Enhanced range/graceful degradation for digital wireless telephone system

• US 6,574,769 B1
• Filed: 11/15/2000
• Issued: 06/03/2003
• Est. Priority Date: 12/12/1997
• Status: Expired due to Fees

• Current Assignee: Thomson Licensing SA (Vantiva SA)
• Original Assignee: Thomson Licensing SA (Vantiva SA)
• Inventors: Stewart, John Sidney, Ramaswamy, Kumar, Knutson, Paul Gothard
• Primary Examiner(s): Chung, Phung M.
• Application Number:
  US09/581,192
• Time in Patent Office: 930 Days
• Field of Search: 714/776, 714/762, 714/758, 714/746, 714/704
• US Class Current: 714/776
• CPC Class Codes: H04L 1/0009 by adapting the channel coding H04L1/1812 takes precedence
• CPC Class Codes: H04L 1/0014 by adapting the source coding
• CPC Class Codes: H04L 1/0025 Transmission of mode-switching indication
• CPC Class Codes: H04L 1/0071 Use of interleaving interleaving per se H03M13/27
• CPC Class Codes: H04L 1/0079 Formats for control data H04L1/16 takes precedence; training sequences H04L25/00 and H04L27/00
• CPC Class Codes: H04L 1/0083 Formatting with frames or packets; Protocol or part of protocol for error control

Abstract

A wireless telephone system comprising a base unit coupleable to one or more external telephone lines and having 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 fixed-size audio packets comprising a plurality of audio data samples and a plurality of error correction bits. Qualities of the digital link are monitored to determine whether a change in error rate occurs. In response to the determination, 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.

Claims

1. A wireless telephone system, comprising:

(a) a base unit coupleable to one or more external telephone lines and having a base transceiver;

(b) at least one wireless handset having a handset transceiver for establishing a digital link with the base unit via the base transceiver over a wireless channel, characterized in that the digital link employs fixed size audio packets comprising a fixed number of audio data samples and a plurality of error correction bits each audio packet having one of a plurality of packet structures, the system further comprising;

(c) means for monitoring the digital link for determining a change in a bit error rate of audio packets received over the channel by one of the handset and the base unit; and

(d) means for changing a packet structure of subsequent audio packets, in response to the determination, so that said subsequent audio packets are of the fixed size and have a smaller number of bits devoted to each of the fixed number of audio data samples and a larger number of bits devoted to error correction if said error rate has increased and vice-versa if said error rate has decreased.

2. The system of claim 1, wherein the data samples are adaptive differential pulse code modulation (ADPCM) samples, each ADPCM sample having a number of bits.

3. The system of claim 1, wherein the base unit comprises means (c) and (d).

4. The system of claim 1, wherein the handset is determined to exceed or fall within a range limit if the bit error rate exceeds or falls below a threshold, wherein the means (d) changes packet structures when the handset exceeds or comes back within a range limit.

5. The system of claim 1, further comprising:

(e) means for monitoring a signal strength received over the channel by one of the handset and the base unit, wherein the means (d) for changing a packet structure of subsequent audio packets changes said packet structure in response to at least one of the determination of means (c) and the determination of means (e).

6. The system of claim 1, wherein said plurality of packet structures comprises a high-quality packet structure having a highest number of bits per audio data sample and a lowest number of error correction bits, a medium-quality packet structure having an intermediate number of bits per audio data sample and an intermediate number of error correction bits, and a low-quality packet structure having a lowest number of bits per audio data sample and a highest number of error correction bits.

7. The system of claim 6, wherein said high-quality audio packet structure employs an burst error correcting cyclic code, said medium-quality audio packet structure employs a 2×

interleaved burst error correcting cyclic code, and said low-quality audio packet structure employs a 4×

interleaved burst error correcting cyclic code.

8. The system of claim 7, wherein said audio packets comprise an audio packet header field, a data field, an audio data field comprising fixed number of audio data samples, and a parity bit field comprising the plurality of error correction bits.

9. The system of claim 8, wherein, in said medium-quality and low-quality packet structures, the extra error correction bits are distributed partly in the parity bit field and partly in the header field to allow more time for said system to achieve lock on a given audio packet at the higher error rates associated with a digital link employing medium-quality or low-quality packet structure audio packets.

10. The system of claim 1, further comprising a plurality of wireless handsets including the handset, wherein:

the base (120) unit comprises a plurality of logical base transceivers; and

each handset of the plurality of handsets has a handset transceiver for establishing a time-division multiple access (TDMA) link over a shared RF channel with the base unit via a corresponding logical base transceiver, in which each handset communicates during an exclusive time slice of a TDMA scheme that allocates time slices to active handsets.

11. A method for communicating between a base unit and a wireless handset of a wireless telephone system, wherein the base unit is coupleable to one or more external telephone lines and has a base transceiver, and the handset comprises a handset transceiver, the method comprising the steps of:

(a) establishing, between the handset transceiver and the base transceiver, a digital link over a wireless channel, characterized in that the digital link employs fixed size audio packets comprising a fixed number of audio data samples and a plurality of error correction bits, each audio packet having one of a plurality of packet structures, the method further comprising the steps of;

(b) monitoring the digital link to determine a change in a bit error rate of audio packets received over the channel by one of the handset and the base unit; and

(c) changing a packet structure of subsequent audio packets, in response to the determination, so that said subsequent audio packets are of the fixed size and have a smaller number of bits devoted to each of the fixed number of audio data samples and a larger number of bits devoted to error correction if said error rate has increased and vice-versa if said error rate has decreased.

12. The method of claim 11, wherein step (b) comprises the step of determining that the handset is exceeding or falling within a range limit if the bit error rate exceeds or falls below a threshold, respectively, step (c) further comprising the step of changing the packet structure when the handset exceeds or comes back within a range limit in accordance with the determination of step (b).

13. The method of claim 11, wherein said plurality of packet structures comprises a high-quality packet structure having a highest number of bits per audio data sample and a lowest number of error correction bits, a medium-quality packet structure having an intermediate number of bits per audio data sample and an intermediate number of error correction bits, and a low-quality packet structure having a lowest number of bits per audio data sample and a highest number of error correction bits.

14. The method of claim 13, wherein said high-quality audio packet structure employs an burst error correcting cyclic code, said medium-quality audio packet structure employs a 2×

interleaved burst error correcting cyclic code, and said low-quality audio packet structure employs a 4×

interleaved burst error correcting cyclic code.

15. The method of claim 14, wherein said audio packets comprise an audio packet header field, a data field, an audio data field comprising fixed number of audio data samples, and a parity bit field comprising the plurality of error correction bits.

16. The method of claim 15, wherein, in said medium-quality and low-quality packet structures, the extra error correction bits are distributed partly in the parity bit field and partly in the header field to allow more time for said system to achieve lock on a given audio packet at the higher error rates associated with a digital link employing medium-quality or low-quality packet structure audio packets.

17. A first transceiver for communicating with a second transceiver, the first transceiver comprising:

(a) an encoder portion for encoding fixed-size audio data packet to be transmitted via a digital link established with the second transceiver via over a wireless channel, characterized in that the digital link employs fixed size audio packets comprising a fixed number of audio data samples and a plurality of error correction bits, each audio packet having one of a plurality of packet structures, the first transceiver further comprising;

(b) a decoder portion for decoding fixed-size audio data packet received from the second transceiver over the wireless channel; and

(c) an error rate monitor coupled to the decoder portion for determining whether, based on a bit error rate determined in said received audio data packets, the first and second transceivers are exceeding or falling within a range limit relative to one another and for requesting that a packet structure of subsequent audio packets be changed, in response to the determination, so that said subsequent audio packets are of the fixed size and have a smaller number of bits devoted to each of the fixed number of audio data samples and a larger number of bits devoted to error correction if the first and second transceivers are exceeding a range limit and vice-versa if the first and second transceivers are falling within a range limit.

18. The first transceiver of claim 17, wherein said plurality of packet structures comprises a high-quality packet structure having a highest number of bits per audio data sample and a lowest number of error correction bits, a medium-quality packet structure having an intermediate number of bits per audio data sample and an intermediate number of error correction bits, and a low-quality packet structure having a lowest number of bits per audio data sample and a highest number of error correction bits, wherein said high-quality audio packet structure employs an burst error correcting cyclic code, said medium-quality audio packet structure employs a 2×

interleaved burst error correcting cyclic code, and said low-quality audio packet structure employs a 4×

interleaved burst error correcting cyclic code.

19. The first transceiver of claim 18, wherein said audio packets comprise an audio packet header field, a data field, an audio data field comprising fixed number of audio data samples, and a parity bit field comprising the plurality of error correction bits.

20. The first transceiver of claim 19, wherein, in said medium-quality and low-quality packet structures, the extra error correction bits are distributed partly in the parity bit field and partly in the header field to allow more time for said system to achieve lock on a given audio packet at the higher error rates associated with a digital link employing medium-quality or low-quality packet structure audio packets.