EP2101517B1 - Processeur audio pour la conversion d'un signal mono en un signal stéréo - Google Patents

Processeur audio pour la conversion d'un signal mono en un signal stéréo Download PDF

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EP2101517B1
EP2101517B1 EP08102617A EP08102617A EP2101517B1 EP 2101517 B1 EP2101517 B1 EP 2101517B1 EP 08102617 A EP08102617 A EP 08102617A EP 08102617 A EP08102617 A EP 08102617A EP 2101517 B1 EP2101517 B1 EP 2101517B1
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input signal
audio
audio input
delayed
delay
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EP2101517A1 (fr
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Pauli Jacobus MINNAAR
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AM3D AS
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AM3D AS
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/002—Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution
    • H04S1/005—For headphones

Definitions

  • the present invention relates to the field of signal processing, especially audio signal processing. More specifically, the invention provides an audio processor capable of converting an audio input signal (mono) to two audio output signals (stereo).
  • the sound image appears to be inside (in the middle) of the head. This is referred to as in-the-head-localization. This is undesirable and is generally experienced to be unpleasant and unnatural since it does not occur in real-life listening where the human hearing will normally always is able to appoint a position in space to a sound source. Furthermore, it can lead to listener fatigue when exposed for longer periods of time. It is therefore desirable to enhance this impoverished sound image in such a way that it is more natural and pleasant to listen to.
  • the enhancement should be applicable not only to a single voice or musical instrument, but to the final mix of signals.
  • the processing should be done without using excessive signal processing power, since implementations typically have to be portable.
  • US 6,084,970 describes a method and a device for converting a monophonic signal into a stereo signal by selectively allocating frequency bands of the input signal to left or right outputs. In this way, some frequency components will be present only in the left output and others only in the right output.
  • This method can be used successfully for processing a single audio track when mixing it into a final mix, especially if the mix already contains some stereo content. This signal should then be played through loudspeakers. However, when presented through headphones, this can lead to an unpleasant sound - especially when this effect is applied to a final mix. Furthermore, the method is not able to provide out-of-head localization. Another disadvantage of the method is that a reasonably large number of filters are needed which leads to an unacceptably high signal processing requirement.
  • a way to alleviate the problem of in-the-head-localization is by means of binaural synthesis, such as described in J. Blauert, "Spatial hearing: The psychophysics of human sound localization", MIT Press, Cambridge, USA, Revised edition, 1997 .
  • HRTFs head-related transfer functions
  • a sound can be processed such that it appears to be placed at some location outside the head. It is however, well known that placing sound sources convincingly in front of the listener is exceedingly difficult. The location of the sound source is often confused with other positions in the median plane and very often in-the-head-localization occurs. It is especially difficult to simulate distance when using HRTFs alone, and furthermore, the sound colour, i.e. the timbre, of the signal can be affected adversely.
  • EP 1 365 629 A shows an audio processor according to the preamble of claim 1 and a method for converting a mono input signal according to the preamble of claim 23.
  • US 2002/015505 A1 shows an apparatus and a method using delays to improve sound recordings.
  • EP 1 251 717 A shows the use of short time delays in the order of 0.1 -0.7 ms to improve localisation of sound produced by a headphone.
  • US 5 173 944 A shows a simple delay circuit to process a mono signal into a pseudo-stereo signal using only four different delay stages with delay times in the order of 15 - 45 ms and two summing stages.
  • the invention provides an audio processor arranged to receive a mono audio input signal and generate first and second audio output signals in response thereto, the processor being arranged to - generate a first delayed version of the audio input signal being delayed by a first delay in relation to the audio input signal,
  • Such an audio processor is capable of taking a single channel audio input signal and provide a stereo output signal in response, i.e. mono-to-stereo conversion and thus provides what could be called "mono-widening".
  • the stereo output signal is suited for stereo headphone listening since it provides the listener with an out-of-head localization without severely changing the timbre of the single channel input signal.
  • the audio processor can be seen as providing a widening effect by simulating a very simple virtual sound source in front of the listener with a single reflection on each side of the listener, each of these reflections being provided with an ipsi-lateral and a contra-lateral contribution. With very little processing power required, the audio processor according to the first aspect is capable of providing an out-of-head localization without suffering from severe coloration, when listened to over headphones.
  • the mono-to-stereo conversion is obtained with very simple processing means in the form of delays and summations and optionally first order filters. Thus, it can be implemented in miniature portable devices with very limited processing capacity, such as mobile phones, hearing aids etc.
  • the audio processor according to the first aspect provides a stereo output signal which is based on the unprocessed input signal, and this helps to ensure that the stereo output signal will have the general timbre in common with the input signal, whereas the delayed versions of the input signal serve to simulate simple acoustic reflections which help to provide an out-of-head localization.
  • the first audio output signal consists only of a sum of the audio input signal and the first and fourth delayed versions of the audio input signal
  • the second audio output signal consists only of a sum of the audio input signal and the second and third delayed versions of the audio input signal, such as defined above.
  • This embodiment is very simple and can be implemented with very limited processing required, and still with a natural timbre and out-of-head localization. However, an improved sound quality can be obtained e.g. by adding more delayed versions of the audio input signal to each of the two output signals, providing appropriate filtering and attenuations, as will be explained in the following.
  • the first delay range is preferably 30 ms to 80 ms, such as 40 ms to 70 ms, such as 50 ms to 60 ms. Such first delay ranges are found suitable for simulating simplified lateral reflections that result in out-of-head localization.
  • the second delay range is preferably 50 ⁇ s to 800 ⁇ s, 200 ⁇ s to 700 ⁇ s, such as 450 ⁇ s to 650 ⁇ s. Such second delay ranges are found suitable for supporting simplified contra-lateral versions of the reflections, since the second delay range is thus within inter-aural time differences experienced in real-life listening.
  • the first and third delays are preferably selected such that a difference between the first delay and the third delay is within a third relay range of 1 ms to 30 ms, such as 3 ms to 15 ms, such as 5 ms to 10 ms.
  • Such delay difference is found suitable for simulating the effect of asymmetric reflections. Compared with identical first and third delays, this delay difference supports out-of-head localization.
  • the audio processor may include a low-pass filter section arranged to
  • Such a low-pass filter section may further be arranged to
  • Low-pass filtering of the delayed versions of the audio input signal before adding these versions to the audio input signal generally provides a more natural sounding output signal, since the timbre at high frequencies is less influenced by the delayed versions of the audio input signal.
  • the low-pass filter section may be arranged to provide a cut-off frequency within the range 300 Hz to 5 kHz, such as within the range 400 Hz to 3 kHz, such as 500 Hz to 1 kHz.
  • the audio processor may include a band-pass filter section arranged to band-pass filter all of the first, second, third and fourth delayed versions of the audio input signal before being summed to form the first and second audio output signals.
  • a band-pass filter section arranged to band-pass filter all of the first, second, third and fourth delayed versions of the audio input signal before being summed to form the first and second audio output signals.
  • the band-pass filter section may be arranged to provide a band-pass frequency range of 100 Hz to 5 kHz, such as 300 Hz to 3 kHz, such as 500 Hz to 1 kHz.
  • All filters, low-pass filters or band-pass filters may be implemented with first order filter sections, which are easy to implement and still provides the intended effect without the complexity required by higher order filters.
  • the first sum may include a fifth delayed version of the audio input signal, and wherein the second sum includes a sixth delayed version of the audio input signal.
  • further delayed versions may be added, also more than the mentioned fifth and sixth delayed versions.
  • the first and second sum may further include a plurality of delayed versions of the audio input signal. Each of this plurality of delayed versions of the audio input signal is preferably provided with different delays so as to simulate multiple reflections. Such higher number of delayed versions of the audio input signal can be used for further increasing sound quality, but at the price of required processing power.
  • the audio processor may include an attenuation section arranged to attenuate the first, second, third and fourth delayed versions in relation to the audio input signal before generating the first and second sums.
  • the delayed versions are attenuated so as to further increase the influence of the audio input signal itself in the output signals, thus resulting in a more natural timbre.
  • the first, second, third and fourth delayed versions may be attenuated by at least 2 dB in relation to the audio input signal, such as by at least 4 dB, such as by at least 6 dB, such as by at least 10 dB.
  • the audio processor may be arranged to attenuate the audio input signal prior to generating the delayed versions of the audio input signal. Especially such attenuation may be used to provide an unchanged loudness when switching from unprocessed (i.e. providing the audio input signal in both output channels) to processed (outputting the first and second audio output signals). Such constant loudness may typically be obtained by an attenuating the audio input signal by 2-3 dB prior to generating the delayed versions of the audio input signal.
  • a second aspect of the invention provides a device including an audio processor according to the first aspect.
  • the audio processor can advantageously be built into portable devices for converting mono audio signals to stereo audio signals which are more pleasant for stereo playback.
  • a non-exhaustive list of such types of devices is: mobile phones, portable computers, headphones, headsets, assistive listening devices, hearing aids and a set of loudspeakers arranged for positioning close to a listener's ears.
  • the device may also be in the form of a stand-alone mono-to-stereo converter device arranged for wired or wireless receipt of the (mono) audio input signal and for wired or wireless output of the (stereo) first and second audio output signals.
  • a third aspect of the invention provides a system including
  • the first and second electro-acoustic transducers may be included in one of: a set of headphones, a headset, a set of hearing aids, a set of hearing assistive devices, a set of loudspeakers arranged for positioning close to a listener's ears.
  • the system may include a set of left and right hearing aid devices arranged for position in respective left and right ears of a user.
  • the invention is advantageous even for hearing aids which typically has very limited processing power available due to the small space combined with requirements for extremely low power consumption.
  • a mono signal from e.g. a CD player, an MP3 player or the like, it is possible to provide a pleasant sound reproduction even though the available bandwidth only allows transmission of a mono signal to the hearing aids.
  • the audio processor is included in a separate unit, wherein the separate unit is arranged to receive the audio input signal from an external device, and to provide the first and second audio output signals to the respective left and right hearing aid devices.
  • the mono-to-stereo conversion is performed by the separate unit and not in the hearing aid devices, thus saving processing power in the hearing aid devices.
  • the separate unit may include a wireless transmitter arranged to transmit signals representing the first and second audio output signals to respective receivers in the left and right hearing aid devices.
  • the separate unit has a wired connection to the left and right hearing aid devices.
  • Another such hearing aid embodiment includes a separate unit arranged to receive the audio input signal from an external device, and to provide this audio input signal to both of the left and right hearing aid devices, wherein the left hearing aid device includes a processor arranged to generate the first audio output signal as a first sum of the audio input signal and first and second delayed versions of the audio input signal, and the right hearing aid device includes a processor arranged to generate the second audio output signal as a second sum of the audio input signal and third and fourth delayed versions of the audio input signal.
  • the audio processor according to the invention is suited for splitting into two separate parts which can be implemented in respective left and right hearing aid devices.
  • the separate unit may include a wireless transmitter arranged to transmit a signal representing the audio input signal to respective receivers in the left and right hearing aid devices. Alternatively, the separate unit has a wired connection to the left and right hearing aid devices.
  • the invention provides a method for converting a single mono audio input signal (X) to a set of first and second audio output signals, the method including
  • the second and fourth delays are within a second delay range of 50 ⁇ s to 1 ms.
  • the invention provides computer-executable program code arranged to perform the method according to the fourth aspect.
  • the program code may be dedicated program code for a specific signal processor, or program code arranged for a general purpose computer, e.g. a Personal Computer.
  • the invention provides a data carrier including a computer executable program code according to the fifth aspect.
  • the data carrier may be such as any type of disk, memory card, memory stick, hard disk etc.
  • Fig. 1 shows a signal diagram with basic elements of a simple mono-to-stereo algorithm embodiment for use in a mono-to-stereo audio processor.
  • Four delays d1, d2, d3, d4 are used, namely delays corresponding to the respective first, second, third and fourth delays as defined in the preceding chapter.
  • the respective first and second sums are each seen to be split into two summation points in this specific embodiment.
  • the mono audio input signal X is directly applied to left L and right R output signals via summation two points, and thus the unprocessed audio input signal X forms a vital part of both of the left L and right R output signals.
  • An optional band-pass filter BPF serves to provide a band-pass filtered version of the audio input signal X.
  • This band-pass filtered version of the input signal X is then used as input for providing band-pass filtered delayed versions S1, S2, S3, S4 of the input signal X.
  • the band-pass filter BPF can in principle be omitted completely, or replaced by only a low-pass filter or a high-pass filter. However, with the band-pass filter BPF included, it is ensured that delayed versions of the lowest and highest audio frequencies are not passed to the output signals L, R. Hereby, the widening effect provided by these delayed versions of the input signal X only causes a minimal influence on the overall perceived timbre when compared to the timbre of the unprocessed input signal X.
  • the left output signal L is a sum of 1) the input signal X, 2) signal S1 being a delayed version of the input signal X, delayed by delay d1, and 3) signal S4 also being a delayed version of the input signal X, delayed by delay d3 and further delayed by delay d4.
  • the right output signal R is sum of 1) the input signal X, 2) signal S3 being a delayed version of the input signal X, delayed by delay d3, and 3) signal S2 also being a delayed version of the input signal X, namely S1 delayed by delay d1 and further delayed by delay d2.
  • the signal diagram illustrated in Fig. 1 is simple and thus suited for implementation on processors with limited signal processing capabilities. If the required memory is available, the delays d1, d2, d3, d4 can preferably be implemented purely as simple sample buffers which would further help to save processing power compared to alternative delay implementations.
  • d1 and d3 are chosen to have a difference of 10 ms which is found suitable to provide a widening effect and thus avoid in-head localization.
  • Delays d2 and d4 may likewise be chosen slightly different, but this is not essential.
  • More delayed version of the input signal X can be added to form the output signals L, R, but a preferred embodiment only consists of two delayed versions of the input signal X per output signal L, R, such as illustrated in Fig. 1 , thereby providing a very simple algorithm.
  • Fig. 2 illustrates a signal diagram of another simple embodiment serving to illustrate the core elements of the invention.
  • two summation points are used for providing the left and right outputs L, R based on the input signal X and delayed versions S1, S2, S3, S4 thereof.
  • Delays d1 and d3 for providing delayed versions S1, S3 of the input signal X are similar to those of Fig. 1 .
  • delays d12, d14 for providing respective delayed versions S2, S4 of the input signal X are different from delays d2, d4 of Fig. 1 .
  • input signal X is used as input to delays d12, d14.
  • d12 should be selected as the sum of d1 and d2 in Fig.
  • d14 should be selected as the sum of d3 and d4 to provide similar output signals L, R.
  • an optional band-pass filter BPF is included so that the delayed versions S1, S2, S3, S4 are band-pass filtered versions of the input signal X.
  • a simple way of reducing the timbre impact of these signals S1, S2, S3, S4 on the output signals L, R, is to gain down these signals S1, S2, S3, S4 in relation to the input signal X, before being added to the input signal X, e.g. by providing a gain of -3 dB, -6 dB or -10 dB, as will be illustrated in the following.
  • Fig. 3 illustrates a signal diagram of another embodiment where the delayed versions of the input signal X are provided with a different layout of delays D1, D2, D3, D4 to provide a set of output signals L, R in response to the input signal X.
  • these delays D1, D2, D3, D4 are different from delays d1, d2, d3, d4 of Fig. 1 , and thus should be selected differently to provide the same result.
  • a first gain G1, e.g. -2 dB is included to attenuate the input signal X prior to being used for the further processing.
  • a second gain G2, e.g. of -6 dB, and a band-pass filter BPF is included to generally attenuate and band-pass filter all delayed versions of the input signal X.
  • Fig. 4 illustrates a signal diagram of an audio processor embodiment split into two separate parts, namely one taking the input signal X and converts it to a first output signal L, and one taking the input signal X and converts it to a second output signal R.
  • this embodiment is suited e.g. for hearing aids where each of the separate parts can be implemented into respective left and right hearing aid devices.
  • delays D5, D6, D7, D8 are different from the delays in the earlier Figures, and thus should be selected so as to obey what is generally described for Fig. 1 .
  • Respective gains and band-pass filters G3, BPF and G5, BPF are included so that all delayed versions of the input signal X are gained and band-pass filtered prior to being summed.
  • gains G4, G6 are included to gain the respective summed signals prior to being output as respective first and second output signals L, R.
  • Fig. 5 illustrates signal diagram of a more complex embodiment.
  • Delays D9, D10, D11, D12 are used for generating delayed versions of the input signal X, as explained earlier, prior to being summed to form respective output signals L, R.
  • gains G9, G10, G11, G12 are included to allow an individual attenuation of the contributions to the output signals L, R. This allows the choice of a higher attenuation for the contra-lateral contributions compared to the ipsi-lateral contributions, which is considered as advantageous, e.g. G9 and G11 can be chosen as -6dB, whereas G10 and G12 can be chosen as -9dB.
  • G7 and G8 can be chosen as -2dB.
  • the input signal X is filtered by filters Filter1, Filter4 prior to being fed to the summation points where it is summed with delayed versions of the input signal X.
  • the delayed versions of the input signal X are based on filtered versions of the input signal X filtered by filters Filter2, Filter3.
  • All of Filter1, Filter2, Filter3, Filter4 can be chosen to be different or be chosen to be similar, and they can be chosen to be low-pass, high-pass or band-pass filters or more complex filters.
  • the filters Filter1, Filter2, Filter3, Filter4 can be chosen to be HRTFs representing a desired 3D direction, thus giving the listener a more precise experience of localizing a virtual sound source.
  • the filters Filter1 and Filter4 may implement the HRTFs in front of the listener, while the filter Filter2 implements an ipsi-lateral HRTF for a direction on the side of the head, whereas the filter Filter3 implements the corresponding contra-lateral HRTF for the same direction on the side of the head.
  • Fig. 6 illustrates a hearing aid system in schematic.
  • Hearing aid devices HL, HR are suited for a listener's respective left and right ear.
  • the hearing aid devices HL, HR have respective electronic circuits EL, ER connected to drive miniature loudspeakers, called receivers in hearing aids.
  • the electronic circuits EL, ER are connected to microphones in order to amplify sound captures in the environments.
  • the hearing aid system can be connected to an external sound source that provides an audio input signal X, either an electrical wired input signal or an input signal in a wireless form.
  • the input signal X is received by a processor unit PU that includes an audio processor AP, such as explained above.
  • This audio processor AP generates a set of stereo output signals L, R based on the single channels input X.
  • This set of stereo output signals L, R are then applied to the respective electronic circuits EL, ER of the hearing aid devices HL, HR.
  • the listener wearing the hearing aid devices HL, HR can listen to the acoustic left and right output signals AL, AR based on sound from a connected external sound source (such as a CD player, an MP3 player, a radio or a mobile phone etc.) in a pleasant way, even though the original sound from the external sound source is in the form of a mono signal.
  • a connected external sound source such as a CD player, an MP3 player, a radio or a mobile phone etc.
  • the audio processor AP converting the mono signal X to a stereo signal L, R is included in a separate unit PU that can be applied with a suitable signal processor, still with a size to fit in a user's pocket.
  • the stereo signals L, R can be applied to the hearing aid devices HL, HR by wire or in wireless form.
  • Fig. 7 illustrates another hearing aid system embodiment capable of generating sound based on an input signal X from an external sound source.
  • the processor unit PU receiving the input signal X does not convert the single channel input X to a stereo signal, but transmit the single channel input signal X to both of the hearing aid devices HL, HR in the form of a Radio Frequency (RF) signal by means of a built-in RF transmitter RFT.
  • RF Radio Frequency
  • Each of the hearing aid devices HL, HR has RF receivers RFL, RFR arranged to receive the RF signal transmitted from the processor unit PU.
  • the input signal X can then be regenerated based on the received RF signals.
  • Each hearing aid device HL, HR then has one part of an audio processor APL, APR, e.g.
  • Fig. 6 and 7 may alternatively be implemented in a stereo headphone, a headset, an assistive listening device, hearing aids, a set of loudspeakers arranged for positioning close to a listener's ears etc.
  • the invention provides an audio processor (or mono-to-stereo converter) arranged to receive a single channel audio input signal X and generate a set of stereo audio output signals L, R in response.
  • the outputs L, R are based on four delayed versions S1, S2, S3, S4 of the input signal X.
  • S1 is delayed by delay d1 in relation to X
  • S2 is delayed by delay d2 in relation to S1.
  • S3 is delayed by a delay d3 in relation to X
  • S4 is delayed by delay d4 in relation to S3.
  • the output L is then generated as a sum of X, S1, and S4, while the output R is generated as a sum of X, S2, and S3.
  • Delays d2 and d4 are selected to be within 50 ⁇ s to 1 ms, e.g. 450-650 ⁇ s.
  • Such processor produces a stereo signal suited for headphone listening without the feeling of in-head localization and still with a natural timbre. Additionally, low-pass or band-pass filters and appropriate gains can be applied for further refinement.
  • the audio processor can be implemented with a low signal processing requirement and is thus suited as mono-to-stereo converter in portable equipment such as mobile phones, hearing aids etc.

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Claims (25)

  1. Processeur audio conçu pour recevoir un signal d'entrée audio (X) mono et produire un premier et un deuxième signal de sortie audio (L, R) en réponse à celui-ci, le processeur étant conçu pour
    - produire une première (S1) version retardée du signal d'entrée audio (X) étant retardée d'un premier (d1) délai par rapport au signal d'entrée audio (X),
    - produire une deuxième (S2) version retardée du signal d'entrée audio (X), retardée d'un deuxième (d2) délai par rapport à la première (S1) version retardée du signal d'entrée audio (X),
    - produire une troisième (S3) version retardée du signal d'entrée audio (X) étant retardée d'un troisième (d3) délai par rapport au signal d'entrée audio (X),
    - produire une quatrième (S4) version retardée du signal d'entrée audio (X), retardée d'un quatrième (d4) délai par rapport à la troisième (S3) version retardée du signal d'entrée audio (X),
    - produire le premier signal de sortie audio (L) sous la forme d'une première somme du signal d'entrée audio (X), de la première (S1) et de la quatrième (S4) version retardée du signal d'entrée audio (X),
    - produire le deuxième signal de sortie audio (R) sous la forme d'une deuxième somme du signal d'entrée audio (X), de la deuxième (S2) et de la troisième (S3) version retardée du signal d'entrée audio (X),
    caractérisé en ce que
    le premier (d1) et le troisième (d3) délai se situent dans une première plage de délais allant de 20 ms à 100 ms, le premier (d1) et le troisième (d3) délai étant différents,
    et le deuxième (d2) et le quatrième (d4) délai se situent dans une deuxième plage de délais allant de 50 µs à 1 ms.
  2. Processeur audio selon la revendication 1, dans lequel la première plage de délais va de 30 ms à 80 ms, par exemple de 40 ms à 70 ms, par exemple de 50 ms à 60 ms.
  3. Processeur audio selon la revendication 1 ou 2, dans lequel la deuxième plage de délais va de 50 µs à 800 µs, de 200 µs à 700 µs, par exemple de 450 µs à 650 µs.
  4. Processeur audio selon l'une quelconque des revendications précédentes, dans lequel le premier et le troisième délai (d1, d3) sont choisis de manière à ce qu'une différence entre le premier délai (d1) et le troisième délai (d3) se situe dans une troisième plage de délais allant de 1 ms à 30 ms, par exemple de 3 ms à 15 ms, par exemple de 5 ms à 10 ms.
  5. Processeur audio selon l'une quelconque des revendications précédentes, comprenant une section de filtrage passe-bas conçue pour
    - assurer un filtrage passe-bas de la première (S1) version retardée du signal d'entrée audio (X) avant la sommation pour former le premier signal de sortie audio (L), et
    - assurer un filtrage passe-bas de la troisième (S3) version retardée du signal d'entrée audio (X) avant la sommation pour former le deuxième signal de sortie audio (R).
  6. Processeur audio selon la revendication 5, dans lequel la section de filtrage passe-bas est conçue pour
    - assurer un filtrage passe-bas de la deuxième (S2) version retardée du signal d'entrée audio (X) avant la sommation pour former le premier signal de sortie audio (L), et
    - assurer un filtrage passe-bas de la quatrième (S4) version retardée du signal d'entrée audio (X) avant la sommation pour former le deuxième signal de sortie audio (R).
  7. Processeur audio selon la revendication 5 ou 6, dans lequel la section de filtrage passe-bas est conçue pour fournir une fréquence de coupure située dans la plage allant de 300 Hz à 5 kHz, par exemple dans la plage allant de 400 Hz à 3 kHz, par exemple de 500 Hz à 1 kHz.
  8. Processeur audio selon l'une quelconque des revendications 5 à 7, comprenant une section de filtrage passe-bande conçue pour assurer un filtrage passe-bande de l'ensemble des première, deuxième, troisième et quatrième (S1, S2, S3, S4) versions retardées du signal d'entrée audio (X) avant la sommation pour former le premier et le deuxième signal de sortie audio (L, R).
  9. Processeur audio selon la revendication 8, dans lequel la section de filtrage passe-bande est conçue pour fournir une plage de fréquences passe-bande allant de 100 Hz à 5 kHz, par exemple de 300 Hz à 3 kHz, par exemple de 500 Hz à 1 kHz.
  10. Processeur audio selon l'une quelconque des revendications 5 à 9, dans lequel toutes les sections de filtrage sont mises en oeuvre sous forme de filtres de premier ordre.
  11. Processeur audio selon l'une quelconque des revendications précédentes, dans lequel la première somme comprend une cinquième (S5) version retardée du signal d'entrée audio (X), et dans lequel la deuxième somme comprend une sixième (S6) version retardée du signal d'entrée audio (X).
  12. Processeur audio selon l'une quelconque des revendications précédentes, comprenant une section d'atténuation conçue pour atténuer les première, deuxième, troisième et quatrième (S1, S2, S3, S4) versions retardées par rapport au signal d'entrée audio (X) avant de produire la première et la deuxième somme.
  13. Processeur audio selon la revendication 12, dans lequel les première, deuxième, troisième et quatrième (S1, S2, S3, S4) versions retardées du signal d'entrée audio (X) sont atténuées d'au moins 2 dB par rapport au signal d'entrée audio (X), par exemple d'au moins 4 dB, par exemple d'au moins 6 dB, par exemple d'au moins 10 dB.
  14. Processeur audio selon l'une quelconque des revendications précédentes, conçu pour atténuer le signal d'entrée audio (X) avant de produire les versions retardées (S1, S2, S3, S4) du signal d'entrée audio (X).
  15. Dispositif (PU) comprenant un processeur audio (AP) selon l'une quelconque des revendications 1 à 14.
  16. Système comprenant
    - un dispositif (PU) selon la revendication 15, et
    - un premier et un deuxième transducteur électroacoustique conçus pour
    - convertir les premier et deuxième signaux de sortie audio (L, R) respectifs en premier et deuxième signaux acoustiques (AL, AR) respectifs, et
    - reproduire les premier et deuxième signaux acoustiques (AL, AR) respectifs dans les oreilles gauche et droite respectives d'un utilisateur.
  17. Système selon la revendication 16, dans lequel le premier et le deuxième transducteur électroacoustique sont compris dans l'un des éléments suivants : un jeu d'écouteurs, un casque d'écoute, un jeu d'appareils auditifs (HL, HR), un jeu de dispositifs d'aide à l'écoute, un jeu de haut-parleurs conçus pour être positionnés à proximité des oreilles d'un auditeur.
  18. Système selon la revendication 16 ou 17, comprenant un jeu de dispositifs auditifs gauche et droit (HL, HR) conçus pour être positionnés dans les oreilles gauche et droite respectives d'un utilisateur.
  19. Système selon la revendication 18, comprenant une unité séparée (PU) comprenant le processeur audio (AP), dans lequel l'unité séparée (PU) est conçue pour recevoir le signal d'entrée audio (X) provenant d'un dispositif externe et pour fournir le premier et le deuxième signal de sortie audio (L, R) aux dispositifs auditifs gauche et droit (HL, HR) respectifs.
  20. Système selon la revendication 19, dans lequel l'unité séparée (PU) comprend un transmetteur sans fil conçu pour transmettre des signaux représentant le premier et le deuxième signal de sortie audio aux récepteurs respectifs des dispositifs auditifs gauche et droit.
  21. Système selon la revendication 18, comprenant une unité séparée (PU) conçue pour recevoir le signal d'entrée audio (X) provenant d'un dispositif externe et pour fournir ce signal d'entrée audio (X) aux dispositifs auditifs gauche et droit, dans lequel le dispositif auditif gauche (HL) comprend un processeur (APL) conçu pour produire le premier signal de sortie audio (L) et le dispositif auditif droit (HR) comprend un processeur (APR) conçu pour produire le deuxième signal de sortie audio (R).
  22. Système selon la revendication 21, dans lequel l'unité séparée (PU) comprend un transmetteur sans fil (RFT) conçu pour transmettre un signal représentant le signal d'entrée audio (X) aux récepteurs (RFL, RFR) respectifs des dispositifs auditifs gauche et droit (HL, HR).
  23. Méthode de conversion d'un signal d'entrée audio (X) mono unique en un ensemble composé d'un premier et d'un deuxième signal de sortie audio (L, R), la méthode comprenant
    - production d'une première (S1) version retardée du signal d'entrée audio (X) étant retardée d'un premier (d1) délai par rapport au signal d'entrée audio (X),
    - production d'une deuxième (S2) version retardée du signal d'entrée audio (X), retardée d'un deuxième (d2) délai par rapport à la première (S1) version retardée du signal d'entrée audio (X),
    - production d'une troisième (S3) version retardée du signal d'entrée audio (X) étant retardée d'un troisième (d3) délai par rapport au signal d'entrée audio (X),
    - production d'une quatrième (S4) version retardée du signal d'entrée audio (X), retardée d'un quatrième (d4) délai par rapport à la troisième (S3) version retardée du signal d'entrée audio (X),
    - production du premier signal de sortie audio (L) sous la forme d'une première somme du signal d'entrée audio (X), de la première (S1) et de la quatrième (S4) version retardée du signal d'entrée audio (X),
    - production du deuxième signal de sortie audio (R) sous la forme d'une deuxième somme du signal d'entrée audio (X), de la deuxième (S2) et de la troisième (S3) version retardée du signal d'entrée audio (X),
    caractérisée en ce que
    le premier (d1) et le troisième (d3) délai se situent dans une première plage de délais allant de 20 ms à 100 ms, le premier (d1) et le troisième (d3) délai étant différents,
    et le deuxième (d2) et le quatrième (d4) délai se situent dans une deuxième plage de délais allant de 50 µs à 1 ms.
  24. Code programme exécutable par un ordinateur conçu pour mettre en oeuvre la méthode selon la revendication 23.
  25. Support de données comprenant un code programment exécutable par un ordinateur selon la revendication 24.
EP08102617A 2008-03-14 2008-03-14 Processeur audio pour la conversion d'un signal mono en un signal stéréo Active EP2101517B1 (fr)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP08102617A EP2101517B1 (fr) 2008-03-14 2008-03-14 Processeur audio pour la conversion d'un signal mono en un signal stéréo
AT08102617T ATE522094T1 (de) 2008-03-14 2008-03-14 Audioprozessor zur umwandlung eines monosignals in ein stereosignal

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP08102617A EP2101517B1 (fr) 2008-03-14 2008-03-14 Processeur audio pour la conversion d'un signal mono en un signal stéréo

Publications (2)

Publication Number Publication Date
EP2101517A1 EP2101517A1 (fr) 2009-09-16
EP2101517B1 true EP2101517B1 (fr) 2011-08-24

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EP (1) EP2101517B1 (fr)
AT (1) ATE522094T1 (fr)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US9549260B2 (en) 2013-12-30 2017-01-17 Skullcandy, Inc. Headphones for stereo tactile vibration, and related systems and methods

Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DK2806661T3 (en) * 2013-05-23 2017-12-11 Gn Resound As A hearing aid with spatial signal enhancement
US10425747B2 (en) 2013-05-23 2019-09-24 Gn Hearing A/S Hearing aid with spatial signal enhancement

Family Cites Families (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5173944A (en) * 1992-01-29 1992-12-22 The United States Of America As Represented By The Administrator Of The National Aeronautics And Space Administration Head related transfer function pseudo-stereophony
TW411723B (en) 1996-11-15 2000-11-11 Koninkl Philips Electronics Nv A mono-stereo conversion device, an audio reproduction system using such a device and a mono-stereo conversion method
US7076071B2 (en) * 2000-06-12 2006-07-11 Robert A. Katz Process for enhancing the existing ambience, imaging, depth, clarity and spaciousness of sound recordings
JP3557177B2 (ja) * 2001-02-27 2004-08-25 三洋電機株式会社 ヘッドホン用立体音響装置および音声信号処理プログラム
EP1251717A1 (fr) * 2001-04-17 2002-10-23 Yellowknife A.V.V. Procédé et circuit pour l'écoute au casque d'un enrégistrement audio

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US9549260B2 (en) 2013-12-30 2017-01-17 Skullcandy, Inc. Headphones for stereo tactile vibration, and related systems and methods
US10063976B2 (en) 2013-12-30 2018-08-28 Skullcandy, Inc. Headphones for stereo tactile vibration, and related systems and methods

Also Published As

Publication number Publication date
ATE522094T1 (de) 2011-09-15
EP2101517A1 (fr) 2009-09-16

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