Wireless sound transmission system and method

A system for providing sound to at least one user, having at least one audio signal source for providing audio signals; a transmission unit with a digital transmitter for wirelessly transmitting the audio signals as data packets; at least one receiver unit with at least one digital receiver for reception of the audio signals from the transmission unit; a mechanism for stimulating the hearing of the user(s) according to audio signals supplied from the receiver unit. The transmission unit transmits each data packet in a separate slot of a TDMA frame at a different frequency in a frequency hopping sequence, in at least some of the slots, the audio signals are transmitted as audio data packets, the same audio packet being transmitted at least twice in the same TDMA frame without expecting acknowledgement, and the TDMA frames being structured for unidirectional broadcast transmission of the audio data packets.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The invention relates to a system and a method for providing sound to at least one user, wherein audio signals from an audio signal source, such as a microphone for capturing a speaker's voice, are transmitted via a wireless link to a receiver unit, such as an audio receiver for a hearing aid, from where the audio signals are supplied to means for stimulating the hearing of the user, such as a hearing aid loudspeaker.

2. Description of Related Art

Presently, in such systems, the wireless audio link usually is an FM (frequency modulation) radio link. According to a typical application of such wireless audio systems, the receiver unit is connected to or integrated into a hearing instrument, such as a hearing aid, with the transmitted audio signals being mixed with audio signals captured by the microphone of the hearing instrument prior to being reproduced by the output transducer of the hearing instrument. The benefit of such systems is that the microphone of the hearing instrument can be supplemented or replaced by a remote microphone which produces audio signals which are transmitted wirelessly to the FM receiver, and thus, to the hearing instrument. In particular, FM systems have been standard equipment for children with hearing loss in educational settings for many years. Their merit lies in the fact that a microphone placed a few centimeters from the mouth of a person speaking receives speech at a much higher level than one placed several feet away. This increase in speech level corresponds to an increase in signal-to-noise ratio (SNR) due to the direct wireless connection to the listener's amplification system. The resulting improvements of signal level and SNR in the listener's ear are recognized as the primary benefits of FM radio systems, as hearing-impaired individuals are at a significant disadvantage when processing signals with a poor acoustical SNR.

A typical application of such wireless audio systems is at school, wherein the teacher uses a wireless microphone for transmitting the captured audio signals via the transmission unit to receiver units worn by the students. Since the receiver units and the respective hearing aids are usually owned by the students, the receiver units may be of different types within a class.

Another typical application of wireless audio systems is the case in which the transmission unit is designed as an assistive listening device. In this case, the transmission unit may include a wireless microphone for capturing ambient sound, in particular from a speaker close to the user, and/or a gateway to an external audio device, such as a mobile phone; here the transmission unit usually only serves to supply wireless audio signals to the receiver unit(s) worn by the user.

Examples of analog wireless FM systems particularly suited for school applications are described, for example, in European Patent Application EP 1 863 320 A1 and International Patent Application Publication WO 2008/138365 A1. According to these systems, the wireless link not only serves to transmit audio signals captured by the wireless microphone, but in addition, also serves to transmit control data obtained from analyzing the audio signals in the transmission unit to the receiver unit(s), with such control data being used in the receiver unit to adjust, for example, the gain applied to the received audio signals according to the prevailing ambient noise and the issue of whether the speaker is presently speaking or not.

In applications where the receiver unit is part of or connected to a hearing aid, transmission is usually carried out by using analog FM technology in the 200 MHz frequency band. In recent systems, the analog FM transmission technology is replaced using digital modulation techniques for audio signal transmission. An example of such digital system is available from the company Comfort Audio AB, 30105 Halmstad, Sweden under the COMFORT DIGISYSTEM® trademark.

A specific example of an analog wireless FM system particularly suited for school applications is described in International Patent Application Publication WO 2008/074350 A1, wherein the system consists of a plurality of transmission units comprising a microphone and a plurality of analog FM receiver units and wherein only one of the transmission units has an analog audio signal transmitter, while each of the transmission units is provided with a digital transceiver in order to realize an assistive digital link for enabling communication between the transmission units. The assistive digital link also serves to transmit audio signals captured by a transmission unit not having the analog transmitter to the transmission unit having the analog transmitter from where the audio signals are transmitted via the analog FM link to the receiver units.

U.S. Patent Application Publication 2002/0183087 A1 relates to a Bluetooth link for a mobile phone using two parallel antennas/transceivers, wherein each data packet is sent once and wherein for a sequence of packets, usually for the next 8 packets, a certain one of the antennas is selected according to previous channel quality measurements as a function of frequency. For each packet of the sequence one of the antennas is selected depending on the respective frequency at which the packet is to be transmitted, wherein the frequency is determined by a frequency hopping sequence.

U.S. Patent Application Publication 2006/0148433 A1 relates to a wireless link between a mobile phone and a base station of the mobile network, wherein two receivers are used in parallel for achieving diversity if the coverage is poor.

Canadian Patent 2 286 522 C relates to a diversity radio reception method, wherein two data packets received in parallel by two receivers are compared and, if they differ from each other, the more reliable one is selected for further processing.

In the publication “Effect of Antenna Placement and Diversity on Vehicular Network Communications” by S. Kaul, K. Ramachandran, P. Shankar, S. Oh, M. Gruteser, I. Seskar, T. Nadeem, 4thAnnual IEEE Communications Society Conference on Sensor, Mesh and Ad Hoc Communications and Networks, 2007, SECON '07, pp. 112-121, a packet level diversity approach is described, wherein in a vehicle-to-vehicle link using roof- and in-vehicle-mounted omni-directional antennas and IEEE 802.11a radios operating in the 5 GHz band a packet level selection diversity scheme using multiple antennas and radios is utilized to improve performance not only in a fading channel but also in line-of-sight conditions. A similar approach is used in “Packet-Level Diversity—From Theory to Practice: An 802.11-based Experimental Investigation” by E. Vergetis et al., MobiCom '06 (see also http://repository.upenn.edu/ese_papers/194), wherein a packet level diversity scheme is applied to a wireless data link between a laptop computer and an access point.

A presentation by S. Shellhammer “SCORT—An Alternative to the Bluetooth SCO Link for Voice Operation in an Interference Environment” document IEEE 802.15-01/145r1, March 2001, and the IEEE P802.15 Working Group for Wireless Personal Area Networks, relates to a proposed alternative for the Bluetooth SCO link for operation in an interference environment, wherein it is proposed to use, in a bi-directional point-to-point link (i.e. full duplex link) for voice transmission, repeated transmission of the same audio packet without involving a receipt acknowledgement by the receiving device.

U.S. Patent Application Publication 2007/0009124 A1 and corresponding U.S. Pat. No. 7,778,432 B2 relate to a wireless network for communication of binaural hearing aids with other devices, such as a mobile phone, using slow frequency hopping, wherein each data packet is transmitted in a separate slot of a TDMA frame, with each slot being associated to a different transmission frequency, wherein the hopping sequence is calculated using the ID of the master device, the slot number and the frame number. A link management package is sent from the master device to the slave devices in the first slot of each frame. The system may be operated in a broadcast mode. Each receiver is turned on only during the transmission during time slots associated to the respective receiver. The system has two acquisition modes for synchronization, with two different handshake protocols. Eight LMP messages are transmitted in every frame during initial acquisition, and one LMP message is transmitted in every frame once a network is established. Handshake, i.e. bi-directional message exchange, is needed both for initial acquisition and acquisition into the established network. During acquisition, only a reduced number of acquisition channels is used, with the frequency hopping scheme being applied to these acquisition channels. The system operates in the 2.4 GHz ISM band. A similar system is known from U.S. Patent Application Publication 2009/0245551 A1 and corresponding U.S. Pat. No. 8,229,146 B2.

U.S. Pat. No. 7,532,610 B2 relates to an adaptive frequency hopping scheme, wherein bad frequencies are empirically excluded from the frequency range used by the frequency hopping algorithm.

International Patent Application Publication WO 2008/135975 A2 relates to a communication network, wherein the receiver wakes up for listening to the preamble of a data packet and goes to sleep again, if no valid preamble is received.

U.S. Patent Application Publication 2006/0067550 A1 relates to a hearing aid system comprising at least three hearing aids between which a wireless communication network is established using the Bluetooth standard, wherein one of the hearing aids is used for receiving signals from another one of the hearing aids, amplifying the signals and forwarding it to the third hearing aid.

U.S. Patent Application Publication US 2007/0086601 A1 relates to a system comprising a transmission unit with a microphone for transmitting a speaker's voice to a plurality of hearing aids via a wireless digital link, which may be unidirectional or bi-directional and which may be used for transmitting both audio data and control data to the hearing aids.

U.S. Pat. No. 7,529,565 B2 relates to a hearing aid comprising a transceiver for communication with an external device, wherein a wireless communication protocol including a transmission protocol, link protocol, extended protocol, data protocol and audio protocol is used. The transmission protocol is adapted to control transceiver operations to provide half duplex communications over a single channel, and the link protocol is adapted to implement a packet transmission process to account for frame collisions on the channel.

European Patent Application EP 1 560 383 A2 relates to a Bluetooth system, wherein the slave device, in a park mode or in a sniff mode, periodically wakes up to listen to transmission from the master and to re-synchronize its clock offset.

SUMMARY OF THE INVENTION

It is an object of the invention to provide for a sound transmission system employing a digital audio link which has relatively low power requirement and which is particularly well-suited for a plurality of receiver units. It is also an object of the invention to provide for a corresponding sound transmission method.

According to the invention, these objects are achieved by a sound transmission system and a sound transmission method as described herein.

The invention is beneficial in that, by using a link protocol wherein the same audio packet is to be transmitted at least twice in the same frame and wherein the frames are structured for unidirectional broadcast transmission of audio data packets, without individually addressing the receiver units (i.e., the frames do not include a receiver address) and without expecting any acknowledgement messages from the receivers, power consumption at the receiver side is kept low and a large number of receiver units can be used in the same network with the transmission unit.

Preferably, the same audio packet is transmitted at least twice in subsequent slots. Preferably, the receiver units use the first verified, i.e., correctly received, copy/version of each data packet as the signal to be supplied to the stimulation means, while not using the audio data of the other copies of the data packet. Usually, in the first slot of each frame, a beacon packet is to be transmitted which contains information for hopping frequency synchronization.

In order to further reduce power consumption, each receiver sleeps at least during times when no data packets are to be expected and wakes up a given guard time before expected arrival of an audio packet different to the previous audio packet. If no start frame delimiter has been received or if the previous audio packet could not be verified, the receiver wakes up a given guard time period before expected arrival of the repetition of the previous audio packet. If a start frame delimiter has been received, the receiver goes to sleep again after a given timeout period after the expected end of transmission of the audio packet; if no start frame delimiter has been received, the receiver goes to sleep again after a given timeout period after the expected end of transmission of the start frame delimiter of the audio packet; thereby further power consumption reduction can be achieved in case of missing packets. Preferably, the start frame delimiter is a 5 bytes code build from the 4 byte unique ID of the network master. This 5 byte code is called the network address, being unique for each network.

In order to achieve further power consumption reduction, each receiver may wake up a given guard time period before expected arrival of the beacon packet of only certain ones of the frames, while sleeping during expected transmission of the beacon packet of the other frames. In particular, the receiver may make up only for beacon packets of frames having a sequence number which fulfills a given condition with regard to the address of the respective receiver unit, so that the transmission unit may send a message to that specific receiver unit by including the message into the beacon packet of a frame having an appropriate sequence number. In addition, each receiver may wake up for the beacon packet of frames having a sequence number fulfilling a certain global condition (for example, every tenth frame), in order to have all receivers periodically listen to the same beacon packet.

These and further objects, features and advantages of the present invention will become apparent from the following detailed description when taken in connection with the accompanying drawings which, for purposes of illustration only, show several embodiments in accordance with the present invention

DETAILED DESCRIPTION OF THE INVENTION

The present invention relates to a system for providing hearing assistance to at least one user, wherein audio signals are transmitted, by using a transmission unit comprising a digital transmitter, from an audio signal source via a wireless digital audio link to at least one receiver unit, from where the audio signals are supplied to means for stimulating the hearing of the user, typically a loudspeaker.

As shown inFIG. 1, the device used on the transmission side may be, for example, a wireless microphone used by a speaker in a room for an audience; an audio transmitter having an integrated or a cable-connected microphone which are used by teachers in a classroom for hearing-impaired pupils/students; an acoustic alarm system, like a door bell, a fire alarm or a baby monitor; an audio or video player; a television device; a telephone device; a gateway to audio sources like a mobile phone, music player; etc. The transmission devices include body-worn devices as well as fixed devices. The devices on the receiver side include headphones, all kinds of hearing aids, ear pieces, such as for prompting devices in studio applications or for covert communication systems, and loudspeaker systems. The receiver devices may be for hearing-impaired persons or for normal-hearing persons. Also on the receiver side, a gateway could be used which relays audio signal received via a digital link to another device comprising the stimulation means.

The system may include a plurality of devices on the transmission side and a plurality of devices on the receiver side, for implementing a network architecture, usually in a master-slave topology.

The transmission unit typically comprises or is connected to a microphone for capturing audio signals, which is typically worn by a user, with the voice of the user being transmitted via the wireless audio link to the receiver unit.

The receiver unit typically is connected to a hearing aid via an audio shoe or is integrated within a hearing aid.

Usually, in addition to the audio signals, control data is transmitted bi-directionally between the transmission unit and the receiver unit. Such control data may include, for example, volume control or a query regarding the status of the receiver unit or the device connected to the receiver unit (for example, battery state and parameter settings).

InFIG. 2, a typical use case is shown schematically, wherein a body-worn transmission unit10comprising a microphone17is used by a teacher11in a classroom for transmitting audio signals corresponding to the teacher's voice via a digital link12to a plurality of receiver units14, which are integrated within or connected to hearing aids16worn by hearing-impaired pupils/students13. The digital link12is also used to exchange control data between the transmission unit10and the receiver units14. Typically, the transmission unit10is used in a broadcast mode, i.e., the same signals are sent to all receiver units14.

Another typical use case is shown inFIG. 3, wherein a transmission device10having an integrated microphone is used for capturing the voice of a person11speaking to a hearing-impaired person13wearing receiver units14connected to or integrated within a hearing aid16. The captured audio signals are transmitted from device10via the digital link12to the receiver units14.

A modification of the use case ofFIG. 3is shown inFIG. 4, wherein the transmission device10is used as a relay for relaying audio signals received from a remote transmission unit110to the receiver units14of the hearing-impaired person13. The remote transmission unit110is worn by a speaker11and comprises a microphone for capturing the voice of the speaker11, thereby acting as a companion microphone.

According to a variant of the embodiments shown inFIGS. 2 to 4, the receiver units14could be designed as a neck-worn device comprising a transmitter for transmitting the received audio signals via an inductive link to an ear-worn device, such as a hearing aid.

The transmission units10,110may comprise an audio input for a connection to an audio device, such as a mobile phone, a FM radio, a music player, a telephone or a TV device, as an external audio signal source.

InFIG. 5, a use case is schematically shown which is similar to that shown inFIG. 2in that a teacher11in a classroom uses a body-worn transmission unit10comprising a microphone17for transmitting audio signals corresponding to the teacher's voice via the digital audio link12to a receiver unit14for reproducing the teacher's voice to students13. However, unlike in the case ofFIG. 2, the receiver unit14is not worn by the respective student13, but rather is connected to or integrated within an audience loudspeaker system18arranged in the classroom.

In each of such use cases, the transmission unit10usually comprises an audio signal processing unit (not shown inFIGS. 2 to 5) for processing the audio signals captured by the microphone prior to being transmitted.

A schematic block diagram of an example of a hearing assistance system according to the invention is shown inFIG. 6. The system comprises a transmission unit10and at least one digital receiver unit14.

The transmission unit10comprises a microphone arrangement17for capturing a speaker's voice, which may be integrated within the housing of the transmission unit10or which may be connected to it via a cable. The transmission unit10also may include an audio signal input19which serves to connect an external audio signal source20, such as a mobile phone, an FM radio, a music player, a telephone or a TV device, to the transmission unit10.

The audio signals captured by the microphone arrangement17and/or the audio signals optionally received from the external audio signal source20are supplied to a digital signal processor (DSP)22which is controlled by a microcontroller24and which acts as an audio signal processing unit which applies, for example, a gain model to the captured audio signals.

In addition, the DSP22may serve to analyze the captured audio signals and to generate control data (control commands) according to the result of the analysis of the captured audio signals. The processed audio signals and the control data/commands are supplied to a digital transmitter28, which is likewise controlled by the microcontroller24.

The digital transmitter28transmits the modulated signals via an antenna36to an antenna arrangement38of the digital receiver unit14, thereby establishing a digital link12. For implementing packet level diversity on the transmitter side, the transmission unit10may comprise a second antenna30which is spaced apart from the (first) antenna36, typically at least one or several wavelengths of the carrier frequency.

In practice, both the digital transmitter28and the digital receiver unit14are designed as transceivers, so that the digital transceiver28can also receive control data and commands sent from the digital receiver unit14.

The transceiver28also may be used for receiving audio signals from an external audio source25, such as a remote microphone used as a companion microphone, via a wireless digital audio link27, with the received audio signals being supplied to the DSP22for retransmission by the transceiver28. Hence, in this case, the transmission unit10serves to relay audio signals from the external audio source to the receiver unit14(see examples ofFIGS. 4 and 11). Alternatively, the transmission unit10may include a separate receiver (not shown in theFIGS. 6 and 7) for receiving the audio signals from the external audio source; in this case the link27would be independent from the link12and thus also could be analog.

The microcontroller24is responsible for management of all transmitter components and may implement the wireless communication protocol, in particular for the digital link12.

The digital receiver unit14comprises or is connected to a loudspeaker42or another means for stimulating a user's hearing. Typically, the receiver unit14is an ear-worn device which is integrated into or connected to a hearing aid comprising the speaker42. The control data transmitted in parallel to the audio signals may serve to control operation of the receiver unit14according to the presently prevailing auditory scene as detected by the DSP22from the audio signal captured by the microphone arrangement17.

InFIG. 7, an example of the audio signal path in the transmission unit10is shown in more detail.

The microphone arrangement17of the transmission unit10comprises two spaced apart microphones17A,17B for capturing audio signals which are supplied to an acoustic beam-former unit44which generates an output signal supplied to a gain model unit46. The output of the beam-former unit44is also supplied to a voice activity detector (VAD) unit48which serves to detect whether the speaker is presently speaking or not and which generates a corresponding status output signal. The output of at least one of the microphones17A,17B is also supplied to an ambient noise estimation unit50which serves to estimate the ambient noise level and which generates a corresponding output signal. The output signals of the units48,50and the processed audio signals from the gain model46are supplied to a unit56which serves to generate a corresponding digital signal comprising the audio signals and the control data which is supplied to the digital transceiver28. The external audio signals optionally received via the audio input19and/or the transceiver28may be supplied to the gain model46.

The units44,46,48,50and56may be functionally realized by the DSP22(see dashed line surrounding these units inFIG. 7).

As already mentioned with regard toFIG. 6, the transmission unit10may comprise a second antenna30which is spaced apart from the first antenna36. Such a dual antenna arrangement may be used to transmit an audio data packet via the first antenna36and to subsequently transmit a repeated copy of the same audio data packet via the second antenna30, as will be explained in more detail with regard toFIGS. 9 and 10.

A more detailed example of the digital receiver unit14is shown inFIG. 8, according to which the antenna arrangement38may comprises two separate antennas38A,38B, wherein the first antenna38A is connected to a first digital receiver61A including a demodulator58A and a buffer59A and the second antenna38B is connected to a second digital receiver61B including a demodulator58B and a buffer59B. The two parallel receivers may be utilized for a applying a packet level diversity scheme to the signals received via the digital link12, as will be explained below in more detail with regard toFIGS. 16 and 17.

The antennas38A,38B receive the signals that are transmitted via the digital link12and the received signals are demodulated by the demodulators58A,58B in the digital radio receivers61A,61B. The demodulated signals are supplied via the buffers59A,59B to a DSP74acting as processing unit which separates the signals into the audio signals and the control data and which is provided for advanced processing, e.g., equalization, of the audio signals according to the information provided by the control data. The processed audio signals, after digital-to-analog conversion, are supplied to a variable gain amplifier162which serves to amplify the audio signals by applying a gain controlled by the control data received via the digital link12. The amplified audio signals are supplied to a hearing aid64. Alternatively, the variable gain amplifier may be realized in the digital domain by using a PWM modulator taking over the role of the D/A-converter and the power amplifier. The receiver unit14also includes a memory76for the DSP74.

Rather than supplying the audio signals amplified by the variable gain amplifier162to the audio input of a hearing aid64, the receiver unit14may include a power amplifier78which may be controlled by a manual volume control80and which supplies power amplified audio signals to a loudspeaker82which may be an ear-worn element integrated within or connected to the receiver unit14. Volume control also could be achieved remotely from the transmission unit10by transmitting corresponding control commands to the receiver unit14.

Alternatively, rather than being ear-worn components, the receiver unit14could be located somewhere in a room in order to supply audio signals to loudspeakers82installed in the same room, whereby a speech enhancement system for an audience can be realized (as indicated by dashed lines inFIG. 8).

Another alternative implementation of the receiver maybe a neck-worn device having a transmitter84for transmitting the received signals via with an magnetic induction link86(analog or digital) to the hearing aid64(as indicated by dotted lines inFIG. 8).

In general, the role of the microcontroller24could also be taken over by the DSP22. Also, signal transmission could be limited to a pure audio signal, without adding control and command data.

Details of the protocol of the digital link12will be discussed by reference toFIGS. 9 to 13. Typical carrier frequencies for the digital link12are 865 MHz, 915 MHz and 2.45 GHz, wherein the latter band is preferred. Examples of the digital modulation scheme are PSK/FSK, ASK or combined amplitude and phase modulations such as QPSK, and variations thereof (for example GFSK).

The preferred codec used for encoding the audio data is ADPCM (Adaptive Differential Pulse-Code Modulation).

In addition, packet loss concealment (PLC) may be used in the receiver unit. PLC is a technique which is used to mitigate the impact of lost audio packets in a communication system, wherein typically the previously decoded samples are used to reconstruct the missing signal using techniques such as wave form extrapolation, pitch synchronous period repetition and adaptive muting.

Preferably, data transmission occurs in the form of TDMA (Time Division Multiple Access) frames comprising a plurality of time slots (for example, 10), wherein one data packet may be transmitted in each slot. InFIG. 9, an example is shown wherein the TDMA frame has a length of 4 ms and is divided into 10 time slots of 400 μs, with each data packet having a length of 160 μs.

As will be explained by reference toFIGS. 14 and 15below, preferably a slow frequency hopping scheme is used, wherein each slot is transmitted at a different frequency according to a frequency hopping sequence calculated by a given algorithm in the same manner by the transmitter unit10and the receiver units14, wherein the frequency sequence is a pseudo-random sequence depending on the number of the present TDMA frame (sequence number), a constant odd number defining the hopping sequence (hopping sequence ID) and the frequency of the last slot of the previous frame.

The first slot of each TDMA frame (slot0inFIG. 9) is allocated to the periodic transmission of a beacon packet which contains the sequence number numbering the TDMA frame and other data necessary for synchronizing the network, such as information relevant for the audio stream, such as description of the encoding format, description of the audio content, gain parameter, surrounding noise level, etc., information relevant for multi-talker network operation, and optionally control data for all or a specific one of the receiver units.

The second slot (slot1inFIG. 9) may be allocated to the reception of response data from slave devices (usually the receiver units) of the network, whereby the slave devices can respond to requests from the master device through the beacon packet. At least some of the other slots are allocated to the transmission of audio data packets, wherein each audio data packet is repeated at least once, typically in subsequent slots. In the example shown inFIGS. 9 and 10, slots3,4and5are used for three-fold transmission of a single audio data packet. The master device does not expect any acknowledgement from the slaves devices (receiver units), i.e., repetition of the audio data packets is performed in any case, irrespective of whether the receiver unit has correctly received the first audio data packet (which, in the example ofFIGS. 9 and 10, is transmitted in slot3) or not. Also, the receiver units are not individually addressed by sending a device ID, i.e., the same signals are sent to all receiver units (broadcast mode).

Rather than allocating separate slots to the beacon packet and the response of the slaves, the beacon packet and the response data may be multiplexed on the same slot, for example, slot0.

The audio data maybe compressed in the transmission unit10prior to being transmitted.

If the transmission unit10comprises two antennas30,36, packet level diversity with regard to the audio data packets may be realized on the transmitter side by transmitting each one of the copies of the same audio data packet alternately via a different one of the antennas30,36. For example, the first copy of the audio data packet (which, in the example ofFIGS. 9 and 10, is transmitted in slot #3, may be transmitted via the antenna36, whereas the second copy (in slot #4) may be transmitted via the antenna30, while the third copy (in slot #5) may be transmitted again via the antenna36. If for example, at the position of the antenna36multi-path fading occurs with regard to the antenna of the receiver unit14, it is unlikely that multi-path fading likewise occurs at the position of the antenna30, so at least one copy will be transmitted/received without fading.

InFIG. 11, an example of a more complex slot allocation scheme is shown, wherein, as in the example ofFIGS. 9 and 10, slot0is allocated to the beacon packet from the master device and slot1is allocated to response data packets. However, in the example ofFIG. 11, each audio data packet is repeated only once and a transmission unit10is used as a relay/gateway between three remote transmission units110A,110B and110C acting as companion microphones and two receiver units14A,14B. Slots2and3, slots4and5and slots6and7are used for transmission of audio data from the first external transmission unit110A, the second external transmission unit110B and the third external transmission unit110C, respectively, towards the relay/gateway transmission unit10, and slots8and9are allocated to transmission of audio data packets from the relay/gateway transmission unit10to the receiver units14A,14B. The beacon packet in slot0is sent from the unit10acting as the master to all slaves, i.e. the units110A,110B,110C,14A and14B. The beacon packet and the response packet can also be time-multiplexed on the same slot0(e.g., even numbered TDMA frames for beacon packets, odd numbered TDMA frames for response packets).

Usually, in a synchronized state, each slave listens only to specific beacon packets (the beacon packets are needed primarily for synchronization), namely those beacon packets for which the sequence number and the ID address of the respective slave device fulfills a certain condition, whereby power can be saved. When the master device wishes to send a message to a specific one of the slave devices, the message is put into the beacon packet of a frame having a sequence number for which the beacon listening condition is fulfilled for the respective slave device. This is illustrated inFIG. 12, wherein the first receiver unit14A listens only to the beacon packets sent by the transmission unit10in the frames number1,5, etc., the second receiver unit14B listens only to the beacon packets sent by the transmission unit10in the frames number2,6, etc., and the third receiver unit14C listens only to the beacon packet sent by the transmission unit10in the frames number3,7, etc.

Periodically, all slave devices listen at the same time to the beacon packet, for example, to every tenth beacon packet (not shown inFIG. 12).

Each audio data packet comprises a start frame delimiter (SFD), audio data and a frame check sequence, such as CRC (Cyclic Redundancy Check) bits. Preferably, the start frame delimiter is a 5 bytes code built from the 4 byte unique ID of the network master. This 5 byte code is called the network address, being unique for each network.

In order to save power, the receivers61A,61B in the receiver unit14are operated in a duty cycling mode, wherein each receiver wakes up shortly before the expected arrival of an audio packet. If the receiver is able to verify (by using the CRC at the end of the data packet), the receiver goes to sleep until shortly before the expected arrival of a new audio data packet (the receiver sleeps during the repetitions of the same audio data packet), which, in the example ofFIGS. 9 and 10, would be the first audio data packet in the next frame. If the receiver determines, by using the CRC, that the audio data packet has not been correctly received, the receiver switches to the next frequency in the hopping sequence and waits for the repetition of the same audio data packet (in the example ofFIGS. 9 and 10, the receiver then would listen to slot4as shown inFIG. 10, wherein in the third frame transmission of the packet in slot3fails).

In order to further reduce power consumption of the receiver, the receiver goes to sleep shortly after the expected end of the SFD, if the receiver determines, from the missing SFD, that the packet is missing or has been lost. The receiver then will wake up again shortly before the expected arrival of the next audio data packet (i.e., the copy/repetition of the missing packet).

An example of duty cycling operation of the receiver is shown inFIG. 13, wherein the duration of each data packet is 160 μs and wherein the guard time (i.e., the time period by which the receiver wakes up earlier than the expected arrival time of the audio packet) is 20 μs and the timeout period (i.e., the time period for which the receiver waits after the expected end of transmission of the SFD and CRC, respectively) likewise is 20 μs. It can be seen fromFIG. 13that, by sending the receiver to sleep after timing out of SFD transmission (when no SFD has been received), the power consumption can be reduced to about half of the value when the receiver is sent to sleep after timeout of CRC transmission.

As already mentioned above, a pseudo-random frequency hopping scheme is used for data transmission. As illustrated inFIG. 14, for calculating the frequency-hopping sequence, an algorithm is used which has, as input parameters, the frequency fpused for the last slot of the previous frame, the hopping sequence ID (HSID) and the sequence number s of the present frame. The algorithm uses a linear congruent generator (LCG) which outputs the frequency for each slot of the frame based on these three input parameters. An example of the computation of fi,iε{0;9}, based on the three parameters HSID, s and fpare given below:

The information necessary to compute the frequency-hopping sequence for the present frame is transmitted in the beacon packet in the first slot of the frame from the master device to the slave devices. The hopping sequence ID is not included in the beacon packet, but rather is transmitted in a pairing phase to the slave devices and is stored in each slave device. Once synchronized to the master device, the slave devices increment the sequence number automatically to calculate the frequency at which the beacon packet of the next frame is to be received.

The Hopping Sequence ID is chosen as an odd number between 1 and 65535 . . . . This number is chosen randomly by the network master (relay unit15) and transmitted to the network slaves (transmission units10and receiver units14) during pairing. This odd number is used as the additive term of the LCG. By selecting the hopping sequence ID randomly, it is provided that the hopping sequence is likely to be unique to the present network, so that there is only low cross-correlation with the hopping sequence of another network which may exist, for example, in the same building. In the unlikely event that two networks select the same hopping sequence ID and disturb each other, a new pairing process in one of the networks is likely to result in a different hopping sequence ID. The use of the frequency of the last slot of the previous frame in the hopping sequence algorithm ensures that there is always a minimum distance between two subsequent slots, namely also between the last slot of the previous frame and the first slot of the present frame.

Preferably, the frequency-hopping scheme is an adaptive frequency-hopping scheme, wherein packet error rate measurements are made for the used frequencies and wherein the master device may decide, based on such measurements, that a sub-set of the available frequencies should be declared as “bad frequencies” and should not be used any longer. If then, the frequency computation algorithm selects one of the bad frequencies, a frequency is pseudo-randomly selected instead, from a set of frequencies composed of all “good frequencies” at the exception of the good frequency used in the preceding slot. Removing the frequency used in the preceding slot from the set of potential replacement frequencies presents the advantage of avoiding the possibility of using the same frequency twice in consecutive slots.

FIG. 15illustrates how synchronization between the master device (for example, the transmission unit10) and the slave devices (for example, one of the receiver units14) may be achieved.

The synchronization is passive in the sense that there is no feedback from the slave device to the master device during synchronization. Usually, the master device, e.g., the transmission unit10, does not distinguish whether a certain one of the slaves, e.g., the receiver units14, is in still a synchronization mode or already in a synchronized mode, so that the transmission operation of the master is always the same, i.e., the same algorithm for determining the hopping sequences is used and the same protocol is used, e.g., beacon packet in the first slot, audio data packets in some of the other slots (as long as audio signals are generated in/supplied to the transmission unit; the audio data packets are not shown inFIG. 15).

Thus, the master device transmits a beacon packet in regular intervals, namely in the first slot of each TDMA frame (according to the example, a beacon packet is sent every 4 ms). The frequency at which the respective beacon packet is sent is calculated according to the same pseudo-random hopping-sequence algorithm which is used for transmitting audio packets in the synchronized state. The hopping sequence is long in the sense that it is much longer/larger than the number of frequency channels (for example, a sequence of the length 100 is likely to show a bad correlation with another sequence of the length 100, depending on the time shift). The slave device listens periodically for the first beacon packet for synchronization, i.e., it is operated in a duty cycling mode. The listening time period is longer than the duration of the beacon packet. Each listening period is done at a different frequency; for example, the first listening period may at the lowest frequency of the available band (i.e., the receiver listens in the lowest one of the frequency channels), and then the listening frequency is increased for each subsequent listening period (thereby going systematically through all frequency channels). After each listening period, the receiver goes back to sleep. The periodicity of the listening periods is chosen close to the beacon packet transmission periodicity (i.e., the frame length), but it is not exactly equal, in order to have a drift between the beacon packet transmission phase and the listening phase. Due to this drift, the listening phase is periodically in phase with the transmission of the beacon packet for a certain duration. When the beacon packet is transmitted at the same frequency as the one used presently for listening, synchronization is achieved and the receiver switches into the synchronized mode/state, wherein it can calculate the hopping sequence presently used by the transmission unit from the information included in the received beacon packet (i.e., the frame sequence number) and the hopping sequence ID stored in the receiver unit from the pairing phase. A more detailed explanation of this synchronization method is given below.

When a receiver is in the synchronization phase, it listens periodically with period TListenPeriodfor a duration TListenDurationat a given frequency and then goes back to sleep. The frequency is changed for each listening phase starting with frequency number0, and incrementing up to e.g., frequency39. The beacon is transmitted on any of the 40 frequencies, following the pseudo-random frequency selection.

The period TListenPeriodis chosen to be close to the beacon transmission period TBeaconPeriodbut not to be exactly equal. The difference ΔT=|TListenPeriod−TBeaconPeriodcauses a drift between the beacon packet transmission phase and the listening phase. Due to this drift, the listening phase is periodically in phase with the transmission of the beacon packet for a certain duration. If the beacon packet is transmitted at the same frequency as the one used for listening, synchronization is achieved. This mechanism is illustrated inFIG. 15.

The values of parameters TListenPeriod, TListenDurationare to be chosen based on the beacon packet period TBeaconPeriodand on the beacon packet duration TBeaconDuration, as a trade-off between the synchronization delay and the synchronization power consumption.

With TListenPeriod=TBeaconPeriod(1+θ), and ΔT=θTBeaconPeriodis the shift in phase of the listening activity for every transmission of the beacon packet.

TListenDurationmust be larger than TBeaconDurationsuch that it is possible to receive a beacon packet. An additional margin ΔT is required such that the listen window is open for the duration of the beacon packet transmission, given the fact that the listen window is drifting compared to the transmission window. A larger margin than ΔT gives the opportunity for the reception of more than one beacon packet in a given transmission window.

The time interval between two in-phase periods will be

When the transmission and listening intervals are in phase, there will be enough time for a limited number of transmission trials, until the windows are out of phase again. The number of possible trials is given by

where └ ┘ means rounded to the nearest integer in a direction towards zero.

The average synchronization delay can then be computed with

T_Synchronization=TInPhasePeriod/NTrialsInPhase1/NChannels.
when NChannels=40, θ=0.01, TBeaconPeriod=4 ms, TListenDuration=600 μs,TSynchronization=1.6 s and the duty cycle will be, in this case,

A further refinement can be obtained if a transmission unit has two radios, i.e., transmitters/transceivers. In such case, the two radios may be used to transmit the beacon messages in an inter-leaved manner, or in parallel and at different frequencies. This method would reduce the synchronization time required at the receiver side.

As illustrated inFIG. 16, by using two spaced-apart antennas38A,38B multi-path fading resulting from destructive interference between several copies of the same signal travelling due to multiple reflections along different signal paths with different lengths (for example, direct signal and signal reflected once), can be mitigated, since the interference conditions are different at different positions, i.e., if destructive interference occurs at the position of one of the antennas, it is likely that no destructive interference occurs at the position of the other antenna. In other words, if the two antennas are sufficiently spaced apart, the fading events are uncorrelated on both antennas.

The present invention may utilize this effect by applying a packet level diversity scheme in the receiver unit. When a data packet has been received by the receiver58A, it will be verified by using the CRC and it will be buffered in the buffer59A. In addition, an interrupt request is sent from the receiver59A to the processing unit74, in order to indicate that a packet has been received. The other receiver58B acts in parallel accordingly: when it receives a data packet, it verifies the data packet and buffers it in the buffer59B and sends an interrupt request to the processing unit74.

When the processing unit74receives such an interrupt request, it reads the data packet from one of the two buffers59A,59B (usually, there is a default setting as to from which one of the buffers the processing unit74will try to read the data packet first) and flushes the other one of the buffers59A,59B, if the data packet was obtained correctly (rather than using interrupt requests, the respective buffer59A,59B could be checked at the end of the last reception slot; i.e., the receivers could operated via polling rather than via interrupts). However, if it is not possible to read the data packet from the default one of the buffers (usually because the respective antenna38A,38B suffered from severe multi-path fading at the reception time), the processing unit74tries to read the data packet from the other one of the buffers and, if it is successful in reading the data packet, it flushes the buffer of the other.

An example of this method is illustrated inFIG. 17, wherein it is assumed that the third transmission of the data packet “A” from the transmission unit10fails at the antenna38A allocated to the receiver58A so that, in this case, the processing unit74reads the data packet from the buffer59B of the receiver58B rather than from the buffer59A of the receiver58A (which, in the example, is the default receiver). Typically, such packet level diversity is applied not only to the audio data packets, but also to the other data packets, such as the beacon packet.

However, it is noted that such packet level diversity is not applicable to ear level receiver units since, due to the small size of ear level receiver units, there is usually not enough space for the required spatial separation of the two antennas required for the above-described packet level diversity scheme.

While various embodiments in accordance with the present invention have been shown and described, it is understood that the invention is not limited thereto, and is susceptible to numerous changes and modifications as known to those skilled in the art. Therefore, this invention is not limited to the details shown and described herein, and includes all such changes and modifications as encompassed by the scope of the appended claims.