Determining the distance and/or acoustic quality between a mobile device and a base unit

ES 2 644 529 T3

DESCRIPTION

Determine the distance and / or acoustic quality between a mobile device and a base unit

Technical field of the invention

The invention relates to determining whether a mobile device is in proximity to a base unit, and in particular to determining the proximity in terms of distance and / or acoustic quality between a mobile device and a base unit using acoustic signals.

Background of the invention

A personal emergency response system (PERS) comprises a base unit that is typically located in a user's home and comprises at least one microphone and one speaker. The base unit is connected to a remote monitoring station via a communication network (for example a mobile network or a PSTN) that allows a user to contact their healthcare provider or an emergency service if they require assistance.

A mobile PERS (MPERS) also has a small mobile or portable device (for example in the form of a mobile phone, wristwatch or pendant) that is carried or carried by the user. The mobile device has a respective microphone and speaker and is wirelessly connected to the base unit. The user can use the mobile device to communicate with the remote monitoring station through the base unit. The mobile device can also comprise one or more sensors to monitor the status or health of the user (for example accelerometers to detect falls by the user). The mobile device may also comprise an emergency or panic button that the user can use to quickly contact the remote monitoring station.

Document US 2010/0311388 describes a system comprising a portable device having a cellular transceiver module, a GPS module and an emergency call button, and a base station that communicates with a response center. A user also has a wearable pendant that has a panic button, and when the panic button is activated, a signal is emitted to the wearable device. If the portable device is not in proximity to the base station, which is determined based on RF signals, the portable device contacts the response center in response to activation of the panic button. If the portable device is in proximity to the base station, the base station receives the signal and communicates with the response center. Although the portable device is in proximity to the base station, the GPS module is disconnected and the cell is put into standby mode.

Therefore, in this system, the power consumed by the portable device can be reduced when it is in close proximity to a base station. However, it is desirable to be able to further reduce the power consumption of a mobile device in this type of system. This is particularly the case during an emergency call, as the call may need to stay with the remote monitoring station for some time. It is also known according to patent application US2008 / 0304361A1, an acoustic telemetry method for determining a distance between a first device and at least one other device using one or more acoustic signals.

Summary of the invention

It is provided that the power consumption of a mobile device in this type of system can be reduced by disconnecting components in the mobile device used to reproduce audio (for example a digital-to-analog converter, audio amplifier, speaker) when the user (and therefore therefore the mobile device) is in close proximity to the base unit. Audio or voice for the user can instead be output from the speaker on the base unit. However, the loudspeaker and related components on the mobile device should only be disconnected if the user is close enough to the base unit to be able to hear and understand the voice broadcast from the loudspeaker on the base unit.

Therefore, it is not appropriate to use conventional RF signal-based methods to determine the proximity of a user and mobile device to a base unit, since these methods may determine that the user and mobile device are in proximity to the unit. baseline (for example based on a received RF signal strength or triangulation by an additional access point), even though the user may not be able to clearly hear and understand the voice from the base unit (for example the user and mobile device may be in a room adjacent to or different from the base unit, with the intermediate door closed).

Therefore, there is a need for an alternative approach that determines the proximity of a mobile device (and a user) to a base unit to allow a speaker to be disconnected from the mobile device during a communication, while ensuring that the user can hear. and picking up the voice from the speaker on the base unit.

ES 2 644 529 T3

It is provided that the proximity of the mobile device to the base unit is determined by evaluating the distance or acoustic quality between the mobile device and the base unit. In particular examples, acoustic quality is evaluated in terms of the quality of the user's voice as received at the base unit and / or the quality of the sound received at the mobile device from the base unit speaker.

The present invention is defined in and by the appended claims.

Various examples are defined in the following articles.

Article 1. A method of determining the distance and / or acoustic quality between a mobile device and a base unit, the method comprising receiving an acoustic signal by a microphone in one of said mobile device and the base unit; determining a correlation of a second signal with the received acoustic signal; and determining the acoustic quality and / or the distance between the mobile device and the base unit based on one or more peaks in the result of the correlation determining step.

Article 2. A method as indicated in Article 1, in which the correlation determination stage comprises determining coefficients of an adaptive filter to eliminate the second signal from the received acoustic signal, and in which the result of the stage The correlation determination method comprises an adaptive filter impulse response.

Article 3. A method as indicated in Article 2, which additionally comprises the stage of smoothing the square magnitude of the determined coefficients.

Article 4. A method as indicated in Article 1, 2 or 3, in which the acoustic signal received by the microphone in said one of said mobile device and the base unit is received from a speaker in the other of said device mobile and the base unit that emits the acoustic signal in response to a control signal.

Article 5. A method as indicated in Article 4, in which the second signal corresponds to the signal used to control the loudspeaker in said other of said mobile device and the base unit to emit the acoustic signal.

Article 6. A method as indicated in Article 4, the method additionally comprising the stage of receiving the acoustic signal in a microphone in said other of said mobile device and the base unit.

Article 7. A method as indicated in Article 6, in which the second signal corresponds to the acoustic signal received in said other from said mobile device and the base unit.

Article 8. A method as indicated in any of Articles 4 to 7, in which the stage of determining the acoustic quality and / or the distance between the mobile device and the base unit comprises detecting a single peak in the result of the determination stage.

Article 9. A method as indicated in Article 8, in which the determining step comprises determining the distance between the mobile device and the base unit from the timing of the peak detected in the result of the determining step.

Item 10. A method as stated in Item 8 or 9, wherein the determination step comprises determining the acoustic quality between the mobile device and the base unit from (i) the spread of the detected peak; and / or (ii) the reverberation time around the detected peak.

Article 11. A method as indicated in Article 4, the method further comprising the step of controlling a loudspeaker in said one of said mobile device and the base unit to emit the acoustic signal; wherein the step of receiving an acoustic signal by the microphone in one of said mobile device and the base unit comprises receiving the acoustic signal from the loudspeaker in said mobile device and said base unit.

Article 12. A method as indicated in Article 11, in which the stage of determining the acoustic quality and / or the distance between the mobile device and the base unit comprises detecting a first peak in the result of the corresponding determination stage to the acoustic signal from the speaker in said one of said mobile device and the base unit and a second peak corresponding to the acoustic signal from the speaker in said other of said mobile device and the base unit.

Article 13. A method as indicated in Article 12, in which the stage of determining the acoustic quality and / or the distance comprises determining the distance between the mobile device and the base unit from the distance between the first and second peaks in the result of the determination stage.

Article 14. A method as indicated in Article 12 or 13, in which the stage of determining the acoustic quality and / or the distance comprises determining the acoustic quality between the mobile device and the base unit

ES 2 644 529 T3 from (i) the ratio of the magnitudes of the detected peaks; (ii) the scattering around the second peak; and / or (iii) the reverberation time around the second peak.

Article 15. A method as indicated in any of Articles 11 to 14, in which the second signal corresponds to the signal used to control the loudspeaker in said other of said mobile device and the base unit to emit the acoustic signal.

Article 16. A method as stated in any of Articles 4 to 15, wherein the control signal used to control the loudspeaker in said other of said mobile device and the base unit comprises a voice signal received from a station remote monitoring.

Article 17. A method as indicated in any of Articles 4 to 15, in which the control signal used to control the loudspeaker in said other of said mobile device and the base unit comprises a signal derived from an audio file stored locally on said mobile device and / or said base unit.

Article 18. A method as indicated in Article 17, in which the loudspeaker on the other of said mobile device and the base unit that emits the acoustic signal in response to the control signal derived from the audio file is activated in response to a trigger signal sent from said one of said mobile device and the base unit.

Article 19. A method as indicated in Item 1, 2 or 3, wherein the step of receiving an acoustic signal comprises receiving an acoustic signal by a microphone in the base unit, the method further comprising the step of receiving a signal acoustic by a microphone on the mobile device, wherein the acoustic signal received by the microphone in the mobile device is used as the second signal in the step of determining a correlation of a second signal with the received acoustic signal.

Article 20. A method as indicated in Article 19, in which the acoustic signal received by the microphones in the mobile device and the base unit is the voice of a user of the mobile device.

Article 21. A method as indicated in Article 19 or 20, in which the stage of determining the acoustic quality and / or the distance between the mobile device and the base unit comprises detecting a single peak in the result of the determination stage.

Article 22. A method as indicated in Article 21, in which the determining step comprises determining the distance between the mobile device and the base unit from the timing of the peak detected in the result of the determining step.

Article 23. A method as indicated in Article 21 or 22, in which the determination step comprises determining the acoustic quality between the mobile device and the base unit from (i) the dispersion around the detected peak; and / or (ii) the reverberation time around the detected peak.

Article 24. A method of reducing the power consumption of a mobile device, the method comprising determining the distance and / or acoustic quality between the mobile device and a base unit according to the method according to any of Articles 1 to 2. 3; and deactivating a speaker of the mobile device if it is determined that the mobile device is closer than a threshold distance to the base unit or the acoustic quality is greater than a threshold value.

Article 25. A system, which includes:

a mobile device; a base unit; and a processor; wherein one of the mobile device and the base unit comprises a microphone for receiving an acoustic signal; and wherein the processor is configured to determine a correlation of a second signal with the received acoustic signal; and determining the acoustic quality and / or the distance between the mobile device and the base unit based on one or more peaks in the result of the determining step.

Article 26. A system as indicated in Article 25, in which the processor is configured to determine the correlation by determining coefficients of an adaptive filter to eliminate the second signal from the received acoustic signal, and in which the result of the stage The correlation determination method comprises an adaptive filter impulse response.

Article 27. A system as indicated in Article 26, in which the processor is additionally configured to smooth the square magnitude of the determined coefficients.

Article 28. A system as indicated in Article 25, 26 or 27, in which the other of said mobile device and the base unit comprises a loudspeaker to emit the acoustic signal in response to a control signal.

ES 2 644 529 T3

Article 29. A system as indicated in Article 28, in which the second signal corresponds to the signal used to control the loudspeaker in said other of said mobile device and the base unit to emit the acoustic signal.

Article 30. A system as indicated in Article 28, in which the other of said mobile device and the base unit comprises a microphone to receive the acoustic signal.

Article 31. A system as indicated in Article 30, in which the second signal corresponds to the acoustic signal received by the microphone in said other of said mobile device and the base unit.

Article 32. A system as indicated in any of Articles 25 to 31, in which the processor is configured to determine the acoustic quality and / or the distance between the mobile device and the base unit by detecting a single peak in the result. of the correlation.

Article 33. A system as indicated in Article 32, in which the processor is configured to determine the distance between the mobile device and the base unit from the timing of the peak detected in the correlation result .

Article 34. A system as indicated in Article 32 or 33, in which the processor is configured to determine the acoustic quality between the mobile device and the base unit from (i) the dispersion around the detected peak; and / or (ii) the reverberation time around the detected peak.

Article 35. A system as indicated in Article 25, wherein said one of said mobile device and the base unit additionally comprises a loudspeaker to emit the acoustic signal; and wherein the microphone in said one of said mobile device and the base unit receives an acoustic signal from the speakers in said mobile device and said base unit.

Article 36. A system as indicated in Article 35, in which the processor is configured to determine the acoustic quality and / or the distance between the mobile device and the base unit by detecting a first peak in the correlation result that corresponds to the acoustic signal from the loudspeaker in said one of said mobile device and the base unit and a second peak corresponding to the acoustic signal from the loudspeaker in said other of said mobile device and the base unit.

Article 37. A system as indicated in Article 36, in which the processor is configured to determine acoustic quality and / or distance by determining the distance between the mobile device and the base unit from the distance between the first and second peaks.

Article 38. A system as indicated in Article 36 or 37, in which the processor is configured to determine the acoustic quality and / or the distance determining the acoustic quality between the mobile device and the base unit from (i) the ratio of the magnitudes of the detected peaks; (ii) the scattering around the second peak; and / or (iii) the reverberation time around the second peak.

Article 39. A system as indicated in any of Articles 35 to 38, in which the second signal corresponds to the signal used to control the loudspeaker in said other of said mobile device and the base unit to emit the acoustic signal.

Article 40. A system as indicated in any of Articles 27 to 39, wherein the control signal used to control the loudspeaker in said other of said mobile device and the base unit comprises a voice signal received from a station remote monitoring.

Article 41. A system as indicated in any of Articles 27 to 39, in which the control signal used to control the loudspeaker in said other of said mobile device and the base unit comprises a signal derived from an audio file stored locally on said mobile device and / or said base unit.

Article 42. A system as indicated in Article 41, in which the loudspeaker on the other of said mobile device and the base unit that emits the acoustic signal in response to the control signal derived from the audio file is configured to activate in response to a trigger signal sent from said one of said mobile device and the base unit.

Article 43. A system as indicated in Article 25, 26 or 27, in which each of the mobile device and the base unit comprises a microphone to receive an acoustic signal, in which the processor is configured to use the signal acoustic received by the microphone in the mobile device as the second signal when determining the correlation of the second signal with the received acoustic signal.

Article 44. A system as indicated in Article 43, in which the acoustic signal received by the microphones in the mobile device and the base unit is the voice of a user of the mobile device.

ES 2 644 529 T3

Article 45. A system as indicated in Article 43 or 44, in which the processor is configured to determine the acoustic quality and / or the distance between the mobile device and the base unit by detecting a single peak in the result of the correlation.

Article 46. A system as indicated in Article 45, in which the processor is configured to determine the distance between the mobile device and the base unit from the timing of the peak detected in the correlation result.

Article 47. A system as indicated in Article 45 or 46, in which the processor is configured to determine the acoustic quality between the mobile device and the base unit from (i) the dispersion around the detected peak; and / or (ii) the reverberation time around the detected peak.

Article 48. A system as indicated in any of Articles 25 to 47, in which the mobile device comprises a loudspeaker, and in which the mobile device is configured to deactivate the loudspeaker if it is determined that the mobile device is closer of a threshold distance to the base unit or the acoustic quality is greater than a threshold value.

Brief description of the drawings

Embodiments of the invention will now be described, by way of example only, with reference to the following drawings, in which:

Figure 1 is a block diagram of a mobile device and the base unit according to the invention;

Figure 2 is a flow chart illustrating a method of operation of a mobile device during a call to a remote monitoring station;

Figure 3 is a graph illustrating acoustic impulse response in a room showing direct and diffuse fields;

Figures 4A-B show an adaptive filter impulse response for a user at two different distances from a base unit;

Figure 5 is a diagram illustrating the processing performed in a first embodiment of the invention; Figure 6 is a flow chart illustrating the processing steps performed in the first embodiment; Figure 7 is a graph illustrating the square magnitude of the impulse response of an adaptive filter and a corresponding smoothed response;

Figure 8 is a diagram illustrating the processing performed in a second embodiment of the invention; and Figure 9 is a flow chart illustrating the processing steps performed in the second embodiment.

Detailed description of the preferred embodiments

Although the invention is described below as applied to a mobile personal emergency response system (MPERS), it will be appreciated that the methods of detecting the proximity of a mobile device to a base unit can be applied to other types of systems.

Additionally, while the invention is described as being to determine whether it is possible to conserve power in a mobile device by disconnecting the loudspeaker (and associated components), it will be appreciated that the result of proximity sensing can be used for many other purposes, such as performing handovers of a call in progress from one communication unit to another between a set of units distributed in the home, or to automatically adjust the gain of an amplifier on a base unit.

Part of an MPERS 2 for a user 4 in which the present invention can be implemented is shown in Figure 1. The MPERS 2 comprises a mobile device 6 and a base unit 8.

The mobile device 6 can take any suitable form, for example a pendant that is worn around the user's neck, a wristwatch, a mobile phone, PDA, etc., and is generally in close proximity to the user 4. The device Mobile 6 comprises transceiver circuitry 10 and associated antenna 12 for communicating with base unit 8. The transceiver circuitry 10 can be configured to use any suitable communication protocol, including, for example, Bluetooth or DECT. Furthermore, the transceiver circuitry 10 can be configured to use a mobile telecommunications protocol, for example, GSM or CDMA, so that the mobile device 6 can directly contact a remote monitoring station if the mobile device 6 is not in range of base unit 8 (for example if user 4 is away from home).

Mobile device 6 also comprises a processor 14, speaker 16, and microphone 18. Processor 14 controls the operation of mobile device 6. Speaker 16 is provided to output audio (usually voice) received from the remote monitoring station via the monitor unit. base 8 (provided that mobile device 6 is in range of base unit 8). In Figure 1, block 16 also represents other components associated with speaker 16, for example a digital-to-analog converter and audio amplifier.

ES 2 644 529 T3

Microphone 18 detects audio (again, usually voice) from user 4 and converts the audio to an electrical signal for transmission to the remote monitoring station (again, via base unit 8 if mobile device 6 is in range) .

The mobile device 6 also comprises a battery 20 or other suitable power source that supplies power to the various components of the mobile device 6. The processor 14 can be configured to control which of the components (for example the circuitry of transceiver 10, speaker 16 and microphone 18) is powered by battery 20 at any particular time.

In some embodiments, the mobile device 6 may include a memory 21 that stores pre-recorded audio files for playback through the speaker 16 of the mobile device 6. These audio files can be broadcast in an emergency situation if the speaker 16 is powered, for example to inform a user 4 that a connection to the remote monitoring station is being established.

Base unit 8 comprises transceiver circuitry 22 and associated antenna 24 for communicating with mobile device 6. As in mobile device 6, transceiver circuitry 22 in base unit 8 can be configured to use any suitable communications protocol. , including, for example, Bluetooth or DECT. In addition, the transceiver circuitry 22 can be configured to use a mobile telecommunications protocol, eg, GSM or CDMA, to allow the base unit 8 to contact the remote monitoring station. Alternatively, or in addition, the transceiver circuitry 22 may be configured to connect to a public switched telephone network (PSTN) via a wall socket in the user's home.

Base unit 8 also comprises a processor 26, speaker 28, and microphone 30. Processor 26 controls the operation of base unit 8. Speaker 28 is provided to output audio (usually voice) received from the remote monitoring station. In some embodiments, the base unit 8 comprises a memory 31 that includes the same pre-recorded audio files as stored in the memory 21 of the mobile device 6 for automatic playback during an emergency call.

The microphone 30 detects audio (again, usually voice) from the user 4 and converts the audio into an electrical signal for transmission to the remote monitoring station.

Base unit 8 also comprises a power supply unit, PSU, 32 that connects to a power source in the user's home and supplies power to the various components of base unit 8. PSU 32 may include also a battery or other suitable power source to act as a backup power source in the event that there is an interruption in the electrical source or the electrical source fails otherwise.

As described above, the invention provides a way to adaptively determine the distance of a user 4 from the base unit 8 in real time, based solely on acoustic measurements. In this way, if user 4 is deemed to be located in close proximity to base unit 8, then speaker 16 on mobile device 6 can be disconnected and audio can only be output from speaker 28 on base unit 8. instead. However, if user 4 is located too far from base unit 8 to clearly hear and understand the sound from speaker 28 on base unit 8, then speaker 28 on mobile device 6 can be connected during a phone call. emergency (similarly if mobile device 6 is out of wireless range of base unit 8).

Unlike radio-based proximity (i.e. RF) methods, embodiments of the invention allow the quality of the voice signal picked up by the base unit 8 to be measured, and / or the quality of the Voice signal from base unit 8 received in mobile device 6, since effects such as reverberation can be taken into account. For example, in cases where user 4 (or base unit 8) is located outside the reverberation radius (also known as the critical distance) of the sound source (i.e. user 4's speaker or base unit 28) where the direct and diffuse contributions of a sound source are equal in intensity, the speaker 16 on the mobile device 6 should be connected.

A flow chart illustrating a method of operation of a mobile device during a call at a remote monitoring station is shown in Figure 2. In step 101, the mobile device 6 initiates a call to the remote monitoring station. The call can be activated by the user 4 by pressing an emergency button or another on the mobile device 6, or by one or more sensors on the mobile device 6 that detect an emergency event (such as a fall by the user 4).

In step 103, the mobile device 6 determines whether it is in wireless range of the base unit 8. This can be determined, for example, by determining the mobile device 6 whether it can receive a signal from the base unit 8. Those skilled in the art subject matter are aware of alternative ways in which this can be determined.

ES 2 644 529 T3

If the mobile device 6 is not in wireless range of the base unit 8, the mobile device 6 contacts the remote monitoring station directly using its transceiver circuitry 10 and antenna 12. In this case, the transceiver circuitry 10 is powered. , speaker 16 and microphone 18 in the mobile device 6 (step 107).

If mobile device 6 is in wireless range of base unit 8, mobile device 6 connects wirelessly to base unit 8, and base unit 8 establishes a call with the remote monitoring station (step 109). In this case, the GSM / CDMA part of the transceiver circuitry 10 (which may be a separate module from that used to establish a wireless connection to the base unit 8) can be disconnected.

In step 111, it is determined whether a far-end user is active. The far end user is the person or computer in the remote monitoring system (eg, answering center) that provides audio (typically voice) to user 4, which is output from speaker 28 on base unit 8 and speaker 16 on mobile device 6. There are numerous methods for detecting the activity (ie voice) of a far-end user, such as a basic power-based method that compares instantaneous power to some long-term noise floor estimates. Those skilled in the art will be aware of alternative ways in which far-end activity can be detected.

If the far-end user is active, the distance between the base unit 8 and the mobile device 6 is determined using the far-end signal, which is preferably audio broadcast by speaker 16 and speaker 28, as received by the microphone 18 on mobile device 6, illustrated respectively by arrows 34 and 36 in Figure 1. This is step 113 in Figure 2.

As an alternative to using a far-end signal, the distance between base unit 8 and mobile device 6 can be determined using sound from a stored audio file that is output from speaker 16 and speaker 28.

If the far end user is not active (for example if he or she is listening to user 4), or if the base unit 8 is not otherwise generating sound with the speaker 28, the distance between the base unit 8 and mobile device 6 is determined using a near-end signal, which is the audio (i.e. voice) of user 4 as received by microphone 18 on mobile device 6 and microphone 30 on base unit 8, illustrated respectively by arrows 38 and 40 in Figure 1. This is step 115 in Figure 2. Detection of the activity of a near-end user can be performed in a similar manner for the far-end user, and experts In the field, they will be aware of alternative ways in which near-end activity can be detected.

Once the distance between mobile device 6 and base unit 8 has been determined in step 113 or 115, it is determined whether mobile device 6 is in acoustic proximity to base unit 8 (i.e. is the user 4 and the mobile device 6 close enough to the base unit 8 so that the user 4 hears and clearly understands the voice emitted by the loudspeaker 28 in the base unit 8?).

If mobile device 6 is in acoustic proximity to base unit 8, speaker 16 on mobile device 6 can be disconnected (step 119). If the mobile device 6 is not in acoustic proximity to the base unit, the speaker 16 on the mobile device 6 can be connected (step 121).

After steps 119 and 121, the method can go back to step 111 and repeat.

Alternatively, measurements of both far-end (or audio file-based) and near-end signals can be performed before using the results of both measurements to make a decision as to whether mobile device 6 is in acoustic proximity to base unit 8.

The following description of embodiments of the invention refers to determining the correlation between two signals using adaptive filtering. It will be appreciated that a correlation, or similarity between two signals can also be determined using the cross-correlation function directly. A brief description of adaptive filtering is provided below.

An adaptive filter is used to remove a correlation signal present in another signal, and is commonly used for acoustic echo cancellation. In acoustic echo cancellation, the adaptive filter models the linear part of the acoustic echo path between a device speaker and a microphone so that acoustic (linear) echo can be removed from the microphone signal, leaving only the acoustic signal. desired clean voice. If modeled correctly, the adaptive filter coefficients correspond to the acoustic echo path. As shown in Figure 3, and can be divided into a direct field that corresponds to the direct coupling between the speaker and microphone and some early reflections, and the diffuse field that corresponds to the late reflections, and that contributes to reverberation. Figure 3 shows the acoustic impulse response in a room with a high reverberation time (approximately 850 ms), showing the direct and diffuse fields. The y-axis represents the amplitude of the waveform and the x-axis shows the discrete time samples for an impulse response sampled at 8 kHz.

ES 2 644 529 T3

There are numerous adaptive filtering algorithms such as normalized least squares mean (LMS) and normalized LMS (NLMS), which are based on Wiener filter theory, and recursive least squares (RLS) and Kalman filtering. In the following embodiments, Wiener filter theory will be used to derive the optimal adaptive filter coefficients.

The top graph in Figure 4 shows the square-magnitude adaptive filter impulse response for a user who is close (for example 1.6 m) to a receiving device (for example an 8 base unit), and the Bottom graph of Figure 4 shows the square magnitude adaptive filter impulse response for a user who is remote (eg 3m) from a receiving device (eg base unit 8). The large peaks at the beginning of the impulse response correspond to the acoustic path between speaker 16 and microphone 18 on mobile device 6, while the second set of delayed peaks corresponds to the acoustic path between speaker 28 on the speaker unit. base 8 and microphone 18 on mobile device 6.

For a so-called dry room with a low reverberation time, the diffuse part of the impulse response is much lower in power compared to direct field. Additionally, the closer a sound source (for example a user's mouth) is located to the receiving device (for example a base unit), the greater the ratio between the intensity in the direct and diffuse field. At the reverberation radius or critical distance around the receiving device, the intensity of the direct field is equal to that of the diffuse field. Outside of this radius, only diffuse contributions are present, and the intelligibility of the speech picked up by the microphone in the base unit drops considerably. This is a problem that radio-based (ie RF) distance sensing solutions cannot account for.

A diagram illustrating the processing in a first embodiment of the invention is shown in Figure 5. A flow diagram showing the corresponding method of determining the proximity between a mobile device 6 and a base unit 8 is shown in Figure 6.

In the first embodiment of the invention, an adaptive filter is combined with a peak detection scheme to determine the distance between a user (with a mobile device 6) and a base unit 8 using, preferably, two sets of two signals. The first embodiment is based on using the audio output from the speaker 28 in the base unit 8 (i.e. a far-end acoustic signal or an acoustic signal generated from an audio file stored in the memory 31 of the base unit. 8) as shown in step 113 of Figure 2.

In particular, proximity detection is based on an echo signal produced by speaker 16 of mobile device 6 as detected by microphone 18 on mobile device 6 (the direct field of this signal is associated with strong coupling between speaker 16 and microphone 18 and a short delay) and an echo signal produced by speaker 28 in base unit 8 as detected by microphone 18 in mobile device 8 (this echo is associated with a coupling that depends on the distance between the mobile device 6 (ie the user) and the base unit 8). The delay of the second echo signal relative to the direct field of the first echo signal is proportional to the distance between the mobile device 6 and the base unit 8.

In this first embodiment, the processing performed in accordance with the invention is preferably performed by the processor 14 in the mobile device 6, although it will be appreciated that the processing could be performed instead by the processor 26 in the base unit 8.

Figure 5 shows the first embodiment in the frequency domain (with frequency ω). The processing by the processor 14 consists of an adaptive filter 42 between the speaker 16 of the mobile device 6 and its microphone 18, and a logic unit 44 that calculates the distance between the user 4 (mobile device 6) and the base station 8.

To understand the effects of the wireless encoding / decoding process between mobile device 6 and base unit 8 on the resulting adaptive filter solution, Included is a mathematical model of the voice encoder 46 and decoder 48 used to transmit the voice signal from the base station 8 to the mobile device 6 (this voice encoder 46 would be implemented by the processor 26 in the base unit 8 and the decoder 48 by processor 14 in mobile device 6). The encoding operation can be modeled as a time-varying quantization and filter operation. In addition to the channel delay on the wireless channel between base unit 8 and mobile device 6 that can be compensated for by an acoustic echo canceller, packet loss and other non-linear distortions on the wireless channel are no longer considered for this purpose. analysis.

In Figure 5, the far-end signal data as received by the base unit 8 from the remote monitoring station (or the stored audio file) is indicated by Χ (ω) which is output from the speaker 28 of the base station 8 as an acoustic signal (step 201 in Figure 6). The far-end signal data is also transmitted wirelessly to mobile device 6 after which encoding is decoded and transmitted through the wireless channel (Ο (ω)), and output from speaker 16 on mobile device 6 as an acoustic signal (step 203 in Figure 6)

ES 2 644 529 T3

X_m(co) = [Χ (ω) Α (ω) + Q (co)] C (cú) A^_1(cü) (1) where Λ (ω) and Q (ro) model the encoder effects, and A^-1(ro) models the decoder.

Assuming an ideal situation with Q (o) = 0 and C (<'>) = e^-j,'^;', where δ is the transmission delay, the acoustic signal generated by the microphone 18 in the mobile device 6 (step 205) is provided by

Υ (ω) = Χ (ω) [Ηι, (ω) + H_m(ro) e '^j⁵] (2)

The optimal weights W (o) for a criterion of least mean square error (MMSE)

<img file="ES2644529T3_D0001.tif" />

consisting of the sum of the acoustic echo path from speaker 28 in base station 8 to microphone 18 in mobile device 6 plus a delayed version of the acoustic coupling between speaker 16 and microphone 18 in mobile device 6. Normally the RF delay associated with the channel is much less than the sound propagation delay, and therefore it will be assumed to be negligible, i.e. C (o) = 1 (however the signal processing associated with encoding and decoding of the signal will be greater than the propagation delay). Therefore, the ideal solution for W (o) simply consists of the sum of the two echo paths.

Furthermore, since the gains of the individual loudspeaker amplifiers 16, 28 are not explicitly included in the model, they are instead assumed to be part of the echo path impulse responses.

Under normal conditions, however, the encoder 46 introduces distortion in the form of quantization noise Q (o) that can be modeled (at high rates) as zero-mean statistically independent additive white noise with power

<img file="ES2644529T3_D0002.tif" />

(4) which means that the quantized noise power follows the power of the decorrelated signal Xe. The factor G is related to the SNR of quantification.

The far-end signal data (or audio file data) is used to determine coefficients for the adaptive filter 42 to minimize the root mean square error between the output of the adaptive filter 42 and the received signal ν (ω) (step 207) .

An optimal solution of the adaptive filter weights can be derived as follows (with an assumption that user 4 is not speaking and that he contributes to the audio for the received signal ν (ω):

Following equation (1) and the assumption that C (o) = 1, the signal emitted by speaker 16 on mobile device 6 is provided by

X_m(co) = [Χ (ω) Α (ω) + Q (co)] Α^_1(ω) (5) = Χ (ω) + Q (o) A^_1(o) (6) where the effective quantization noise is given by Xq (o) = Q (&lt;b) A '¹(&lt;b).

The signal received by microphone 18 on mobile device 6 is provided by

Υ (ω) - Χ (ω) Η „(ω) + Χ_Μ(ω) Η_Μ(ω) (7) = Χ (ω) [Ηι, (ω) + H_m(ro)] + Q (ro) A^_1(or) H_m(ro) (8) = X (co) [H_b(co) + H_m(ro)] + X_Q(co) H_m(co) (9) where the last term represents the echo of the quantization noise of the input signal.

The optimal weights W (o) to minimize the root mean square error (MSE) between the output of filter 42 and the acoustic signal received at microphone 18 are provided by the Wiener solution

ES 2 644 529 T3

<img file="ES2644529T3_D0003.tif" />

The denominator in (10), which indicates the power spectral density (PSD) of the speaker 16 of the mobile device 6 can be expanded as

<img file="ES2644529T3_D0004.tif" />

The cross-correlation in the numerator of (10) can be simplified as

<img file="ES2644529T3_D0005.tif" />

Looking back at (10), if there is no correlation between the signal from speaker 16 and the signal from microphone 18 across all frequencies, then the Wiener solution is zero.

The expectations of the second and third terms are zero since

<img file="ES2644529T3_D0006.tif" />

The simplification of the last term follows from (11). The resulting Wiener solution is provided by

W_ORpt (°&gt;) = H_nl (ω) + ^ (16)

U +1

Therefore, returning to Figures 5 and 6, the better the signal-to-noise ratio (SNR) of the quantizer, the lower the value of G in equation 16, and the closer the Wiener solution is to the sum of the two impulse responses. This essentially means that the poorer the quantizer, the more difficult it is to find the difference in arrival time between the two echoes. To compensate for this, a gain factor of G + 1 can be applied to the far-end signal as shown by block 50 in the upper branch of Figure 5, although this is normally not necessary since the applied gain is unity base 8 housing a much larger speaker 28, is much higher than the gain applied to speaker 16 in mobile device 6.

A least squares mean algorithm such as the NLMS can be used to update adaptive filter coefficients and track changes in the optimal Wiener solution. This adaptation is carried out by correlating the input signal of the adaptive filter with the residual signal 51. The update recursion of the NLMS coefficient is provided by

ES 2 644 529 T3 (17) w (n +1) = w (n) i ^μ x (n) r (n) IMI * where w (n) and w (n + 1) represent the current and updated time domain filter coefficients of length M, μ the size of the adaptation stage (μ &lt;1) , x (n) the input vector containing the M current input samples, and r (n) the residual signal. The second term in (17) contains the cross-correlation between the input signal and the residual. M is assumed to be large enough to model the acoustic path. For a static environment, the filter coefficients in (17) converge on the optimal solution when x (n) r (n) ~ 0.

Regarding the channel delay and delay produced by signal processing (i.e. encoding and decoding), too much delay can influence the accurate estimation of the time difference between H_m and Hb since this results in a Wiener solution with H_m delayed by an amount equal to the group delay of the processing plus channel delay. Therefore, a compensation delay may be applied to the output of speaker 28, as shown by block 52 in the upper branch of Figure 5.

Since the Wiener solution has been expressed in terms of the individual acoustic echo paths between the speaker 16 and the microphone 18 of the mobile device 6, the logic unit 44 (processor 14) determines whether the mobile device 6 is close to the base unit. 8 (and therefore, in a preferred embodiment, if you disconnect the speaker 18 from the mobile device 6).

In particular, the logic unit 44 examines the energy in the acoustic impulse response (for example as shown in Figure 3), and locates the direct field associated with the speaker 16 on the mobile device 6 and the direct field of the path echo associated with speaker 28 in base unit 8 (step 211 in Figure 6).

Preferably, prior to step 211, the square magnitude of the adaptive filter coefficients is smoothed. Next, a moving average (MA) filter is used to smooth the square magnitude of the adaptive filter coefficients. This finite impulse response (FIR) filter is provided by

1] (18) where N is the length of the filter. The longer the filter, the smoother and wider there is of the adaptive filter's impulse response. The value of N can be set depending on practical considerations of the smallest conceived distance between mobile device 6 and base unit 8. At 8 kHz, a sample corresponds to 125 ps, the time required for sound to travel 4, 3 cm. Preferably, N can be set between 4 and 12 samples (corresponding to distances of 17 and 50 cm) to avoid false detections.

Selecting a value for N that is too large results in staining of the square magnitude impulse response. Figure 7 shows the smoothed square magnitude impulse response of the adaptive filter 42.

The logic unit 44 can locate the direct field associated with the speaker 16 in the mobile device 6 and the direct field of the echo path associated with the speaker 28 in the base unit 8 in step 211 of Figure 6 by identifying maximum values local in the smoothed magnitude impulse response, i.e. those samples with a magnitude greater than both of the neighboring left and right samples, i.e.

<img file="ES2644529T3_D0007.tif" />

(19)

The second largest local maximum value can be taken to correspond to the impulse response of the echo path (Hb) from the base unit 8. The distance d between this and the global maximum value (i.e. the echo path H_m from the speaker 16 of the mobile device 6) corresponds to the sample distance between the mobile device 6 / user 4 and the base unit 8 (step 213 in Figure 6). This distance is then converted to the distance in meters according to

<img file="ES2644529T3_D0008.tif" />

(20) where f_s is the sampling frequency in Hz and c is the speed of sound in air in ms'¹ (which is approximately

343 ms'¹). In the example shown in Figure 7, the distance is calculated to be 64 x 343/8000 = 2.75 m.

ES 2 644 529 T3

Depending on this value,

Sw = 1 (i.e. speaker 16 is connected), if D&gt; Thl (21)

Sw = 0 (ie loudspeaker 16 is disconnected), if D &lt;Thl (22) where Th1 is some threshold distance, for example Th1 = 3 m. It will be appreciated that there are other methods of identifying peaks in the total impulse response to determine the distance between the direct field associated with the speaker 16 on the mobile device 6 and the direct field of the echo path associated with the speaker 28 on the unit. base 8.

Additionally, the magnitude of the second largest maximum that corresponds to the echo path impulse response from the speaker 28 in the base unit 8 compared to that of the overall maximum value can be used as another measure (or in conjunction with the measure away), that is

Sw = 0 (i.e. speaker 16 is disconnected), if p2 / p_max &gt; Th2 (23)

Sw = 1 (i.e. speaker 16 is connected), otherwise (24) where Th2 is some threshold that depends on the speaker amplifier gain of base unit 8, and is preferably less than unity, p_max is the magnitude of the global maximum and p2 is the amplitude of the second largest maximum. The Th2 threshold can be determined experimentally based on the amplifier settings for speakers 16 and 28. Unlike the distance measurement, this measurement takes into account the quality of the acoustic path between user 4 and mobile device 6 and the unit. of base 8, that is to say if there are losses of acoustic energy due to reflections and reverberation.

It will be appreciated that there are other measures to evaluate the acoustic quality between the user 4 and mobile device 6 and the base unit 8, such as using the spread of the impulse response around the second peak as a measure of the amount of reverberation. This dispersion can be based on a measure of localized scarcity such as

<img file="ES2644529T3_D0009.tif" />

where N is the length of the analysis window around the second peak and w the filter coefficients in that analysis window. || || iy || · ||₂ represent vector norms 1 and 2, respectively. A second measure that can be used is the estimate of the reverberation time or T60 of the impulse response around the second peak. T60 indicates the reverberation time which is defined as the time it takes for the energy in the diffuse field of the echo to decay by 60 dB. So in a highly reverberant room this time will be quite long (about 1 s, like the impulse response in Figure 3), but for a dry room this is normally less than 350 ms. T60 is measured by analyzing the impulse response and the so-called energy decay curve (EDC).

It will be appreciated that the above discussion of this first embodiment assumes that the speaker 16 of the mobile device 6 is connected, and the impulse response corresponding to the echo from the base unit 8 is provided by the second largest maximum in the magnitude response. smoothed square of the adaptive filter 42.

However, if the speaker 16 of the mobile device 6 is disconnected (either by default or following an earlier decision to disconnect the speaker 16) while processing is taking place, then the impulse response corresponding to the base unit 8 will be provided by the global maximum in the smoothed square magnitude response of the adaptive filter 42. This may be a more robust solution to the problem. In this case, the mobile device 6 still receives the far-end signal data through the wireless connection between the mobile device 6 and the base unit 10 (or already has the audio file stored in its memory 21), and use this signal or file in the processing steps as described above. The position of the global maximum is converted to a distance without the reference point derived from the echo path from the speaker 16 using the first filter-bypass as a reference point instead. This is acceptable since the speaker 16 of the mobile device 6 is so close to the microphone 18 that the first local peak is in any case normally located in the first few adaptive filter coefficients. Also, when the speaker 16 is disconnected, the power measurement with Th2 cannot be used, but the distance and dispersive measurement can still be used.

ES 2 644 529 T3

In either case, (ie if the speaker 16 and associated components are connected or disconnected), the processing can run adaptively in the background during a call as long as the microphone 18 on the mobile device 6 is turned on.

In the case where the output from speaker 28 is generated from the stored audio file, the above processing differs in that encoder 46 and decoder 48 are not required, so Q (ro) = 0 and Λ (ω ) = 1.

In this implementation of the first embodiment, when a call is initiated, a trigger signal can be sent from the mobile device 6 to the base unit 8 causing the base unit 8 to initiate the call and play the pre-recorded audio file from the memory 31. At the same time, the audio file is extracted from memory 21 on mobile device 6 and fed into the adaptive filter (in this implementation, again, it is not always required to output the audio file through speaker 16 of the device mobile 6, as in the previously described embodiment). Broadcasting the audio file in this manner provides the most efficient implementation of the first embodiment of the invention, in terms of the amount of signaling required between mobile device 6 and base unit 8 to set up the call and the processing required to determine proximity or acoustic quality.

It will also be appreciated that it is possible that the far-end signal is provided directly to the mobile device 6 as well as the base unit 8, in the case where the processing is similar to the stored audio file in the embodiment described in the previous paragraph. .

Although this first embodiment has been described with reference to the analysis of an acoustic signal received from the speaker 28 in the base unit 8 in the microphone 18 of the mobile device 6 (the acoustic signal either from a far-end source or a file of stored audio), along with the acoustic signal received from speaker 16 on mobile device 6 (when speaker 16 is active), It will be appreciated that the distance or acoustic quality between the mobile device 6 and the base unit 8 could be determined instead by analyzing the acoustic signal received at the microphone 18 of the mobile device 6 from the base unit 8 and the corresponding emitted audio signal. by speaker 28 on base unit 8 and received by microphone 30. As will be appreciated by those skilled in the art, the processing required in these implementations will generally be similar to that described in the preceding paragraphs.

In a still further modification to the first embodiment, the distance between the mobile device 6 and the base unit 8 could be determined by analyzing an acoustic signal received at the microphone 30 in the base unit 8 that has been emitted by the speaker 16 in the mobile device 6 and optionally along with the same acoustic signal received at microphone 18 in mobile device 6 from its speaker 16 or the same acoustic signal at microphone 30 in base unit 8 that has been emitted by its speaker 28. Again, as will be appreciated By those skilled in the art, the processing required in these implementations will generally be similar to that described in the preceding paragraphs.

However, these implementations are less preferred since the audio output from the speaker 16 on the mobile device 6 is generally not as loud as the audio output from the base unit 8 (due in part to power and size restrictions, thus such as the fact that the mobile device 6 is likely to be close to the user 4 when in use), and therefore the peaks in the signals received at the base unit 8 are likely not as strong.

A diagram illustrating the processing in a second embodiment of the invention is shown in Figure 8. A flow diagram showing the corresponding method of determining the proximity between a mobile device 6 and a base unit 8 is shown in Figure 9.

In the second embodiment of the invention, an adaptive filter is again combined with a peak detection scheme to determine the distance between user 4 (with mobile device 6) and base unit 8 using two sets of two signals. The second embodiment is based on using the near-end signal as shown in step 115 of Figure 2.

In particular, proximity detection is based on a signal produced by the voice of user 4 of mobile device 6 as detected by microphone 18 on mobile device 6 (the direct field of this signal is associated with a strong coupling between user 4 and the microphone 18 and a short delay) and the same signal produced by the voice of the user 4 in the microphone 18 in the base unit 8 (this signal is associated with a coupling that depends on the distance as well as the environment acoustic between user 4 (ie mobile device 6) and base unit 8). The delay of the second signal relative to the direct field of the first signal is proportional to the distance between the mobile device 6 and the base unit 8.

In this second embodiment, the processing performed in accordance with the invention is preferably performed by the processor 26 in the base unit 8, although it will be appreciated that the processing could instead be performed by the processor 14 in the mobile device 6.

This embodiment of the invention does not make use of the speaker 16 in the mobile device 6, so it does not matter whether the speaker 16 is turned on or not.

ES 2 644 529 T3

Microphone 18 in mobile device 6 detects audio (i.e. voice), S (ro), from user 4 (step 301 in Figure 9) and transmits the received audio signal X (ro) to base unit 8 via a channel Ο (ω) (step 303).

Microphone 30 in base unit 8 also detects the same audio (ie voice), S (n&gt;), from user 4 (step 305).

Processing by processor 26 consists of an adaptive filter 54 and logic unit 56 to calculate the delay between the signals arriving at the microphone 18 of the mobile device 6 and the microphone 30 of the base unit 8.

Similarly, a model 60 of the speech coder 58 used to transmit the voice signal X (n&gt;) from the mobile device 6 to the base station 8 is included to assist with analysis (this speech coder 58 would be implemented by processor 14 in mobile device 6 and set-top box 60 would be implemented by processor 26 in base unit 8). As in the first embodiment, the encoding operation can be modeled as a time-varying quantization and filtering operation, and separated from the channel delay in the wireless channel between the mobile device 6 and the base unit 8 that can be compensated by a acoustic echo canceller, packet loss and other non-linear distortions in the wireless channel are not considered for the purpose of this analysis.

The processing is performed in a similar way to the first embodiment, that is, the coefficients of an adaptive filter are determined to minimize the mean square error of the signals (step 307), the square of the magnitude of the coefficients is optionally smoothed (step 309), the direct fields associated with the user's voice in each of the received signals are identified (step 311) and the distance between the mobile device 6 and the base unit 8 is determined from the identified direct fields (step 313).

In particular, the optimal solution for adaptive filter weights in the sense of MMSE is provided by the Wiener solution,

<img file="ES2644529T3_D0010.tif" />

Ε {χ '(ω) Υ (ω)}

E {xf (o) X „(&lt;»)} where

X_b(to) = Χ (ω) + X_Q(ro) (26) (27)

<img file="ES2644529T3_D0011.tif" />

<img file="ES2644529T3_D0012.tif" />

The numerator and denominator in (26) can be further simplified to

<img file="ES2644529T3_D0013.tif" />

<img file="ES2644529T3_D0014.tif" />

respectively, where G is the factor related to the quantification SNR. The resulting Wiener solution is therefore,

W_opt((D) =

H_b(o) (G + 1) H_m(co) (31)

Consider the free field scenario where the transfer functions Hb (n&gt;) and Νη (ω) are simply lags, that is

<img file="ES2644529T3_D0015.tif" />

<img file="ES2644529T3_D0016.tif" />

ES 2 644 529 T3 where it is assumed that δι&gt; 82 then

W_opt(co) = (G + 1) ^δ2) (34) which then corresponds to an impulse delayed in time t = δι - δ2. If the processing delay for encoder 58 and decoder 60 is compensated for by inserting a delay 62 into the microphone path of the base unit 8, then 82 can be assumed to be zero, and the location of the peak in the impulse response of time domain provides an estimate of the sample distance between user 4 and base unit 8.

The distance between user 4 and base unit 8 can be compared to a threshold distance Th3 (which can be the same as Th1) and, depending on this value

Sw = 1 (i.e. speaker 16 is connected), if D&gt; Th3 (35)

Sw = 0 (i.e. speaker 16 is disconnected), if D &lt;Th3 (36)

As in the first embodiment, the acoustic quality of the voice can be determined from the acoustic impulse response. This can be based on equations 23, 24 and 25 above.

It will be appreciated that this second embodiment can also be implemented directly by calculating the cross-correlation between the time domain signals Yb and Y in Figure 8, ^N^CmW = KÍ ^ Sy (ⁿ) and b (nT) (37) ι \ ι τ τ for the sample delay τ &lt;N. Those skilled in the art are aware that there are numerous other ways to calculate the cross-correlation between two signals.

As indicated above, the method in Figure 2 can operate using an analysis of both the far-end signals and the near-end signals as each signal becomes active, and the proximity decision can be made based on the result. of both measurements. For example, if one embodiment outputs Sw = 1 (ie connect speaker 16) and the other embodiment determines that Sw = 0 (ie disconnect speaker 16), then the final decision in light of this uncertainty may be to keep the speaker 16 on the connected mobile device 6.

In a modification to the first and second embodiments described above, where there are bandwidth restrictions in the wireless channel between the mobile device 6 and the base unit 8, for example preventing the transmission of 8 kHz sampled audio between the device mobile 6 and base unit 8, the acoustic signal can be transformed to a coarser representation on both mobile device 6 and base unit 8. This representation should retain timing cues to determine the distance between the mobile device 6 and the base unit 8, but it can also convey other characteristics related to the properties of the signal to determine the degradation in acoustic quality caused by the acoustic path or distance. between mobile device 6 and base unit 8.

Representations that retain significant timing hints of the signal and still make use of the adaptive filtering operation include smoothing of the signal in the form of rectification and calculation of the signal envelope. This operation can also be followed by an anti-overlap filter and sub-sampling of the signal to reduce the number of samples transmitted per second. In these embodiments, the transmitted voice signal may not be usable for transmission to the far-end response center. Those skilled in the art will be aware of alternative signal representations that retain important time domain characteristics that can be used to determine the distance and acoustic quality between the mobile device 6 and the base unit 8.

The signal representations that can be used to determine the acoustic quality between the mobile device 6 and the base unit 8 are related to the energy in the signals, or higher order statistical moments. These can be transmitted less often and compared on either the mobile device 6 or the base unit

8.

It will be appreciated that the processing performed in accordance with embodiments of the invention can be implemented using hardware or software, or a combination of both.

ES 2 644 529 T3

An alternative approach to RF-based methods for determining the proximity of a mobile device 6 (and a user 4) to a base unit 8 has therefore been described, which, in preferred embodiments, allow a decision to be made regarding to the operation of the speaker 16 of the mobile device 6 during a communication after evaluating the ability of the user 4 to hear and perceive the voice from the speaker 28 in the base unit 8.

Although the invention has been illustrated and described in detail in the drawings and description above, such illustration and description are to be considered illustrative or exemplary and not restrictive; the invention is not limited to the disclosed embodiments.

Variations to the disclosed embodiments may be understood and made by those skilled in the art in practicing the claimed invention, from a study of the drawings, the disclosure, and the appended claims. In the claims, the word "comprising" does not exclude other elements or steps, and the indefinite article "a", "a" does not exclude a plurality. A single processor or other unit can fulfill the functions of several elements indicated in the claims. The mere fact that certain measures are mutually indicated in different dependent claims does not indicate that a combination of these measures cannot be used to their advantage. A computer program can be stored / distributed on a suitable medium, such as optical storage medium or solid state medium supplied together with or as part of other hardware, but it can also be distributed in other ways, such as via the Internet or other systems. telecommunication systems wired or wireless. Any reference signs in the claims should not be construed as limiting the scope.