GB2416955A

GB2416955A - Conference calls in mobile networks

Info

Publication number: GB2416955A
Application number: GB0416846A
Authority: GB
Inventors: David Ashbrook; David Pollington
Original assignee: Vodafone Group PLC
Current assignee: Vodafone Group PLC
Priority date: 2004-07-28
Filing date: 2004-07-28
Publication date: 2006-02-08
Anticipated expiration: 2024-07-28
Also published as: GB0416846D0; GB2416955B

Abstract

Conference calls in a GSM network are provided by processing incoming speech from participants 10-13 in a conference bridge 18 including a voice activated detector 19, which can distinguish silent periods. Speech components are encoded by a multi-channel variable bitrate coder and sent through a multi-channel link 20 to a receiving terminal 14. Non-speech components are coded as a silence signal to reduce transmission bandwidth requirements. At receiving terminal 14 voice components are passed to a spatial processor, which distinguishes between source participants by the channel containing the coded speech. Each participant's voice is located at a spatially separated location, to aid identification. The spatial processor may alternatively be located within the network, which sends stereo signals to the terminal 14.

Description

24 1 6955

CONFERENCE CALLS IN MOBILE NETWORKS

The invention relates to conference calls in mobile networks and more particularly to conference calls in telecommunication systems having three or more transmitter/receiver units, at least one of which is a mobile or cellular transmitter/receiver unit registered with a mobile or cellular telecommunications network.

Conference calling In such telecommunication systems allows multiple callers in different locations to interact In a single voice call. A problem with such conference calls is differentiating between participants using a single mono audio output. This is exacerbated by network bandwidth limitations in such conference calls as well as signal compression that results in many of the subtle clues used by a person to recognise a person's voice being lost.

It has been suggested in US2003/0044002 At to create spatially resolved audio signals for a listener as representative of one or more callers. In this way, analogue audio signals appear to emanate from different spatial locations around the listener. It is a problem with such a method that significant bandwidth is required to relay signals simultaneously from all participants in the conference call.

According to the invention, there is provided a method of performing a conference call in a telecommunication system having three or more transmitter/receiver units, at least one of which is a mobile transmitter/receiver unit registered with a mobile telecommunications network, the method including, receiving in the network a signal from an originating unit, encoding any voice component within said signal, generating and encoding a silence signal on detection of any non-voice component within said signal, producing from said encoded speech signal a spatially resolved audio signal characteristic of said originating unit, and producing at a receiving unit from said spatially resolved audio signal a spatially resolved analogue audio signal.

By producing a silence signal in the absence of a voice component, the bandwidth needed to transmit the signals can be reduced.

The following is a more details description of some embodiments of the invention, by way of example, reference being made to the accompany drawings in which: Figure I is a schematic view of a mobile telecommunications network in a conference call configuration with mobile transmitting/receiving units, each unit having a spatial processor, Figure 2 is a schematic view of a mobile unit of the kind shown in Figure 1 showing the provision of an identifier on a receiving unit of the mobile network of Figure 1, Figure 3 is a similar view to Figure 1 but showing the spatial processor incorporated in the mobile telecommunications network.

Referring first to Figure 1, the network is a GSM PS network including a plurality of mobile transmitter/receiver units of which five 10, 11, 12, 13, 14 are shown in Figure 1. The network includes a mobile switching centre (MSC) 15, a base station controller (BSC) 16 and a plurality of base transceiver stations (BTS), one of which is shown at 17.

The MSC 15 includes a conference bridge 18 that permits the plurality of mobile units 10, l 1, 12, 13, 14, to communicate in a single voice call. Each mobile unit 10, 1 1, 12, 13, 14 transmits to the MSC 15 via a BTS and BSC, a signal that includes a voice component and a non-voice component. The non-voice component is transmitted when the user of a unit 10, 11, 12, 13, 14 is not speaking and will represent

background noise.

At the MSC 15, the signal is fed to a voice activity detection system (VAD) 19. Such systems are known and determine the present or absence of a speech component in a given signal, especially the beginnings and ending of voice segments. Such a VAD 19 uses algorithms to detect such speech components. The VAD l9 passes a speech component of the signal to multi-channel variable bitrate audio/speech coder (not shown) of known type. Such a coder produces an optimal bitstream for conveying the speech component to a receiving unit I 0, 1 l, 1 2,1 3,14 in one of the channels.

When the VAD l9 detects the absence of a speech component in the signal, the VAD signals to the coder to provide a minimal number of bits to the channel so indicating silence or no speech. The coded signals are then fed via a BSC 16 and a BTS 17 to a receiving unit 14. Each channel includes a tagging identifier that identifies each transmitting unit and the channels may be fed as separate real-time protocol (RTP) 20 data flows. Accordingly, in general, one channel will be carrying the speech of the current speaker and "silent" channels for each of the other speakers.

At the receiving unit 14, the channels are passed to a spatial processor 21. The spatial processor 21 uses the tagging Identifier and the coded speech signal to generate a analogue spatially resolved audio signal characteristic of the transmitter unit 10, 11, 12, 13 from which the signal emanates. If the receiving unit 14 has stereo loud speakers then the spatial orientation of the signals will be mainly located in front of the person using the receiving unit 14 as seen in Figure 1, assuming the unit is held in front of the person. Better spatial separation in 3D around the person (that is, in a 3D space around the person) may be obtained using, for example, stereo headphones 26.

Such spatial processors 21 are known and will not be described in further detail.

The logging identifier may be used to produce a relative indication of the transmitting unit 10, 11, 12, 13 at the receiving unit 14. As seen in Figure 2, this may take the forth of a number 22 displayed on a screen 23 of the receiving unit 14, a different number Identifying each transmitting unit 10, 11, 12, 13. The indication is relative since the numeral does not link the numeral to a person.

lO It would be possible for the signal from a transmitting unit 10, 11, 12, 13 to contain metadata that both identifies the transmitting unit lO, 11, 12, 13 and also contains data identifying the person transmitting such as a name and/or a digital image to provide an absolute indication. Alternatively, the transmitting unit 10, 11, 12, 13 could send an MMS visiting card at the start of the conference call that would be used by the receiving unit 14 to identify the caller. A further possibility is for the caller to be identified from information held in the address book of a receiving unit 14, assuming the called information is stored in the address book before the start of the conference call.

This information could also be used by the user of the receiving unit 14 to reposition the spatial orientation of the transmitters either angularly around the receiving person and/or radially relative to the person. l

The analogue spatial audio signal produced at the receiving unit 14 may be a 3-D signal lying in an imaginary sphere around the receiving person. Alternatively, this may be a 2-D signals lying in an imaginary circle around the receiving person. It has been found that, in general 2-D signals in front of the receiving person are the most easily differentiated possible since the human body is best able to distinguish sounds over the front hemisphere of the body. The use of a 2-D or 3-D signal allows the user to distinguish more easily between participants and provides a more natural listening environment so reducing listening fatigue. It would also allow louder speakers to be moved further away and to move softer speakers closer. It gives the potential for a large number of conference participants due to the extended sound field and, since no background noise is received, increases the illegibility.

It would be possible to use only stereo panning in which the audio signals were chosen from a left ear signal, a right ear signal and a left and right ear signal and possibly mid-left and mid-right signals.

The advantage of using a VAD 19 to process the transmitted signal is that it allows the bandwidth to be reduced smce the channels transmitting "silence" data use little bandwidth. This also has the advantage of reducing implementation costs. The system can be implemented using standard protocols. The bandwidth is also reduced by the use of the multichannel variable bitratc coder.

In the embodiment described above, the spatial processor 21 is provided in the units 10, 11, 12, 13, 14. In an alternative embodiment, shown in Figure 3, the spatial processor is provided in the MSC 15. In this Figure, parts common with Figure 1 have the same reference numerals and will not be described in detail. In this embodiment, the signal transmitted from the BTS 17 to the receiving unit 14 is a stereo signal. This has the advantage of obviating the need for a spatial processor 21 in each unit 10, l l, 12, 13, 14 but requires greater bandwidth to transmit the stereo signal. In addition, this embodiment would not allow the receiving unit user to re position the analogue spatial audio signals. Further, it requires the use of generic music coders to preserve the pre-spatialised information.

It is possible that the systems described above could be implemented in a CS network or using a CS uplink and a PS downlink although this may reduce the options 1 5 available.

In the embodiments described each unit is a mobile telecommunications device.

However, it should be appreciated that only one of the units need be a mobile telecommunications device; the other units may be fixed terminals for example terminals connected to the PSTN by a cable.

The system may also be implemented in a UMTS (3G network) or any other mobile or cellular telecommunications network- such as a 2G (GSM) or VolP network - in which at least some of the terminals or units communicate wirelessly with the network.

Claims

1. A method of performing a conference call in a telecommunication system having three or more transmitter/receiver units, at least one of which is a mobile transmitter/receiver unit registered with a mobile teleeonNnunieations network, the method including: receiving in the network a signal from an originating unit, encoding any voice component within said signal, generating and encoding a silence signal on detection of any non-voice component within said signal, producing from said encoded speech signal a spatially resolved audio signal characteristic of said originating unit, and producing at a receiving unit from said spatially resolved audio signal a spatially resolved analogue audio signal

2. A method according to claim 1 wherein the signal from the originating unit is received by a voice activity detection system to determine the presence of a voice component in said signal.

3. A method according to claim I or claim 2 wherein the encoded speech signal and the silence signal are coded by a multi-channel variable bitrate coder.

4. A method according to any one of claims 1 to 3 wherein the spatially resolved audio signal is generated by stereo-panning.

A method according to any one of claims I to 3 wherein the spatially resolved audio signal is generated by 2D or 3D virtualised audio.

6. A method according to any one of claims l to 5 and including transmitting to the receiving unit a signal identifying the originating unit and producing at the receiving unit a visual indication of the identity of the originating unit.

7. A method according to claim 6 wherein the visual indication is a relative identifier.

8. A method of according to claim 7 wherein the relative identifier is a numerical 1 5 indication.

9. A method according to claim 6 wherein the visual indication is an absolute identifier.

lO. A method according to claim 9 wherein the absolute identifier is taken from information stored In an address book of the receiving unit.

A method according to claim 9 wherein the absolute identifier is derived from information transmitted from the originating unit.

12. A method according to claim 10 or claim 11 wherein the absolute identifier comprises an MMS message transmitted from the originating unit.

13. A method according to claim 11 or claim 12 wherein the information comprises a digital image.

14. A method according to any one of claims 1 to 12 and comprising producing said spatially resolved audio signal in the network and transmitting the signal to the receiving unit.

15. A method according to any one of claims I to 12 transmitting the encoded voice component and the silence signal to the receiving unit and generating the spatially resolved audio signal in the receiving unit.

16. A method according to any one of claims 1 to 15 wherein the network Is a PS network.

17. A method according to any one of claims 1 to 16 and comprising repositioning the spatial position of an analogue voice signal at a receiving unit.

18. A method according to claim 17 when dependent on any one of claims 9 to 13 wherein absolute identifier is used to re-position an analogue voice signal.

l9. A method according to any one of claims 1 to 18 wherein the spatially resolved analogue audio signal is produced by a stereo sound source at the receiving unit.

20. A method of performing a conference call in a telecommunications system having at least three mobile transmitter/receiver units and a fixed network substantially as hereinbefore described with reference to the accompanying drawings.