EP1078523A1 - Speakerphone is also module for video conferencing system - Google Patents

Speakerphone is also module for video conferencing system

Info

Publication number
EP1078523A1
EP1078523A1 EP20000912482 EP00912482A EP1078523A1 EP 1078523 A1 EP1078523 A1 EP 1078523A1 EP 20000912482 EP20000912482 EP 20000912482 EP 00912482 A EP00912482 A EP 00912482A EP 1078523 A1 EP1078523 A1 EP 1078523A1
Authority
EP
European Patent Office
Prior art keywords
system
apparatus
data
network
video conferencing
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP20000912482
Other languages
German (de)
French (fr)
Inventor
Daniel A. Ash
Norberto Pellicci
Weizhong Yang
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Koninklijke Philips NV
Original Assignee
Koninklijke Philips NV
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Priority to US264058 priority Critical
Priority to US26405899A priority
Application filed by Koninklijke Philips NV filed Critical Koninklijke Philips NV
Priority to PCT/EP2000/001344 priority patent/WO2000054502A1/en
Publication of EP1078523A1 publication Critical patent/EP1078523A1/en
Application status is Withdrawn legal-status Critical

Links

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers; Analogous equipment at exchanges
    • H04M1/60Substation equipment, e.g. for use by subscribers; Analogous equipment at exchanges including speech amplifiers
    • H04M1/6033Substation equipment, e.g. for use by subscribers; Analogous equipment at exchanges including speech amplifiers for providing handsfree use or a loudspeaker mode in telephone sets
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers; Analogous equipment at exchanges
    • H04M1/72Substation extension arrangements; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selecting
    • H04M1/725Cordless telephones
    • H04M1/737Cordless telephones characterised by transmission of electromagnetic waves other than radio waves, e.g. infra-red
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00Television systems
    • H04N7/14Systems for two-way working
    • H04N7/141Systems for two-way working between two video terminals, e.g. videophone
    • H04N7/142Constructional details of the terminal equipment, e.g. arrangements of the camera and the display
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00Television systems
    • H04N7/14Systems for two-way working
    • H04N7/141Systems for two-way working between two video terminals, e.g. videophone
    • H04N7/148Interfacing a video terminal to a particular transmission medium, e.g. ISDN

Abstract

A communication system has a speakerphone and a physically separate sub-system interfacing the speakerphone to a network. The sub-system comprises either a wallmount device for interfacing to a telephone network or a sub-system for video conferencing. The speakerphone and the sub-system communicate through a digital proprietary protocol, e.g., time-division multiplexed. By physically separating the system into a speakerphone and a sub-system that interfaces to the network, the same speakerphone can be used for audio conferencing when used with the appropriate wallmount, and also as audio accessory to a video conferencing system that handles the same protocol.

Description

Speakerphone is also module for video conferencing system.

The invention relates to a system and method for audio communication. The invention relates in particular to a speakerphone and to video conferencing system.

Several technologies are currently available for conferencing, among which are speakeφhones and video conferencing. The term "speakeφhone" refers generally to an audio communication telephone device that allows hands-free operation. A speakeφhone for audio conferencing is equipped with a high-fidelity speaker, and with noise-canceling and echo- canceling systems accommodated in an acoustically optimized enclosure. Accordingly, speakeφhones are much more expensive than handsets. Video conferencing communicates audio and video, compressed or full motion video, via, e.g., satellite, cable, ISDN or phone line and in real time. For more background and detailed information on video conferencing, see, e.g., U.S. patent 5,793,415; U.S. patent 5,799,190; U.S. patent 5,515,373; U.S. patent 5,500,880 (Philips Electronics), all incoφorated herein by reference.

The inventors have realized that the speakeφhones currently on the market have some drawbacks. For example, the known speakeφhones are typically designed for a specific network interface, e.g., POTS, ISDN, PSTN, LAN, Ethernet, DSL, etc. Each speakeφhone currently commercially available is solely used for audio communication based on a single protocol, and other protocol interfaces or data transfer mechanisms are not provided. As a result, they cannot be upgraded: if the end-user wishes to go from, e.g., a PSTN-based system to an ISDN-based system, the only option seems to be to purchase an entirely new system. As another example, the inventors have realized that some companies market different audio systems for speakeφhone and video conferencing, although the audio functionality is largely the same in both technologies. It is therefore an object of the invention to provide a more user-friendly communication system with the emphasis on modularity and flexibility, to the advantage of both the end-user and the manufacturer.

To this end, the invention provides an end-user communication system that has a first apparatus and a second apparatus. The first apparatus comprises a baseline unit with a user-interface for audio communication via a communication network. The second apparatus comprises an interface sub-system for interfacing the baseline unit to the network. The subsystem and the baseline unit are capable of communicating using a first protocol for digital transmission based on, for example, time-division multiplexing. The sub-system is capable of communicating via the network using a second protocol independent of the first protocol. The first and second apparatus are physically separate devices or at least, when physically attached to each other, detachable from each other, e.g., for enabling use of the first apparatus with another version of the second apparatus that has been upgraded or that is suitable for another type of communication network. In a speakeφhone configuration of the communication system, the baseline unit has a speakeφhone functionality, and the interface sub-system interfaces the speakeφhone to a telephone network. In this set-up the interface sub-system is typically configured as a wallmount device.

In a video conferencing configuration, the combination of the baseline unit and the sub-system form parts of a video conferencing system wherein the baseline unit provides the user-interface for audio communication, and wherein the interface sub-system not only enables audio processing, but also carries out the processing for the video part of the communication. Preferably, the user interface of the baseline unit enables user-control of the video conferencing system. Preferably, the baseline unit is capable of receiving and transmitting data to further equipment other than the second apparatus. Assume that the first protocol is based on, e.g., time-division multiplexing (TDM) wherein each time slot represents a respective channel. Assigning audio, control commands and data from/to other equipment to specific time slots allows then to identify the different information types (audio, commands, and data), thus rendering the system highly versatile and truly multimedia. The data communication with the further equipment preferably uses wireless communication in order to minimize the number of cables between the first and second apparatus. This is especially relevant to table-top usage of the baseline unit in both speakeφhone and video conferencing configuration of the communication system in the invention.

The invention is based on the insight that separating the user-interface functionality from the network interface functionality, i.e., accommodating them in different apparatus, has several advantages. A first asset is that the baseline unit can now be used in any network as long as the interface sub-system is compatible with the latter. Since the most expensive components (e.g., microphones, speakers, means for echo cancellation) in speakeφhone equipment are housed in the baseline unit of the invention, this separation enables the user to upgrade or modify his/her communication system by simply adjusting or replacing the interface sub-system. Replacing or modifying the interface unit is typically much less expensive than purchasing a new baseline unit. For the manufacturer and retailer of speakeφhones or video conferencing systems, the physical separation of audio user-interface from network interface has the advantage that a single version of the baseline unit serves the needs of all customers. Differentiating takes place in the network interface, i.e., in the subsystem, which is the least expensive part. Accordingly, this aspect of the invention takes into account environmental ("green") considerations. A second asset is that the baseline unit can be used as a stand-alone speakeφhone or as an audio accessory to a video conferencing system if the latter is provided with a suitable interface to the baseline unit. The baseline unit of the invention preferably has a telephone user-interface so that the video conference call can be made from the baseline unit in a way similar to making a telephone call. A third advantage is that the digital first protocol between the baseline unit and the interface is independent of the second protocol between the interface and the network. This independence can be used to have data input/output (e.g., through an IrDA port) at the baseline unit if the latter is provided with an infrared transceiver and the necessary software. It is then possible to communicate data received from, e.g., a notebook computer. In other words, by making the baseline unit independent of the network through the interface unit, the baseline unit's hardware and software and the first protocol can be optimized for data acquisition, audio signal processing and control commands for video conferencing. A fourth advantage is that the systems at the sites of different end-users only need to comply with interfacing to the commonly used communication network and proper codecs.

The invention is further explained by way of example and with reference to the accompanying drawings, wherein:

Fig .1 is a block diagram of a speakeφhone system with the baseline unit of the invention; - Fig. 2 is block diagram of a video conferencing system with the baseline unit of the invention; - Fig. 3 is a block diagram of the baseline unit of the invention;

Fig. 4 is a block diagram of a system of the invention illustrating both the speakeφhone and video conferencing functionalities in one system;

Figs. 5 and 6 are tables for illustrating the TDM protocols for audio conferencing and video conferencing, respectively; and - Figs. 7 and 8 show GUI aspects in the invention. Throughout the figures, same reference numerals indicate similar or corresponding features.

Speakeφhones that are currently commercially available neither offer the option to interface to a video conferencing system nor have the ability to transfer data. Furthermore, the known speakeφhones have been designed for a specific network interface, for example, for ISDN, PSTN, POTS, etc. Therefore, if the user wishes to go from, e.g., a PSTN-based speakeφhone to, e.g., an ISDN-based one, he/she is to purchase an entirely new speakeφhone; and two completely separate systems are required to carry out the speakeφhone and video conferencing tasks that relate to audio processing aspects. For example, Polycom markets several speakeφhones and video conferencing systems as separate entities. Their speakeφhones cannot work together with their video conferencing systems, and merely form separate appliances used for separate functions only. Today's speakeφhones are solely used for audio communications based on a single protocol, and neither offer other protocol interfaces nor provide data transfer mechanisms. The invention now provides a communication system that is modular and flexible. The invention allows the same apparatus to interface with a video conferencing system as an accessory, and also to be used as a standalone speakeφhone in combination with a dedicated separate interface to a specific one of the communication networks, such as, PSTN, ISDN, LAN, etc. In addition, the speakeφhone has an IrDA port for enabling data transfer via infrared. IrDA (Infrared Data Association) is an organization that is sponsored by the industry for establishing international standards regarding infrared communication links. A focused ray of infrared light is modulated and sent from a transmitter to a receiver over a relatively short distance. Infrared data transport has become important in wireless data communication due to the popularity of, e.g., laptop computers, personal digital assistants (PDAs), digital cameras, etc. For example, an IrDA link enables communicating a file between a notebook computer and another data processing system. The IrDA link requires dedicated hardware and software. According to the IrDA- 1.1 standard, the maximum data size that may be transmitted is 2048 bytes and the maximum transmission rate is 4 Mbps.

Fig. 1 is a first basic block diagram of a communication system 100 in the invention. System 100 has a speakeφhone configuration. The portion at the end-user's site comprises a first apparatus 101 with a baseline unit 102 and second apparatus 103 comprising a sub-system 104. Apparatus 103 is, for example, a wallmount and is coupled between apparatus 101 and a telephone network 106. Baseline unit 102 has a user-interface 108 for audio communication with one or more other parties (not shown) via network 106. Sub-system 104 communicates with baseline unit 102 using a first protocol for digital communication. Sub-system 104 serves to interface baseline unit 102 to network 106 that uses a second protocol different from the first protocol. The second protocol is independent of the first protocol. Sub-system 104 translates the signals from the first protocol to signals for the second protocol and vice versa. Protocol conversion itself is known and is described within various contexts in, for example, U.S. patent 5,649,001; U.S. patent 5,790,180; U.S. patent 5,778,189; U.S. patent 5,182,748; U.S. patent 5,805,582. A speakeφhone is typically a table-top device, and a hardwired interconnection between apparatus 101 and 103 may be considered a nuisance as it is an obstacle limiting the use of the table's top. Therefore, communication between baseline unit 102 and sub-system 104 is preferably wireless.

Fig. 2 is a second block diagram of a communication system 200 in the invention. System 200 is configured for video conferencing. System 200 comprises apparatus 101 as described above and an apparatus (or set of apparatus) 203 that processes (process) the input received from apparatus 101. In addition, apparatus 203 comprises means (not shown) to process video data generated at this end and video data received via network 204 from another user (not shown). The combination of apparatus 101 and apparatus 203 forms a video conferencing system for real time communication of audio and video using a third (network) protocol. The first protocol is independent from the network protocol in the sense that apparatus 203 provides any protocol conversion needed for the network communication. Apparatus 101 thus is useable as a separate speakeφhone, as discussed under Fig. 1, and as an accessory to video conferencing apparatus 203. Preferably, the interconnection between apparatus 101 and 203 is wireless. Fig. 3 is a block diagram with main components of baseline unit 102 in apparatus 101. Unit 102 comprises a DSP 302 for signal processing and as a main microprocessor. DSP 302 comprises, for example, an ADSP-21065L from Analog Devices. DSP 302 enables control of several peripherals so as to allow communication with the outside world. DSP 302 has first and second synchronous bi-directional serial ports 304 and 306. First serial port 304 provides a four-wire synchronous interface and is connected to audio codecs 308, 310 and 312. Audio codecs 308-312 each comprise, for example, a stereo ADC/DAC such as the AD 1847 of Analog Devices. Codec 308 is coupled to first and second microphones 314 and 316, and to a loudspeaker 318. Codec 310 is coupled to third and fourth microphones 320 and 322. Codec 312 is coupled to a fifth microphone 324. o

Microphones 314, 316, 320, 322 and 324 are directional in this example. Microphones 314, 316 and 320 are, for example, built-in into apparatus 101, and microphones 322 and 324 are external microphones that are optionally connected to baseline unit 102.

DSP 302 acquires and supplies audio data through synchronous serial port 304. Serial port 304 is configured for a time-division multiplexed (TDM) interface where a single framing pulse is followed by several time-slots of 16-bit audio data. Each time-slot is a different channel and since serial port 304 is bi-directional, data transmission and data reception is perceived as simultaneously. The protocol at port 304 uses, for example, 8 time slots, 16 bits per time slot over a 4-wire bus, thus keeping the transmit and the receive data lines separate. TDM data is buffered in and out of DSP 302 using DMA. Accordingly, DSP 302 is interrupted every 128 samples (8ms based on 62.5μs sample interval at a 16KHz sample rate).

As discussed above apparatus 101 has five audio inputs via microphones 314, 316, 320, 322 and 324 and a single audio output via speaker 318. As the audio inputs are being sampled, they are buffered to memory as further discussed below. After 128 groups of five samples each have been buffered, an interrupt is generated. This DMA interrupt allows DSP 302 to switch the DMA channel to a new buffer and begin processing newly acquired data. DSP 302 performs an acoustic echo canceling (AEC) algorithm on the audio data, buffers the AEC results and eventually sends the echo-canceled audio data to sub-system 104 or sub-system 202 depending on the system's configuration. In the case of the video conferencing configuration of Fig. 2, sub-system 202 mixes the audio data received from apparatus 101 with audio data obtained from other sources such as ISDN or POTS audio sources (see below). Sub-system 202 sends to apparatus 101 the mixed audio data obtained from the other sources. Upon receipt of these mixed audio data, baseline unit 101 buffers these data and eventually plays them out via speaker 318.

The AEC process operates on a large block of samples at a time. This could well occupy DSP 302 for milliseconds from start to finish. Accordingly, the AEC process is preferably performed as a foreground task, whereas other events such as keypad entry (further discussed below), IrDA UART communications (further discussed below) and TDM DMA must be processed at the interrupt level to ensure a timely response.

Second serial port 306 functions as a communications interface to sub-system 104 in the speakeφhone system 100 or to sub-system 202 in video conferencing system 200. Port 306 is connected to sub-system 104 or sub-system 202 via a high-speed differential transceiver 326 and uses a three-wire TDM communication format. In the speakeφhone configuration of system 100, baseline unit 102 is coupled to a telephone network such as for, e.g., POTS, ISDN, Ethernet or DSL. Since each type of telephone network 106 has its own protocol, sub-system 104 translates protocols and signaling as is discussed below. Each network protocol requires a different wall mount apparatus 103. For example, apparatus 103 allows interfacing to a POTS line through a dedicated sub-system 104. Baseline unit 102 stores a program for detecting the configuration of the system (here POTS) wherein it is being used (discussed below). Baseline unit 102 stores a program for recognizing a ring-detect command and a hang-up-detect commands from sub-system 104. POTS sub-system 104 further provides ADC samples from the POTS line. In the video conferencing configuration 200 of a system in the invention baseline unit 102 sends and receives audio data, system commands (e.g., telephone dialing commands for setting up the video conference) and IrDA data packets. Again, baseline unit 102 stores a program for detecting the configuration of the system (communicating with a subsystem 202 wherein it is being used. User-interface 108 of baseline unit 102 comprises a keypad 328. DSP 302 interfaces to keypad 328 based on a four-row and six-column matrix, that thus has a maximum capacity of twenty-four buttons in this embodiment. DSP 302 scans each row for one of the associated six pushed buttons. If a button has been pushed, DSP 302 times out a safety interval in order to verify the button is still pushed, and then takes the requested action. Keypad 328 includes, for example, a standard telephone keypad (0-9,*,#), call on/off, mute on/off, re-dial, OK, conference, flash, and volume +/-. Keypad 328 allows an audio conference or a video conference be initiated in similar manners. Keypad 328 also comprises cursor control keys for allowing user-interaction with an on-screen GUI in video conferencing configuration of system 200. User-interface 108 further comprises a small speaker 330 for ordinary audio calls only.

Apparatus 101 enables file or data transactions, e.g., a Word document or a spreadsheet, through infra-red, e.g., from a notebook PC or laptop PC for display on the screen at the receiving end of the video conferencing system or from apparatus 101 to a PC. Also, it enables apparatus 101 to control the user's PC or another piece equipment through infra-red commands. To this end, baseline unit 102 comprises first and second infra-red transceivers 332 and 334 coupled to DSP 302 via first and second IrDA UART infra-red controllers 336 and 338, respectively. Preferably, transceivers 334 and 336 are capable of data rates in the order of 4Mbps. UART stands for "universal asynchronous receiver-transmitter". A UART handles asynchronous serial communication. DSP 302 communicates with controllers 330 and 332 over a memory-mapped (at different addresses), parallel 8-bit interface that resembles a typical RS-232 UART type device. Transceivers 334 and 336 are spaced 180 degrees apart on apparatus 101. Infra-red is transmitted in a narrow bundle and, accordingly, the direction and angle under which the light is received is important. This configuration allows all parties gathered around a conference table to direct their IrDA devices towards table-top apparatus 101 without having to rotate apparatus 101. The intent of using two IrDA sensors is to prevent that two parties upload their data into apparatus 101 simultaneously. DSP 302 communicates with only one controllers 336 and 338 at any given time, whichever was the first to establish a link. The other one of controllers 336 and 338 is disabled until the link is terminated.

Note that a stand-alone apparatus 101 may use the IRDA link for a laptop modem interface, apparatus 101 behaving as a laptop modem.

Controllers 336 and 338 each comprise a PC87109VBE Advanced UART and Infrared Controller from National Semiconductor. The PC87109 is a serial communication device with infra-red capability. It supports six modes of operation: UART, Shaφ-IR, IrDA 1.0 SIR, IrDA 1.1 MIR and FIR, and Consumer-IR (also referred to as TV Remote or Consumer Remote Control). The UART supports all operational modes. The UART uses one DMA channel. One channel is required for infra-red based applications since infra-red communications work in half-duplex fashion. To further ease driver design and simplify the implementation of infra-red protocols, a 12-bit timer with 125ms resolution has also been included. Controllers 336 and 338 are compatible with standard UART signaling, IrDA 1.0 SIR (2.4Kbps- 115.2Kbps) and IrDA 1.1 MIR/FIR (0.576Mbps, 1.152Mbps and 4.0Mbps). Each of controllers 336 and 338 has eight banks of eight control registers each, and a 32 byte FIFO for data transmit and a 32 byte FIFO for data receive. A single interrupt output can be programmed to notify DSP 302 of an empty or almost empty transmit-FIFO, a full or almost full receive-FIFO, and of any errors detected by the controller 336 or 338. Each of controllers 336 and 338 has an interrupt output that is connected to one of the interrupt inputs of DSP 302. Each of IrDA UART controllers 336 and 338 also supports a DMA request output and DMA acknowledge output for high-speed, low-overhead communications. Since DSP 302 supports parallel DMA devices, the IrDA interface can be implemented by using DMA -driven I/O or by code that responds to FIFO interrupts directly.

Baseline unit 102 comprises an EEPROM 340, here configured as a 512KB x 8 EEPROM. DSP 302 boots from EEPROM 340. EEPROM 340 also stores non-volatile data such as user-programmed telephone numbers, voice prompts and, if speakeφhone system 100 supports answering machine functionalities, messages. DSP 302 has an internal SRAM 342, that in this example has a capacity of 544Kbits, configured into 4K x 48 program memory and 12K x 32 data memory. An external 512K X 32 SDRAM 344 is necessary because the acoustic echo-canceling requires storing large amounts of samples per microphone. As the system features increase, SDRAM 334 can be used for program storage. DSP 302 is designed for glueless interfacing to SDRAM and is capable of generating refresh cycles.

Fig. 4 is a block diagram clarifying the interaction between baseline unit 102 and sub-systems 104 and 202 in systems 100 and 200, respectively.

Apparatus 101 is illustrated as accommodating DSP 302; a block 402 with audio in/out functionality represented by microphones 314, 316, 320, 322 and 324 and loudspeaker 318 discussed above; UI 108; a block 404 representing the IrDA functionality as provided by IrDA UART controllers 336 and 338 and transceivers 332 and 334 as discussed above; and a block 406 that serves as the interface to wallmount 103 and to system 203. Block 406 comprises, among other things, transceiver 326 and electrical drivers (not shown) in a hardwired embodiment of the system in the invention. Block 406 enables apparatus 101 to communicate with apparatus 103 or apparatus 203.

Apparatus 103, the wallmount interface to telephone network 106 in the speakeφhone configuration, comprises a processor 408 that serves as a codec. Note that baseline unit 102 and wallmount 103 communicate via a digital first protocol, e.g., based on TDM as illustrated below, and that both audio and IrDA data can be coded through wallmount 103.

Apparatus 203 or set of apparatus 203 comprises an interface/driver block 410 corresponding to interface/driver block 406 in baseline unit 102, a DSP 412 and a host processor 414. The typical video conferencing features such as a display monitor and a remote control for the video conferencing system are not shown here. Note that UI 108 of apparatus 101 provides complete user-control of the video conferencing system. Apparatus 203 functions as the main codec for audio, video and other data (e.g., IrDA-generated) and communicates with remote equipment via network 204. Note that baseline unit 102 and apparatus 203 communicate via a digital first protocol, e.g., based on TDM as illustrated below.

Interface/driver block 410 enables DSP 302 and DSP 412 to communicate with each other. DSP 412 comprises, for example, an ADSP-21065L from Analog Devices, as does DSP 302 discussed above. DSP 412 serves to demultiplex the TDM data received from baseline unit 102 and to send the data to host processor 414 either upon pre-processing or directly. Host processor 414 comprises, for example, a GxM from Cyrix and handles the final disposition of the data to network 204, either compressed or uncompressed, for communication to the other end. The reverse holds true for data received from network 204: host 414 decompresses and reformats the data so that DSP 412 can time-multiplex them and send them to baseline unit 102 via interface/driver blocks 410 and 406 using the TDM digital protocol.

The IrDA data received by IrDA transceiver 332 and 334 of baseline unit 102 gets sent to DSP 302. There, the data can be filtered if need be, and then this data is inserted at the appropriate place in the TDM data stream and sent to either video processing sub-system 203 or wallmount unit 103.

In sub-system 203, the TDM data stream is received by DSP 412 and then demultiplexed. From there, since the IrDA data is just that: data, it can be processed or manipulated by DSP 412 and sent through to host 414 for final disposition or it can be passed through untouched and processed in host 414. Host 414 can then send this data (either compressing it or not) through to any of the communications links (LAN, ISDN or POTS) for transmission to the other end. The reverse is true for the remote system sending its IrDA data to local sub-system 203.

In the configuration 100 with wallmount unit 103, the IrDA interface is being used for some sort of modem functionality. The data is received by processor 408 and is encoded for transmission via ISDN or POTS. Local processor 408 in wallmount unit 103 encodes and decodes the data for transmission to or reception from the remote system at the other end of network 204. in the art.

Host 414 has a Windows CE operating system. A set of Windows CE libraries is provided to handle IrDA communications. Note that apparatus 101 does not directly interface to host 414. As discussed, baseline 102 sends data to apparatus 203 over a synchronous serial interface. DSP 412 captures the data and parses the data between audio, keypad command and IrDA data content. The IrDA data is from there provided to the host 414 over an ISA bus. Between the Windows CE libraries and an IrDA device exists a hardware abstraction layer of host code that is unique to the IrDA peripheral device. Here, the IrDA peripheral is DSP 412. DSP 412 provides status and data in a format similar to the data that DSP 302 in baseline 102 reads from actual IrDA UART devices 336 and 338. This means that the maximum IrDA data rate is constrained by the bandwidth allocated for IrDA data on the synchronous serial interface between baseline unit 102 and apparatus 203.

The serial interface allocates 8, 16-bit time-slots in each direction every 62.5us (16KHz). That multiplies out to 2.048Mbps. Since the highest speed IrDA standard calls for 4Mbps, the maximum data rate is not attained in this example. The lowest data rate mode supported by IrDA is SIR (Slow-Speed Infrared Mode), allowing throughput between 2.4Kbps and 115.2Kbps.

IrDA operates half-duplex, but does allow for bi-directional communications. Baseline unit 102 can transmit as well as receive IrDA data. Transmit is not as important, since the main puφose of IrDA is to allow video conference sub-system 203 to display user laptop documents, spreadsheets and graphics. Transmit is necessary for handshaking, for sending a beacon during idle times and, when apparatus 101 operates stand-alone, for transporting network data from the modem back to the laptop.

IrDA UARTs 336 and 338 is initially configured for transmission, since host 203 transmits first to discover potential peripherals. After sending discovery requests, IrDA UARTs 336 and 338 is placed in receive mode to wait for potential peripherals to respond. The TDM protocols between baseline unit 102 and wallmount 103, and between baseline unit 102 and video conferencing sub-system 203 are discussed below with reference to the tables I and II in Figs. 5 and 6. Table I in Fig. 5 gives an overview of the time-slot allocation in the TDM protocol between apparatus 101, comprising baseline unit 102, and wallmount 103. Apparatus 101 is referred to as "CUB" in table I. Wallmount 103 interfaces to a single POTS line in this example. The TDM protocol uses 8- time-slots per frame, 16-bits per time slot, and a 3-wire bus for transport of audio and commands between apparatus 101 ("Cub") and wallmount 103. Wallmount 103 indicates its identification code (ID) on time-slot 6. Cub 101 is designed to initialize serial port 306 as a slave where serial clock and frame-sync are driven by wallmount 103. Cub 101 is designed to recognize certain IDs associated with the type of network 106, being POTS in this example.

POTS wallmount 103 comprises an Analog Devices AD 1847, stereo ADC/DAC or codec, referred to as processor 408 above and in Fig. 4. Codec 408 samples at an 8KHz rate. Since the interface is a 3-wire, synchronous serial, transmit and receive data share the same signal. Cub 101 outputs to AD 1847 the control word on time slot 0, POTS output data on time-slot 1 and auxiliary output data on time-slot 2. POTS wallmount 103 outputs status on time-slot 3, POTS input data on time slot 4, auxiliary input data on time slot 5, a unique identification code (ID) on time-slot 6 and a ring detect state on time-slot 7.

Table II in Fig. 6 gives an overview of the time-slot allocation in the TDM protocol between apparatus 101, comprising baseline unit 102, and video conferencing sub- system 203 referred to as "Tiger".

Cub 101 and Tiger 203 have a direct, 3-wire, synchronous serial interface. Data is transmitted between Cub 101 and Tiger 203 at 8.192Mbps. With 32 time-slots per frame, each time-slot containing 16-bits of data, that results in a 16KHz frame rate. Time slots 0-14 contain data sent from Tiger 203 to Cub 101. Time-slots 16-30 contain data sent from Cub 101 to Tiger 203. Time-slots 15 and 31 are not driven by either Cub 101 or Tiger 203, thus allowing for safe change in data direction. Time-slot 6 contains the Tiger ID code. Table II lists all 32 time-slot assignments for the Tiger wall mount.

Cub 101 works in the slave TDM mode when it communicates with wallmount 13 or with Tiger 203. After powered up CUB 101 initializes itself to a default state, whereupon it polls the interface to its host (wallmount 103 or Tiger 203) until it gets a valid ID. The ID received determines the operating state assumed by Cub 101. Because wallmount 103 and Tiger 203 are the masters of the host interface, it is important that a valid ID be sent to Cub 103 before the connection is established.

Figs. 7-8 illustrate a GUI aspect of video conferencing system 200. Apparatus 101 ("Cub") has a set of cursor control buttons 702, 704, 706, 708 and 710 for user-control of a cursor 802 being displayed on a display monitor 804 of video conferencing system 200. Buttons 702-708 correspond in arrangement and shape with the shape of segmented cursor 802 that resembles that of a split-puck with segments 806, 808, 810 and 812. Buttons 702-708 are for moving cursor 802 up, down, to the left and to the right, respectively, in a menu with options arranged as bullets, e.g., bullets 814 and 816, arranged in a grid. Manipulating one of buttons 702-708 causes the cursor to split into its segments 806-812 that smoothly move to the nearest option in the direction corresponding with the selected one of buttons 702-708. Upon arriving, the segments are assembled and form a puck while surrounding the bullet. Button 712 serves to validate a selection. The visual correspondence between hard buttons 702-708 and the puck's segments 806-812 greatly enhances the user-friendliness of system 200 as the operation is conceptually clear. Smooth graphics facilitates navigation as well because the user is less likely to lose track of the cursor on the screen. Of course, this aspect is also useful in GUI's other than the one presented within the video conferencing context. In summary, Cub 101 differs from existing speakeφhone systems in a variety of ways. First, its modularity allows it to assume more than one role, namely to interface to video conferencing system (Tiger) 203 as an accessory, or to interface to wallmount 103 as a speakeφhone. When used as an accessory to Tiger 203, Cub 101 offers highest possible audio quality by having microphones 314, 316, 320 (and optional external microphones 322 and 324) positioned closer to the audio sources. Another very important feature is that Cub 101 allows to make a video conferencing call just as one would with a normal speakeφhone. Second, Cub's ability to transfer data via an IrDA link during a video conferencing call is unique. For example, someone can be in the middle of a video conferencing call and decide to send to the far-end a copy of their presentation from their notebook. All they have to do is upload it through cub's IrDA port (404) to get it to the other end. A third difference is provided by the upgradeability of Cub 101 having a standard interface to Tiger 203 or wallmount 103. The user can always upgrade to a video conferencing system, or convert from one communications network to another with a simple wallmount change. The way this is achieved is by designing universal baseline unit 102 that interfaces to Tiger 203 or wallmount 103 via a proprietary data link protocol, illustrated in Tables I and II. In addition, Cub 101 comprises all the electronics necessary to perform audio echo cancellation (AEC), a keypad for placing a call and controlling an on-screen cursor when connected to Tiger 203, and an IrDA port for data transfers. Wallmount unit 103 and Tiger 203 contain the required power and network interfaces.

The benefits thus attained are manifold. As an interactive speakeφhone, Cub 101 offers excellent multi-microphone audio quality, and facilitates data transfers over an IrDA communications port. Moreover, it allows an individual to change from a PSTN speakeφhone to an ISDN speakeφhone by simply changing wallmount unit 103. As an accessory to Tiger 203, Cub 101 allows a video conferencing call to be placed in the same fashion as a ordinary telephone call: by entering the number in the keypad. For example, the current method used for making a video conferencing call is to use on-screen messages. However, it has turned out that a video conferencing call is preferable started as a telephone call. This is available with Cub 101 connected to Tiger 203. Cub 101 offers an upgradeable path from a stand-alone speakeφhone to a Tiger video conferencing system, while maintaining the exact same speakeφhone functionality. Therefore, an individual with budgetary, and/or real estate limitations (small table) has an attractive growing path to the latest interactive conferencing technology from Philips Electronics (Tiger), without needing to purchase two separate pieces of equipment for their audio and video conferencing solutions. One last benefit may be overlooked, but is extremely valuable is the following. Cub's universal baseline unit 102 contains the most expensive components, therefore, if manufacturing projections are off or have to be adjusted to another type of interface, it does not affect the manufacturer at all. For example, if more IDSN speakeφhones have been projected than eventually can be sold, a costly inventory is not carried since the same baseline unit can be used in other speakeφhone systems.

Claims

CLAIMS:
1. An end-user communication system comprising: a first apparatus with a baseline unit that has a user-interface for enabling audio communication via a communication network; and a second apparatus with a sub-system for interfacing the baseline unit to the network; wherein: the sub-system and the baseline unit are capable of communicating using a first protocol for digital transmission; and the sub-system is capable of enabling communication via the network using a second protocol independent of the first protocol.
2. The system of Claim 1, wherein:
- the baseline unit has a speakeφhone functionality; the communication network comprises a telephone network; - the sub-system enables to interface the baseline unit to the telephone network
3. The system of Claim 1, wherein: the first and second apparatus form part of a video conferencing system; and - the second apparatus has means for video processing.
4. The system of Claim 3, wherein:
- the user interface of the baseline unit enables user-control of the video conferencing system.
The system of Claim 4, wherein: the user interface of the baseline unit enables user-control of the video conferencing system through control of an on-screen cursor using control keys at the baseline unit.
6. The system of Claim 1, wherein the first protocol comprises a time-division multiplexed data and command protocol.
7. The system of Claim 1, wherein the baseline unit is capable of receiving wireless data for being sent via the network.
8. A method of enabling telephone conferencing via a network, the method comprising: - receiving audio signals at a first apparatus; converting the audio signals into digital data at the first apparatus; - transmitting the digital data to a second apparatus; and coding the digital data at the second apparatus for interfacing to the network.
9. The method of Claim 8, further comprising:
- receiving wireless data at the first apparatus; using a TDM protocol to transmit the digital data and the wireless data to the second apparatus.
10. A method of enabling video conferencing via a network, the method comprising: receiving audio signals at a first apparatus;
- converting the audio signals into digital data at the first apparatus; transmitting the digital data to a second apparatus; and - coding the digital data at the second apparatus for interfacing to the network.
11. The method of Claim 10, further comprising:
- enabling user-control of the video conferencing from the first apparatus.
12. The method of Claim 10, further comprising: receiving wireless data at the first apparatus; using a TDM protocol to transmit the digital data and the wireless data to the second apparatus.
EP20000912482 1999-03-08 2000-02-18 Speakerphone is also module for video conferencing system Withdrawn EP1078523A1 (en)

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US264058 1994-06-22
US26405899A true 1999-03-08 1999-03-08
PCT/EP2000/001344 WO2000054502A1 (en) 1999-03-08 2000-02-18 Speakerphone is also module for video conferencing system

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US8934381B2 (en) 2001-12-31 2015-01-13 Polycom, Inc. Conference endpoint instructing a remote device to establish a new connection
US8934382B2 (en) 2001-05-10 2015-01-13 Polycom, Inc. Conference endpoint controlling functions of a remote device
US8976712B2 (en) 2001-05-10 2015-03-10 Polycom, Inc. Speakerphone and conference bridge which request and perform polling operations

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AU3423800A (en) 2000-09-28
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JP2002539684A (en) 2002-11-19
AU773804B2 (en) 2004-06-10
WO2000054502A1 (en) 2000-09-14

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