EP2518724A1 - Kombinierte Audioeinheit bestehend aus Mikrofon und Kopfhörer, die Mittel zur Geräuschdämpfung eines nahen Wortsignals umfasst, insbesondere für eine telefonische Freisprechanlage - Google Patents

Kombinierte Audioeinheit bestehend aus Mikrofon und Kopfhörer, die Mittel zur Geräuschdämpfung eines nahen Wortsignals umfasst, insbesondere für eine telefonische Freisprechanlage Download PDF

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Publication number
EP2518724A1
EP2518724A1 EP12164777A EP12164777A EP2518724A1 EP 2518724 A1 EP2518724 A1 EP 2518724A1 EP 12164777 A EP12164777 A EP 12164777A EP 12164777 A EP12164777 A EP 12164777A EP 2518724 A1 EP2518724 A1 EP 2518724A1
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EP
European Patent Office
Prior art keywords
signal
speech
headset
physiological sensor
microphone
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Granted
Application number
EP12164777A
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English (en)
French (fr)
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EP2518724B1 (de
Inventor
Michael Herve
Guillaume Vitte
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Parrot SA
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Parrot SA
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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
    • G10L21/0208—Noise filtering
    • 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
    • H04R1/1083—Reduction of ambient noise
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • 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
    • G10L21/0208—Noise filtering
    • G10L2021/02085—Periodic noise
    • 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
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02165—Two microphones, one receiving mainly the noise signal and the other one mainly the speech signal
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00—Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
    • H04R2460/13—Hearing devices using bone conduction transducers

Definitions

  • the invention relates to a headset type microphone / headset combined.
  • Such a headset can in particular be used for communication functions such as "hands-free" telephony functions, in addition to listening to an audio source (music for example) coming from a device on which the headphones are connected. .
  • one of the difficulties is to ensure sufficient intelligibility of the signal picked up by the microphone (“microphone”), that is to say the speech signal of the close speaker (the helmet wearer).
  • microphone that is to say the speech signal of the close speaker (the helmet wearer).
  • the helmet can indeed be used in a noisy environment (metro, busy street, train, etc.), so that the microphone will not only capture the speech of the wearer of the helmet, but also the surrounding noise.
  • the wearer can be protected from these noises by the helmet, especially if it is a model with closed headphones isolating the ear from the outside, and even more if the headset is provided with an "active control of noise".
  • the distant speaker the one at the other end of the communication channel
  • the noise picked up by the microphone coming to overlap and interfere with the speech signal of the nearby speaker (the helmet wearer).
  • the signal collected by the physiological sensor is usable only in the low frequencies.
  • the noises generally encountered in a usual environment street, metro, train .
  • the physiological sensor delivers a signal naturally devoid of noise component noise (which is not possible with a classic microphone).
  • the EP 0 683 621 A2 for its part, it describes more precisely how to integrate the physiological sensor and the external microphone into one and the same ear canal.
  • the signal collected by the physiological sensor is not strictly speaking speech since speech is not only formed of voiced sounds, it contains components that are not born at the level of the vocal cords: the frequency content is for example much richer with the sound coming from the throat and emitted through the mouth.
  • the internal bone conduction and the crossing of the skin has the effect of filtering certain vocal components, which makes that the signal delivered by the physiological sensor is exploitable only in the lowest part of the spectrum. That is why this signal is supplemented by another signal, delivered by a conventional microphonic sensor, to which it is combined.
  • the general problem of the invention is, in such a context, to deliver to the remote speaker a voice signal representative of the speech transmitted by the near speaker, a signal which is freed from the parasitic components of external noise present in the environment of the close speaker .
  • Another aspect of the invention resides in the ability to efficiently use the signal from the physiological sensor to control various signal processing functions. This signal makes it possible to access new information concerning the content of the speech, which will then be used for the denoising as well as for various auxiliary functions that will be explained below, in particular the calculation of a cutoff frequency of a dynamic filter.
  • the microphone / headset combination comprises low-pass filtering means of the first speech signal before combination by the mixing means, and / or high-pass filtering means of the second speech signal before denoising and combination by the means. mixing.
  • These low-pass and / or high-pass filtering means comprise an adjustable cutoff frequency filter
  • the headset comprises means for calculating the cutoff frequency, operating as a function of the signal delivered by the physiological sensor.
  • the means for calculating the cutoff frequency may in particular comprise means for analyzing the spectral content of the signal delivered by the physiological sensor, able to determine the cutoff frequency as a function of the relative levels of the signal / noise ratio evaluated in a plurality of distinct frequency bands of the signal delivered by the physiological sensor.
  • the denoising means of the second speech signal are non-frequency noise reduction means with, in a particular embodiment of the invention, the microphone assembly which comprises two microphones, and the noise reduction means.
  • non-frequency which comprise a combiner able to apply a delay to the signal delivered by one of the microphones and to subtract this delayed signal from the signal delivered by the other microphone.
  • the two microphones can be aligned in a linear array in a main direction directed towards the mouth of the wearer of the helmet.
  • denoising means of the third speech signal delivered by the mixing means including frequency noise reduction means.
  • input receiving means and operating an intercorrelation between the first and the third speech signal, and outputting a speech presence probability signal. function of the result of the intercorrelation.
  • the denoising means of the third speech signal receive as input this speech presence probability signal for selectively: i) making a noise reduction differentiated according to the frequency bands as a function of the value of the speech presence probability signal, and ii) perform a maximum noise reduction on all the frequency bands in the absence of speech.
  • post-processing means capable of selectively frequency band equalizing in the part of the spectrum corresponding to the signal collected by the physiological sensor. These means determine an equalization gain for each of the frequency bands, this gain being calculated from the respective frequency coefficients of the signals delivered by the microphone (s) and signals delivered by the physiological sensor, considered in the frequency domain. They also operate smoothing on a plurality of successive signal frames of the calculated equalization gain.
  • the reference 10 generally designates the helmet according to the invention, which comprises two atria 12 joined by a hoop.
  • Each of the atria is preferably constituted by a closed shell 12, housing a sound reproduction transducer, applied around the ear of the user with the interposition of an insulating pad 16 isolating the ear from the outside.
  • This helmet is provided with a physiological sensor 18 for collecting the vibrations produced by a voiced signal emitted by the wearer of the helmet, and which can be picked up at the level of the cheek or the temple.
  • the sensor 18 is preferably an accelerometer integrated in the pad 16 so as to be applied against the cheek or the temple of the user with the closest possible coupling.
  • the physiological sensor may in particular be placed on the inside of the skin of the pad so that, once the helmet is in place, the physiological sensor is applied against the cheek or the temple of the user under the effect of a slight pressure resulting from the crushing of the material of the pad, with only the interposition of the skin of the pad.
  • the headset also comprises a microphone array or antenna, for example two omnidirectional microphones 20, 22, placed on the shell of the earpiece 12. These two front and rear mics 22 and 20 are omnidirectional microphones arranged relative to each other. other so that their alignment direction 24 is approximately directed towards the mouth 26 of the helmet wearer.
  • the Figure 2 is a block diagram showing the different blocks and functions implemented by the method of the invention as well as their interactions.
  • the method of the invention is implemented by software means, which can be broken down and schematized by a number of illustrated blocks 30 to 64 Figure 2 . These processes are implemented in the form of appropriate algorithms executed by a microcontroller or a digital signal processor. Although, for the sake of clarity, these various treatments are presented in the form of separate blocks, they implement common elements and correspond in practice to a plurality of functions globally executed by the same software.
  • the reference 28 also designates the sound reproduction transducer placed inside the hull. of the earpiece.
  • These various elements deliver signals that are processed by the block referenced 30, which can be coupled to an interface 32 to the communication circuits (telephone circuits) and receives at the input E the sound intended to be reproduced by the transducer 28 (speech of the remote speaker during a telephone call, music source out periods of telephone communication), and delivers on the output S a signal representative of the speech of the next speaker, that is to say, the wearer of the headset.
  • the signal to be reproduced applied to the input E is a digital signal converted into analog by the converter 34, then amplified by the amplifier 36 for reproduction by the transducer 28.
  • the signal collected by the physiological sensor 18 is a signal mainly comprising components in the lower region of the sound spectrum (typically 0-1500 Hz). As explained above, this signal is naturally non-noisy.
  • the signals collected by the microphones 20, 22 will be used mainly for the high spectrum (above 1500 Hz), but these signals are strongly noisy and it will be essential to carry out a strong denoising treatment to eliminate the components of parasitic noise, the level of which may be such, in certain environments, that they completely obscure the speech signal picked up by these microphones 20, 22.
  • the first stage of the treatment is an anti-echo treatment, applied to the signals of the physiological sensor and the microphones.
  • the sound reproduced by the transducer 28 is captured by the physiological sensor 18 and the microphones 20, 22, generating an echo that disrupts the operation of the system, and must be eliminated upstream.
  • This anti-echo treatment is implemented by the blocks 38, 40 and 42, each of these blocks receiving on a first input the signal emitted by the sensor 18, 20 or else 22 and on a second input the signal reproduced by the transducer. 28 (echo generator signal), and outputs, for further processing, a signal whose echo has been eliminated.
  • the anti-echo treatment is for example carried out by an adaptive algorithm treatment such as that described in FIG. FR 2 792 146 A1 (Parrot SA), which can be referred to for more details.
  • This is an echo cancellation or AEC technique consisting in dynamically defining a compensation filter modeling the acoustic coupling between the transducer 28 and the physiological sensor 18 (or the microphone 20, or the microphone 22, respectively) by a linear transformation between the signal reproduced by the transducer 28 (that is to say the signal E applied at the input of the blocks 38, 40 or 42) and the echo picked up by the physiological sensor 18 (or the microphone 20 or 22).
  • This transformation defines an adaptive filter which is applied to the reproduced incident signal, and the result of this filtering is subtracted from the signal collected by the physiological sensor 18 (or the microphone 20 or 22), which has the effect of canceling the major part acoustic echo.
  • This modeling is based on the search for a correlation between the signal reproduced by the transducer 28 and the signal collected by the physiological sensor 18 (or the microphone 20 or 22), that is to say on an estimate of the impulse response.
  • the coupling constituted by the body of the earphone 12 supporting these various elements.
  • the processing is performed in particular by an adaptive APA ( Affine Projection Algorithm ) algorithm , which provides fast convergence, well suited to hands-free applications with intermittent speech rate and a level that can quickly vary.
  • adaptive APA Affine Projection Algorithm
  • the iterative algorithm is executed with a variable pitch, as described in FIG. FR 2 792 146 A1 supra.
  • the pitch ⁇ varies continuously according to the energy levels of the signal picked up by the microphone, before and after filtering. This step is increased when the energy of the sensed signal is dominated by the energy of the echo, and, conversely, reduced when the energy of the signal picked up is dominated by the energy of the background noise and / or the speech from the remote speaker.
  • the signal collected by the physiological sensor 18 after the anti-echo processing by the block 38 will be used as the input signal of a block 44 for calculating a cutoff frequency FC.
  • the next step consists in filtering the signals, with a low-pass filter 48 for the signal of the physiological sensor 18 and with a filter high pass 50, 52 for the signals collected by the microphones 20, 22, respectively.
  • These filters 48, 50 and 52 are preferably infinite impulse response type IIR (recursive filter) type digital filters, which have a relatively steep transition between the bandwidth and the rejected band.
  • IIR infinite impulse response type
  • these filters are adaptive filters whose cutoff frequency is variable and determined dynamically by the block 44.
  • the cut-off frequency FC which is preferably the same for the low-pass filter 48 and the high-pass filters 50 and 52, is determined from the signal of the physiological sensor 18 after the anti-echo treatment 38.
  • an algorithm calculates the signal-to-noise ratio for a plurality of frequency bands in a range between, for example, 0 and 2500 Hz (the noise level being given by a calculation of the energy in a higher frequency band, for example between 3000 and 4000 Hz, because it is known that in this zone the signal can only be noise, because of the properties of the component constituting the physiological sensor 18).
  • the cutoff frequency chosen will correspond to the maximum frequency for which the signal / noise ratio exceeds a predetermined threshold, for example 10 dB.
  • the following step consists in operating, by means of block 54, a mix to reconstruct the complete spectrum with, on the one hand, the lower region of the spectrum given by the filtered signal of the physiological sensor 18 and, on the other hand, the top of the spectrum given by the filtered signal of the microphones 20 and 22 after passing through a combiner-phase shifter 56 for operating a denoising in this part of the spectrum.
  • This reconstruction is performed by summing the two signals, which are applied synchronously to the mixing block 54 so as to avoid any deformation.
  • the signal that we want to denoise (that is, the signal from the near speaker located in the upper part of the spectrum, typically the components of frequency greater than 1500 Hz) is derived from the two microphones 20, 22 disposed a few centimeters from each other on the shell 14 of one of the earphones of the helmet. As indicated, these two microphones are arranged relative to each other so that the direction 24 they define is approximately oriented in the direction of the mouth 26 of the helmet wearer. As a result, a speech signal emitted from the mouth will reach the microphone before 20 and then the rear microphone 22 with a delay, and therefore a substantially constant phase shift, while the ambient noise will be picked up without phase shift by the two microphones 20 and 22. (which are omnidirectional microphones), given the distance of sources of parasitic noise compared to the two microphones 20 and 22.
  • phase shifter combiner 56 which comprises a phase-shifter 58 applying a delay ⁇ to the signal of the rear microphone 22 and a combiner 60 for subtracting this delayed signal from the signal from the microphone before 20.
  • a differential network of first-order microphones equivalent to a single virtual microphone whose directivity can be adjusted as a function of the value of ⁇ , with 0 ⁇ ⁇ ⁇ ⁇ A ( ⁇ A being the value corresponding to the natural phase difference between the two microphones 20 and 22, equal to the distance between the two microphones divided by the speed of sound, a delay of about 30 microseconds for a spacing of 1 cm).
  • An appropriate choice of this parameter can be achieved by attenuating about 6 dB on surrounding diffuse noises. For more details on this technique, we can for example refer to:
  • This signal is subjected by the block 62 to a frequency noise reduction.
  • this frequency noise reduction is operated differently in the presence or absence of speech, by evaluating a probability p of absence of speech from the signal collected by the physiological sensor 18.
  • this probability of absence of speech is derived from the information given by the physiological sensor.
  • the signal delivered by this sensor has a very good signal / noise ratio up to the cutoff frequency FC determined by the block 44. But beyond this cutoff frequency the signal / noise ratio is still good, and often better than that of the microphones 20 and 22.
  • the sensor information is exploited by calculating (block 64) the frequency intercorrelation between the combined signal delivered by the mixing block 54 and the signal unfiltered physiological sensor, before filtering by the low-pass filter 48.
  • Smix ( f ) and smix ( f ) being the frequency (complex) vector representations, for the n- frame, respectively of the combined signal delivered by the mixing block 54, and of the signal of the physiological sensor 18.
  • the algorithm searches for frequencies for which there is only noise (situation of absence of speech): on the spectrogram of the signal delivered by the mixing block 54 certain harmonics are embedded in the noise, while they stand out more on the signal of the physiological sensor.
  • the peaks P 1 , P 2 , P 3 , P 4 , ... of this intercorrelation calculation indicate a strong correlation between the combined signal delivered by the mixing block 54, and the signal of the physiological sensor 18, and the Emergence of these correlated frequencies indicates the likely presence of speech for these frequencies.
  • the value coefficient_normalization makes it possible to regulate the distribution of the probabilities according to the value of intercorrelation, and to obtain values between 0 and 1.
  • the system that has just been described makes it possible to obtain excellent overall performance, typically of the order of 30 to 40 dB of noise reduction on the speech signal of the nearby speaker.
  • This gives the impression to the distant speaker (the one with which the wearer of the headset is in communication ) that his interlocutor (the helmet wearer) is in a quiet room.
  • the low frequency content collected at the cheek or temple by the physiological sensor 18 is different from the low frequency content of the sound emitted by the mouth of the user, as it would be captured by a microphone located a few centimeters from the mouth, or even by the ear of an interlocutor.
  • the use of the physiological sensor and the filtering described above certainly makes it possible to obtain a very good signal in terms of signal-to-noise ratio, but which may present for the interlocutor who hears it a tone a little deaf and unnatural.
  • the equalization can be performed automatically, from the signal delivered by the microphones 20, 22, before filtering.
  • the Figure 4 shows an example, in the frequency domain (thus after Fourier transform) of the ACC signal produced by the physiological sensor 18, with respect to a MIC microphone signal that would be captured a few centimeters from the mouth.
  • differentiated gains G 1 , G 2 , G 3 , G 4 ,... are applied to different frequency bands of the part of the spectrum located in the low frequencies.
  • the algorithm calculates respective Fourier transforms of the two signals, providing a series of frequency coefficients (expressed in dB) NormPhysioFreq_dB (i) and NormMicFreq_dB (i) respectively corresponding to the standard of the ⁇ th Fourier coefficient physiological sensor signal and the standard of the ⁇ th Fourier coefficient of the microphonic signal.
  • DifferenceFreq_dB i NormPhysioFreq _ d ⁇ B i - NormMicFreq _db i .
  • the gain that will be applied will be less than unity (negative in dB); Conversely, if the difference is negative, the gain to be applied will be greater than unity (positive in dB).
  • Gain_dB i ⁇ .

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Physics & Mathematics (AREA)
  • Health & Medical Sciences (AREA)
  • Quality & Reliability (AREA)
  • Computational Linguistics (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Multimedia (AREA)
  • Otolaryngology (AREA)
  • General Health & Medical Sciences (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Details Of Audible-Bandwidth Transducers (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Telephone Set Structure (AREA)
EP12164777.0A 2011-04-26 2012-04-19 Kombinierte Audioeinheit bestehend aus Mikrofon und Kopfhörer, die Mittel zur Geräuschdämpfung eines nahen Wortsignals umfasst, insbesondere für eine telefonische Freisprechanlage Not-in-force EP2518724B1 (de)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
FR1153572A FR2974655B1 (fr) 2011-04-26 2011-04-26 Combine audio micro/casque comprenant des moyens de debruitage d'un signal de parole proche, notamment pour un systeme de telephonie "mains libres".

Publications (2)

Publication Number Publication Date
EP2518724A1 true EP2518724A1 (de) 2012-10-31
EP2518724B1 EP2518724B1 (de) 2013-10-02

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EP12164777.0A Not-in-force EP2518724B1 (de) 2011-04-26 2012-04-19 Kombinierte Audioeinheit bestehend aus Mikrofon und Kopfhörer, die Mittel zur Geräuschdämpfung eines nahen Wortsignals umfasst, insbesondere für eine telefonische Freisprechanlage

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US (1) US8751224B2 (de)
EP (1) EP2518724B1 (de)
JP (1) JP6017825B2 (de)
CN (1) CN102761643B (de)
FR (1) FR2974655B1 (de)

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EP2945399A1 (de) 2014-05-16 2015-11-18 Parrot Audiokopfhörer mit aktiver anc-geräuschkontrolle mit vorbeugung gegen sättigungseffekte eines feedback-mikrophonsignals
EP3163572A1 (de) * 2015-10-29 2017-05-03 BlackBerry Limited Verfahren und vorrichtung zur unterdrückung von umgebungsgeräuschen in einem sprachsignal, das an einem mikrofon der vorrichtung erzeugt wird
EP3171612A1 (de) 2015-11-19 2017-05-24 Parrot Drones Audio-headset mit aktiver geräuschkontrolle, anti-okklusionskontrolle und löschung der passiven schalldämpfung je nach vorliegen oder nicht-vorliegen einer stimmaktivität des headset-benutzers
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CN102761643B (zh) 2017-04-12
US8751224B2 (en) 2014-06-10
JP6017825B2 (ja) 2016-11-02
FR2974655A1 (fr) 2012-11-02
FR2974655B1 (fr) 2013-12-20
US20120278070A1 (en) 2012-11-01
EP2518724B1 (de) 2013-10-02
JP2012231468A (ja) 2012-11-22

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