Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04B—TRANSMISSION
  • H04B7/00—Radio transmission systems, i.e. using radiation field
  • H04B7/005—Control of transmission; Equalising
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
  • H04L1/00—Arrangements for detecting or preventing errors in the information received
  • H04L1/0078—Avoidance of errors by organising the transmitted data in a format specifically designed to deal with errors, e.g. location
  • H04L1/0083—Formatting with frames or packets; Protocol or part of protocol for error control
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
  • H04L1/00—Arrangements for detecting or preventing errors in the information received
  • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
  • H04L1/0009—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the channel coding
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
  • H04L1/00—Arrangements for detecting or preventing errors in the information received
  • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
  • H04L1/0014—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the source coding
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
  • H04L1/00—Arrangements for detecting or preventing errors in the information received
  • H04L1/0078—Avoidance of errors by organising the transmitted data in a format specifically designed to deal with errors, e.g. location
  • H04L1/0079—Formats for control data
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
  • H04L1/00—Arrangements for detecting or preventing errors in the information received
  • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
  • H04L1/0023—Systems modifying transmission characteristics according to link quality, e.g. power backoff characterised by the signalling
  • H04L1/0025—Transmission of mode-switching indication
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
  • H04L1/00—Arrangements for detecting or preventing errors in the information received
  • H04L1/004—Arrangements for detecting or preventing errors in the information received by using forward error control
  • H04L1/0056—Systems characterized by the type of code used
  • H04L1/0071—Use of interleaving

Definitions

• the present invention relates to wireless telephone systems and, in particular, to multi-line wireless telephone systems having limited power for radio-frequency (RF) communications.
• RF radio-frequency
• a cordless or wireless telephone handset unit communicates via either analog or digital radio signals with a base unit, which is typically connected via a standard telephone line to an external telephone network. In this manner, a user may employ the wireless handset to engage in a telephone call with another user through the base unit and the telephone network.
• Multi-line wireless telephone systems are also in use in various situations, such as businesses with many telephone users. Such systems employ a handset that communicates with up to N handsets simultaneously, typically with digital communications schemes, such as time division multiple access (TDMA). It is desirable to implement the features of current private branch exchange (PBX) systems in a multi-line wireless telephone system.
• PBX private branch exchange
• Digital wireless telephone systems typically employ error- detection and correction schemes, such as FEC (forward error correction).
• FEC forward error correction
• some of the bits transmitted between handset an d base unit carry audio and other forms of data, and some of the bits are devoted to error correction.
• a given digital audio packet may comprise an audio packet header section (data), an audio data samples section (audio), and an FEC data section (error correction).
• a wireless telephone system comprises a base unit coupleable to one or more external telephone lines and has a base transceiver, and at least one wireless handset.
• the wireless handset has a handset transceiver for establishing a digital link with the base unit via the base transceiver over a wireless channel, wherein the handset and base unit communicate via the digital link by audio packets comprising a plurality of audio data samples and a plurality of error correction bits.
• the digital link qualities are monitored to determine a change in error rate.
• a packet structure of subsequent audio packets is changed from a first packet structure to a second packet structure to change the relative number of bits devoted to audio data samples and to error correction bits.
• Fig. 1 is a block diagram of TDMA multi-line digital wireless telephone system, in accordance with an embodiment of the pre sent invention
• Fig. 2 is a schematic representation of the TDMA audio packet structure used in the digital wireless telephone system of Fig. 1, in accordance with an embodiment of the present invention
• Fig. 3 is a schematic representation of alternative audio packet structures used in the TDMA packet structure of Fig. 2;
• Fig. 4 is a block diagram illustrating the transceiver architecture of the base unit and handsets of the system of Fig. 1, in accordance with an embodiment of the present invention.
• Fig. 1 there is shown a block diagram of spread spectrum TDMA multi-line digital wireless telephone system
• TDMA system 100 comprises a base unit 110, which has receiver an d transmitter units 112 and 11 1 , respectively, and is coupled to external telephone network 1 16 via telephone line(s) 115.
• S ystem a base unit 110, which has receiver an d transmitter units 112 and 11 1 , respectively, and is coupled to external telephone network 1 16 via telephone line(s) 115.
• N wireless handsets 120 j , 1202, . . . 120 ⁇ Each has a transmitter and receiver unit (transceiver), such as transmitter 121 and receiver 122 of handset 120 ⁇ .
• Base unit 110 also has a controller/microprocessor 113 for controlling and monitoring th e overall functions of the base unit 110.
• receiver unit 112 comprises N separate logical receivers, and transmitter unit
• system 111 comprises N separate logical transmitters, so that receiver an d transmitter units 112 and 111 provide N total logical transceiver units, one for each of N wireless handsets.
• M handsets (0 ⁇ M ⁇ N) are operating or active (i.e., in the process of conducting a telephone call).
• system 1 00 employs a digital TDMA scheme, in which each operating handset only transmits or receives data during its own "time slice" or slot.
• System 100 thus provides a wireless network between the base station 110 and each handset 120 j (1 ⁇ i ⁇ N).
• System 100 preferably employs block error coding to reduc e error.
• digitally compressed audio packets such as ADPCM (adaptive differential pulse code modulation) samples
• ADPCM adaptive differential pulse code modulation
• Block codes and ADPCM are preferred because of their low latency, which allows the wireless phone behavior to mimic that of a standard corded phone.
• Channel codes such as convolutional codes or turbo codes, or stronger source coding such as LPC (linear predictive coding), transform coding, o r formant coding incur more delay, which makes the system less like the equivalent corded telephone.
• a TDMA scheme is employed that monitors various qualities of the TDMA link, such as signal strength and/or bit error rates at a given receiver, to estimate the channel characteristics. Based on this estimate, th e transmitter switches between lower bit rate, higher distortion source coding and comparatively high bit rate, low distortion source coding. The freed up bits in the former case are used to increase the error coding bits or "strength.”
• the receiver m a y determines that the range limit is being or has been reached. At this point, the receiver alerts the relevant portion of the system to cause subsequent packets to be transmitted with the audio data portion a t lower quality or more distortion to devote more bits to error correction, i.e. to increase error coding strength. When at a later time the handset moves within range, this can also be detected, and higher-quality audio data resumed with fewer bits devoted to error correction, since a weaker error coding can be tolerated when th e handset is closer to the base.
• Structure 200 comprises a 2 ms (Td) field 210 of digital data, which comprises eight audio packets, such as audio packet 220.
• Each audio packet is a set of audio data transmitted either to a given handset from the base unit or vice- versa, during a given time-slice in an overall "epoch" scheme, during which time no other handsets receive or transmit data over th e system' s data channel.
• Each packet is labeled Ti or Ri, to indicate whether it is being transmitted from the base unit 110 or received b y the base unit 110, to or from a given handset 120 j .
• voice data is exchanged in packets containing 16 samples of voice data.
• these samples are 4 - bit ITU-T G.721 or G.727 ADPCM samples (i.e., a 32Kbps ADPCM signal).
• G.727 3 or 2 bit ADPCM samples 24 o r 16Kbps ADPCM signals, respectively
• an additional 16 or 32 bits are freed up for coding in each packet.
• the audio packet structure for a given handset may be changed, as described above, and as illustrated in further detail in Fig. 3.
• each audio packet 220 comprises various sub-fields or sections, whether structured as structure 310, 320, or 330. However, in the pre sent invention, bits are allocated differently among these fields in order to achieve better performance at range limits.
• each audio packet 220 comprises audio packet header, a data section, and audio data (ADPCM samples) section, and a parity bit section that contains FEC data.
• the audio packet header typically contains information identifying the audio packet (such as the handset), the current place in the epoch, and the like.
• Each audio packet has a fixed size, for example 100 total bits.
• each handset receives 1 6 ADPCM samples during each time slice of the epoch allocated for the handset to receive audio data; and transmits to the base unit 1 6 ADPCM samples during each time slice of the epoch allocated for th e handset to transmit audio data.
• the number of bits allocated to error coding is increased by lowering each sample' s quality.
• ADPCM and related technical issues are described in detail i n International Telecommunication Union (ITU), Recommendation G.727, ( 12/1990), "5-, 4-, 3- and 2-Bits Sample Embedded Adaptive Differential Pulse Code Modulation (ADPCM)," http://www.itu.ch.
• system 100 employs three levels of coding for encoding audio packets 220: high, medium, and low quality coding levels, inversely corresponding to the amount of bits devoted to error correction. These levels of coding correspond, respectively, to audio packet structures 310, 320, and 330.
• the default or standard structure has the standard (highest) quality of audio data samples transmitted in field 313.
• 16 ADPCM samples are transmitted per packet.
• 64 of the 100 bits are allocated to ADPCM section 313, to allow for 16 high-quality 4-bit ADPCM samples to b e transmitted.
• Structure 310 uses an (85,75) burst error correcting cyclic code, and has a payload of 74 bits (fields 312 and 313), one padding bit, and 10 error coding bits, which allows the code to correct a burst of 4 bit errors.
• In medium-quality packet structure 320 48 of the 100 bits are allocated to ADPCM section 323, to allow for 16 medium-quality 3 -bit
• Medium-quality structure 320 uses a 2x interleaved (38,29) burst error correcting cyclic code, which can correct a burst of up to 8 errors.
• Structure 320 has a payload of
• structure 320 uses 16 fewer audio data bits (64 — 48 or 74 — 58) , structure 320 has 16 more bits devoted to error coding (26 — 10).
• low-quality packet structure 330 In low-quality packet structure 330, only 32 of the 100 bits are allocated to ADPCM section 333, to allow for 16 low-quality 2-bi t ADPCM samples to be transmitted. Structure 330 uses a 4x interleaved ( 19, 11) burst error correcting cyclic code, which c an correct a burst of up to 16 errors. In the coding level that employs this packet structure, a payload of 42 bits is provided, with 32 coding bits and 2 padding bits.
• the extra bits freed up for coding by switching to the medium- or low-quality levels are employed as extra coding bits and are distributed in a particular way in the changed audio structure. For example, 16 extra coding bits are distributed in audio packet structure 320 partly in parity section 324 and partly in header section 321.
• all interleaving is done over the payload and check bits of the packet, and packet headers are not included in the coding.
• interleaving is utilized in th e lower coding levels (audio packet structures 320, 330), instead of simply increasing the length of parity section 314, to increase burs t error correction capability.
• the 2x interleaved 3 - bit sample structure the even bits after the header go to one decoder, and the odd bits go to another.
• This decoder is preferably implemented as a time multiplexed decoder, or the data will b e buffered and processed sequentially in a high speed decoder, since the data rate is quite slow compared to today's ASIC operating speeds.
• the received data is separated after the header into 4 streams.
• Data where the index is divisible by 4 is one stream; data where the remainder of said division is 1 is the second stream; wh ere the remainder is 2 is the third stream; and where the remainder is 3 is the fourth stream.
• These interleaved streams are preferably decoded either sequentially by a fast decoder, or by a multiplexed decoder. In one embodiment, the single high speed decoder is preferred.
• Fig. 4 there is shown a block diagram illustrating the transceiver architecture 400 of the base unit 1 1 0 and/or handsets 12O j of system 100 of Fig. 1, in accordance with a n embodiment of the present invention.
• both base unit 110 and handsets 120 j implement transceiver architecture, so that either a handset 120 j or base unit 110 may request a changed coding level/audio packet structure.
• only one of handset 12O j and base unit 110, for a given base unit/handset pair implements transceiver architecture 400.
• audio packet signals are received by a transceiver 400 (or either handset 120 j or base unit 110) via RF channel 401.
• ECC error correction
• an error rate monitor 410 monitors qualities of the received signal to determine the current bit error rate. (In an alternative embodiment, unit 410 or another unit monitors the signal strength. )
• error rate monitor 410 requests a change in the coding level, i.e. a change in the audio packet structure used to transmit sub sequent audio packet structures.
• the handset 120 j or base station 110 initiates requests a change to an increased coding level.
• the request to change coding levels is fed back to th e remote transmitter, and is preferably acknowledged either by a header change or by additional control data. If this request is correctly received, therefore, the packet header will acknowledge th e change to an increased coding level.
• the packet headers are increased in length for structures 320, 330 to allow more time for the system to lock up, since the reception conditions are degraded when the increased coding levels are required.
• error rate monitor 410 is implemented in base unit 110 and detects that handset 120 j is reaching a first range limit, it notifies base unit 110 to change from packet structure 310 to 320. Base unit 110 then notifies handset 120 j to switch to this format, and also changes its own transmission format. Subsequently, therefore, both base unit 110 and handset 120 j encode audio packets to be transmitted, in accordance with audio packet structure 320. If a second range limit is then reached and detected, the third structure 330 may be adopted. When the handset comes back within the mo s t recently exceeded range limit, a higher-quality structure (320 or 310) may be adopted once again. This allows range limits to be exceeded and addressed with more robust and stronger error correction, albeit at the cost of somewhat increase distortion or decreased audio quality. However, this is preferable to losing the connection altogether or incurring other artifacts that occur when uncorrectable errors increase.
• the present invention comprises a multi-rate coding system for the digital wireless telephone link, which implements a variable code rate technique, in which error rate measurements taken at the receiver are used to decide whether to ask the transmitter to increase error coding.
• qualities of the TDMA link such as signal strength or bi t error rate are monitored, and a switch from a high bit rate high quality voice coding scheme to a lower bit rate lower quality voice coding scheme is initiated.
• the error correction and detection coding is upgraded to use the bandwidth released by the lower bit rate coding method. This will reduce the number of bit errors an d improves the overall channel integrity, reducing the likelihood of a hang-up or other serious loss of communications.
• the present invention may be implemented in any digital audio transmission system, with or without the multi-line capability described above.
• the coding method described above is applicable in any digital modulation format.
• th e coding method of the present invention may be used in wireless data communications, extending the range of the wireless office notebook, for example.
• the present invention may be applied in a system having a fixed bandwidth and where data distortion is unacceptable, such as a data modem system.
• the receiving modem requests the transmitting modem to increase the relative proportion of parity or coding bits, at the cost of reduced data bit throughput, instead of at the cost of increased d ata distortion.
• the coding and data rate are changed to increase the immunity to noise and burst errors.
• packet structures other than those particularly described above may be used, as long as the packet structure allows the ability to support the different coding levels and data rates .
• the present invention may be implemented as an application on top of a more primitive packet structure, provided the packet structure is robus t enough.
• the structure protocol of the invention may be extended to support more than three coding levels.
• the present invention is implemented in a limited-power digital wireless telephone system having only a single base unit and handset, which does not employ TDMA.
• wireless system described above may be a cellular system where base unit 110 represents a base station serving one of the cells in a cellular telephone network.

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• Engineering & Computer Science (AREA)
• Computer Networks & Wireless Communication (AREA)
• Signal Processing (AREA)
• Quality & Reliability (AREA)
• Detection And Prevention Of Errors In Transmission (AREA)
• Mobile Radio Communication Systems (AREA)

Priority Applications (5)

Applications Claiming Priority (2)

Publications (1)

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Family Applications (1)

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Cited By (5)

Families Citing this family (16)

Citations (3)

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• 1998
  • 1998-09-01 KR KR1020007006305A patent/KR100557817B1/ko not_active Expired - Fee Related
  • 1998-09-01 EP EP98942340A patent/EP1040612B1/en not_active Expired - Lifetime
  • 1998-09-01 CN CNB988136201A patent/CN1227853C/zh not_active Expired - Fee Related
  • 1998-09-01 WO PCT/US1998/018105 patent/WO1999031838A1/en not_active Ceased
  • 1998-09-01 US US09/581,192 patent/US6574769B1/en not_active Expired - Fee Related
  • 1998-09-01 DE DE69804502T patent/DE69804502T2/de not_active Expired - Lifetime
  • 1998-09-01 AU AU90419/98A patent/AU9041998A/en not_active Abandoned
  • 1998-09-01 JP JP2000539606A patent/JP2002509387A/ja active Pending

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