameters from at least a part of the first audio signal and at least a part of the second audio signal.
  • a step 104 comprises processing the first audio signal and the second audio signal into at most two processed audio signals.
  • a step 105 comprises re-instating the inter-channel parameters into the at most two processed audio signals resulting in a first output audio signal and a second output audio signal.
  • Fig. 2 shows an example architecture of an audio enhancement device 200 according to some embodiments of the invention. In the specific example, the hearing aid comprises the audio enhancement device 200.
  • the audio enhancement device comprises an extraction circuit 210 for receiving at least one first input audio signal from at least one first audio source, and at least one second input audio signal from at least one second audio source.
  • the audio enhancement device 200 comprises three first audio sources 211, 212, and 213, each of them being here a microphone, associated with a first hearing aid. Each of these first audio sources generates respective first input audio signal.
  • the second audio sources 214, 215, and 216 are associated with the second hearing aid, and they generate respective second input audio signals.
  • the number of first audio source and the second audio sources may vary in each of the hearing aids. The minimum number of these audio sources in a hearing aid is one.
  • the three first input audio signals and the three second input audio signals are provided to the extraction circuit 210, which comprises circuitry for performing the steps 101-103 of the audio enhancement method according to the invention. Therefore, the extraction circuit 210 outputs: the first audio signal 201 and the second audio signal 202 that are provided to a processing circuit 230 in which these audio signals 201 and 202 are processed, and the inter-channel parameters 207 that are provided to a re-instating circuit 240 in which these inter-channel parameters are re-instated.
  • the step 102 of the method according to the invention that comprises extracting the first audio signal from at least one first input signal and the second audio signal from at least one second input signal preferably comprises beam-forming. This is beneficial when more than one audio source is used for a hearing aid.
  • beam-forming can be realized. The preferred way of beam-forming is this using a Filtered-Sum Beam-former disclosed in WO99/27522, which in contrast to many other beam forming systems, the FSB system seeks to maximize the sensitivity of the microphone array towards a desired signal rather than to maximize attenuation towards an interferer.
  • the inter-channel parameters comprise one of, e.g., inter-aural differences in time (ITDs), inter-aural differences in level (ILDs), inter-aural coherence and inter-aural correlation.
  • ITDs and ILDs allow to spatially localize a physical sound source. They are often used as the localization cues.
  • Inter-aural coherence and inter-aural correlation describe the perceived width of sound sources in a scene.
  • There are various ways to determine the inter- channel parameters. The preferred way of extracting inter-channel parameters is disclosed in J. Breebaart, S. van de Par, A. Kohlrausch, and E. Schuijers, "Parametric coding of stereo audio", Eurasip J. Applied Signal Proc, 2005, issue 9: special issue on anthropomorphic processing of audio and speech, pp. 1305-1322.
  • the step 104 of the audio enhancement method is performed in the processing circuit 230.
  • the processing circuit 230 processes the first audio signal and the second audio signal into the respective first processed audio signal and the second processed audio signal.
  • the processing performed in processing circuit 230 comprises preferably one of linear and non-linear processing.
  • the examples of such processing are: a gain control, a noise suppression, an echo cancellation, a dynamic range reduction. Important perceptual cues can be changed during such processing.
  • circuits mentioned above and further throughout the description of the invention comprise an electronic circuit or are a suitably programmable processor, in which one suitably programmed processor may be used to implement all circuits shown in the drawings.
  • Fig. 3 shows an example architecture of the audio enhancement device 200 wherein extracting a first audio signal from at least one first input signal and a second audio signal from at least one second input signal comprises noise cancellation.
  • the audio enhancement device depicted in Fig. 3 comprises two paths each associated with the first audio signal 201 and the second audio signal 202.
  • the noise cancellation performed in the extraction circuit 210 comprises separate noise cancellation for each of these audio signals.
  • the noise cancelling part of the extraction circuit 210 for the first audio signal comprises an adaptive beam- former 311, an adaptive filter 321, and a subtraction circuit 331.
  • the noise cancelling part of the extraction circuit 210 for the second audio signal comprises an adaptive beam- former 312, an adaptive filter 322, and a subtraction circuit 332.
  • the operation of the noise canceller present in the extraction circuit 210 is explained on the example of the noise cancelling part corresponding to the first audio signal 201.
  • the part corresponding to the second audio signal 202 operates in a similar way.
  • Fig. 3 depicts three first audio sources providing three first input audio signals to the adaptive beam- former 311.
  • the adaptive beam- former 311 outputs an enhanced first audio output signal based on the at least one first input audio signals. Additionally, the adaptive beam- former 311 outputs a speech- free noise reference signal which is fed into the adaptive filter 321.
  • This adaptive filter might be e.g. a side lobe canceller that calculates a signal corresponding to a localized interfering source (e.g. another speaker). This signal is consequently subtracted in the subtracting circuit 331 from the enhanced first audio output signal resulting in the first audio signal 201.
  • the inter-channel parameters 207 are extracted for a noise component comprised in the first audio signal 201 and the second audio signal 202 in an extraction circuit 361.
  • the extraction circuit 361 uses e.g. the known techniques to extract the inter- channel parameters such as this disclosed in J. Breebaart, S. van de Par, A. Kohlrausch, and E. Schuijers, "Parametric coding of stereo audio", Eurasip J. Applied Signal Proc, 2005, issue 9: special issue on anthropomorphic processing of audio and speech, pp. 1305-1322.
  • the first audio signal 201 and the second audio signal 202 are processed in the processing circuit 230, which e.g. attenuates the remaining noise using spectral processing, resulting in the first processed audio signal 203 and the second processed audio signal 204.
  • the first audio signal 201 and the second audio signal 202 might be processed independently of each other in respective circuits 341 and 342, both being comprised in the processing circuit 230.
  • the processing performed in the processing circuit 230 provides different attenuations for the first audio signal 201 and the second audio signal 202 the inter-channel parameters for speech might be changed.
  • the processing in the circuits 341 and 342 might be synchronized when needed, which is schematically depicted by the communication link between the circuits 341 and 342.
  • the spatial cues of the noise are possibly modified by the adaptive filter 321 and 322. Therefore, to compensate these modifications the original inter-channel parameters
  • circuit 207 extracted for the noise can be modified in a circuit 220 based on the information provided from the adaptive filter 321 and 322, respectively, resulting in the modified inter- channel parameters 208.
  • the communication links between the circuits 321 and 220, and between the circuits 322 and 220 are not shown in Fig. 3.
  • the circuit 220 can be configured to modify the original inter-channel parameters 207 in order to compensate these cue changes based on the information provided from the processing circuit 230.
  • the communication link between the circuits 230 and 220 is not shown in Fig. 3.
  • the re-instating circuit 240 re-instates the modified inter-channel parameters
  • Fig. 4 shows an example architecture of the audio enhancement device 200 wherein processing the first audio signal and the second audio signal comprises parametric stereo encoding.
  • the extraction circuit 210 performs the steps 101 and 102 of the audio enhancement method according to the invention, resulting in the first audio signal 201 and the second audio signal 202.
  • These audio signals 201 and 202 are fed into the processing circuit 230 which comprise a parametric stereo encoder 410 providing a mono down mix 401 and the inter-channel parameters 207 and a circuit 420 processing the mono down mix 401.
  • the parametric stereo encoder 410 converts the first audio signal and the second audio signal into a mono down mix.
  • the parametric encoding extracts spatial parameters with a high spectro-temporal resolution, based on which the optimal mono down mix is generated. This mono down mix preserves all non-spatial aspects of a sound field.
  • the details of the parametric stereo encoding are described in J. Breebaart, S. van de Par, A. Kohlrausch, and E. Schuijers, "Parametric coding of stereo audio", Eurasip J. Applied Signal Proc, 2005, issue 9: special issue on anthropomorphic processing of audio and speech, pp. 1305-1322.
  • the circuit 420 processes the single mono down mix 401, therefore the circuit 420 needs to be present only in one of the two hearing aids.
  • the processing performed in the circuit 420 might comprise an automatic gain control, noise reduction algorithms, or alike audio enhancement algorithms.
  • the single processed audio signal 203 is provided at the output of the circuit 420.
  • the mono down mix signal 401 is analyzed in the circuit 420 in terms of level by a memory less non-linearity, such as e.g. squaring device, followed by low-pass filter.
  • the resulting running signal level estimate is converted to a gain.
  • the gain is typically large for low signal levels and small for high signal levels.
  • the gain is applied to the mono down mix signal using multiplication by the gain to result in the processed audio signal 203 being level- adjusted.
  • the inter-channel parameters 207 are re-instated into the processed audio signal 203 using parametric stereo decoding, which results in a first output audio signal 205 and a second output audio signal 206 with a proper inter-channel parameters.
  • the parametric stereo decoding is described in J. Breebaart, S. van de Par, A. Kohlrausch, and E. Schuijers, "Parametric coding of stereo audio", Eurasip J. Applied Signal Proc, 2005, issue 9: special issue on anthropomorphic processing of audio and speech, pp. 1305-1322.
  • Fig. 5 shows an example architecture of the audio enhancement device 200 according to some of the embodiments in which a circuit 220 for modifying the inter-channel parameters 207 compensates cue changes introduced during processing the first audio signal 201 and the second audio signal 202 into the processed audio signal 203.
  • the circuit 220 is configured to modify the original inter-channel parameters 207 in order to compensate the cue changes based on the information 601 provided from the processing circuit 230.
  • modifying the inter-channel parameters comprises adapting to hearing characteristics of a user.
  • modifying the inter-channel parameters is controlled by a user.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)

Abstract

La présente invention porte sur un dispositif d'amélioration audio qui comprend un circuit d'extraction (210) pour extraire un premier signal audio à partir d'un premier signal d'entrée et un second signal audio à partir d'un second signal d'entrée, et extraire des paramètres inter-canaux à partir d'au moins une partie du premier signal audio et d'au moins une partie du second signal audio, un circuit de traitement pour traiter (230) le premier signal audio et le second signal audio en des signaux audio traités, et un circuit de rétablissement pour rétablir (240) les paramètres inter-canaux dans les signaux audio traités afin d'obtenir un premier signal audio de sortie et un second signal audio de sortie.
PCT/IB2009/052818 2008-07-07 2009-06-30 Amélioration audio WO2010004473A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
EP08159793 2008-07-07
EP08159793.2 2008-07-07

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WO2010004473A1 true WO2010004473A1 (fr) 2010-01-14

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Cited By (5)

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Publication number Priority date Publication date Assignee Title
WO2014058835A1 (fr) * 2012-10-08 2014-04-17 Stc.Unm Système et procédés pour simuler une sortie multisensorielle en temps réel
US9282419B2 (en) 2011-12-15 2016-03-08 Dolby Laboratories Licensing Corporation Audio processing method and audio processing apparatus
US9401152B2 (en) 2012-05-18 2016-07-26 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US10425745B1 (en) 2018-05-17 2019-09-24 Starkey Laboratories, Inc. Adaptive binaural beamforming with preservation of spatial cues in hearing assistance devices
US11708741B2 (en) 2012-05-18 2023-07-25 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders

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Cited By (13)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US9282419B2 (en) 2011-12-15 2016-03-08 Dolby Laboratories Licensing Corporation Audio processing method and audio processing apparatus
US10074379B2 (en) 2012-05-18 2018-09-11 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US9401152B2 (en) 2012-05-18 2016-07-26 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US9721578B2 (en) 2012-05-18 2017-08-01 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US9881629B2 (en) 2012-05-18 2018-01-30 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US10217474B2 (en) 2012-05-18 2019-02-26 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US10388296B2 (en) 2012-05-18 2019-08-20 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US10522163B2 (en) 2012-05-18 2019-12-31 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US10950252B2 (en) 2012-05-18 2021-03-16 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US11708741B2 (en) 2012-05-18 2023-07-25 Dolby Laboratories Licensing Corporation System for maintaining reversible dynamic range control information associated with parametric audio coders
US9898249B2 (en) 2012-10-08 2018-02-20 Stc.Unm System and methods for simulating real-time multisensory output
WO2014058835A1 (fr) * 2012-10-08 2014-04-17 Stc.Unm Système et procédés pour simuler une sortie multisensorielle en temps réel
US10425745B1 (en) 2018-05-17 2019-09-24 Starkey Laboratories, Inc. Adaptive binaural beamforming with preservation of spatial cues in hearing assistance devices

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