Abstract:
A digital cordless telephone system using lossless pulse code modulation (PCM) for encoding an audio signal. By using an efficiently implemented and uncompressed encoding scheme, the system substantially reduces implementation costs and improves the quality of transmission for white signals such as modem signals. A transceiver implementation includes a PCM coder, an RF transmitter, an RF receiver, and a PCM decoder. Also presented is a method for communicating an audio signal from a transmitter unit to a remote receiver unit. The method includes steps of sampling an audio signal into a PCM data stream, modulating the PCM data stream onto a carrier, transmitting the carrier, receiving the carrier, demodulating the PCM data stream from the carrier, and generating a reconstructed audio signal from the PCM data stream.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The invention relates to digital communication and, more particularly, to the coding of audio and data signals in cordless telephones. 
     2. Description of the Related Art 
     Cordless telephones can use two basic types of audio transmission: analog or digital. Digital cordless telephones offer multiple advantages over analog cordless phones, but are typically more expensive to implement. Digital coding of the audio allows a series of “1”s and “0”s to be sent over the radio part of the cordless telephone. This allows the audio information to be securely transmitted and received. Digital coding offers a higher quality transmission because analog noise that occurs in the communication link is not added into the audio information, providing a low-noise link. The penalty for digital audio is the cost of implementation. A significant amount of circuitry is required to implement the digitizing and compression of the audio. 
     Narrow band digital cordless telephones have historically implemented digital audio using one of two methods: ADPCM and CVSD. CVSD (continuously variable slope delta-modulation) is a simple method for digitally encoding a voice signal. Because of the simplistic way the voice is digitized, however, quality suffers in this method. At practical data rates, the quality of CVSD digital voice is not at the level of a wired telephone. ADPCM (adaptive differential pulse code modulation) was the solution to the voice quality problems of CVSD, but came at a greater implementation cost penalty. ADPCM yielded voice quality equal to that of a wired telephone; however, the ADPCM digitizing technique requires a relatively complex implementation, driving up the cost to the end user. 
     The other issue in a digital narrow band cordless telephone that must be considered is the data rate, which is inversely related to the energy per bit the radio uses for transmission. In a narrow band cordless telephone with a fixed transmission power, higher data rates result in a shorter range due to the correspondingly lower energy per bit. CVSD has typically been implemented at 48 kb/s. ADPCM has typically been implemented at 32 kb/s. These numbers show that ADPCM generally provides a longer range than CVSD along with the previously stated voice quality advantage, but again, with a penalty in implementation cost. 
     Although various designs of digital cordless telephones are available, those designs have in many ways not adequately met the consumer&#39;s need for quality as well as economy. A digital cordless telephone that meets those expectations of consumers, thus, would provide significant improvement and advance in the technology. The consumer cordless telephone market demands lower and lower implementation costs at a higher quality level. As this happens, new trade-offs and approaches are needed. 
     SUMMARY OF THE INVENTION 
     Described herein is a cordless telephone system and method using PCM techniques for encoding an audio signal. The audio signal may be human voice, music, a modem signal, or any other analog signal in a predetermined frequency range. The PCM coding scheme provides distinct advantages over the previously used lossy coding schemes, such as ADPCM and CVSD. By using this efficiently implemented coding scheme, the system substantially reduces implementation costs, a significant consideration in cordless telephones designed for residential use and in other settings where cost is an important factor. Using a PCM encoding scheme, such as A-law or μ-law, provides a low cost digital telephone with excellent voice quality and a range that is acceptable for a low-cost digital cordless telephone. Further, since PCM does not degrade the audio signal quality, using it improves the quality of transmission for signals such as modem signals, another important factor for many end users. 
     This disclosure presents a communication system, such as a portable telephone, that has a handset and a base unit. The handset and base unit are coupled wirelessly, through an RF or IR link. The base unit receives an incoming telephone signal from a telephone connection and converts it to an incoming wireless PCM signal that is transmitted to the handset. The handset then converts incoming wireless PCM signal to an incoming audio signal that may be heard by a user. In the outgoing direction, the base unit receives an outgoing audio signal and in response generates an outgoing wireless PCM signal. The base unit receives the outgoing wireless PCM signal, converts it to an outgoing telephone signal, and provides the outgoing telephone signal to the telephone connection. The communication system may be configured to communicate with a telephone network through wired, fiber-optic, cellular, or wireless local loop links. Additionally, the links may carry analog or digital signals. 
     In one embodiment, the communication system is comprised in a wireless local loop system. The base unit communicates with a plurality of customer-specific portable units through wireless PCM links. The base unit is coupled to a central telephone office through an RF link. 
     A method is presented for communicating an audio signal from a transmitter unit to a remote receiver unit. The method includes steps of sampling an audio signal into a PCM data stream, modulating the PCM data stream onto a carrier, transmitting the carrier, receiving the carrier, demodulating the PCM data stream from the carrier, and generating a reconstructed audio signal from the PCM data stream. The PCM data stream may be encoded with linear, μ-law, or A-law quantization levels. In one embodiment of the method, the quantization scheme is selectable by the user, allowing the user to switch between logarithmically spaced quantization levels (μ-law or A-law) best suited for human voice, and evenly-space quantization levels (linear) that may provide better service for some modem signals. The carrier may be an RF carrier using amplitude-shift keying, frequency-shift keying, phase-shift keying, combinations of these, or other modulation schemes to convey the PCM data stream. Alternatively, the carrier may be an IR or visible-light signal transmitted through free space or through an optical fiber. Modulation schemes for the optical carrier include on-off keying (OOK), amplitude-shift keying, frequency-shift keying, and phase-shift keying, among others. 
     Further, this disclosure presents a transceiver with a PCM coder, an RF transmitter, an RF receiver, and a PCM decoder. The PCM coder receives a transmit audio signal and samples it to generate a PCM data stream. The RF transmitter modulates an RF carrier with the PCM data stream to generate an RF transmit signal. RF transmitter also transmits the RF transmit signal to a remote unit. The RF receiver receives an RF signal from a remote unit, and demodulates the RF received signal to extract a received PCM data stream. The PCM decoder receives the received PCM data stream and decodes it into a received audio signal. 
     The transceiver may be embodied in a cordless telephone handset, in which case also includes a microphone and a speaker that convert the audio signals to and from acoustic waves. The transceiver may also have a modem port that directly sends and receives the audio signals to and from a modem. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     Other objects and advantages of the invention will become apparent upon reading the following detailed description and upon reference to the accompanying drawings in which: 
     FIG. 1 is a representative view of a cordless telephone system; 
     FIG. 2 is a block diagram of the transceiver from FIG. 1; 
     FIG. 3 is a block diagram of the base unit from FIG. 1; and 
     FIG. 4 is a representative view of a mobile communications system. 
    
    
     While the invention is susceptible to various modifications and alternative forms, specific embodiments thereof are shown by way of example in the drawings and will herein be described in detail. It should be understood, however, that the drawings and detailed description thereto are not intended to limit the invention to the particular form disclosed, but on the contrary, the intention is to cover all modifications, equivalents and alternatives falling within the spirit and scope of the present invention as defined by the appended claims. 
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     A digital cordless telephone system generally includes two separate units: a handset and a base unit. These units generally communicate through a wireless link such as a radio or optical signal transmitted through free space (although the signal may alternatively be transmitted through a waveguide or an optical fiber). To transmit an audio signal or other analog signal through a digital link, the analog signal is digitized so that it can be represented by a stream of information symbols. Digitizing the audio signal involves sampling it so that values are recorded only at discrete points in time, and quantizing it, so that its amplitude is recorded as one of a discrete set of possible values. For human voice, the spectral power distribution has a bandwidth of approximately 3 kHz, so a sampling rate of 8000 samples/second (8 kS/s) records sufficiently many samples to reproduce the signal. With this sampling rate, 256 appropriately-chosen quantization levels are sufficient for producing a “toll-quality” digital audio signal. 
     According to the present invention, pulse code modulation (PCM) is the technique of transmitting the quantized samples as digital data through a communication channel. A 64 kbps (=8 kS/s×8 bits/sample) PCM data stream can faithfully convey a high quality digital audio signal. Slower—or more efficient—data rates may be used to carry the same audio signal. The typical spectral power distribution of human voice is highly peaked over the 200-800 Hz frequency range, with diminishing amplitudes at higher frequencies. Since the spectrum is not white over its frequency range, it can be deduced that a digitized voice signal can be compressed and decompressed without much loss of fidelity. This compression may be performed in conjunction with the digitization, using different forms of differential pulse code modulation (DPCM) such as delta modulation, “linear delta mod” (LDM), continuously variable slope delta modulation (CVSD) and various forms of adaptive differential pulse code modulation (ADPCM). Because of their reduced data rates, these coding schemes are commonly used in many communications systems for voice signals. These techniques, however, add to the complexity of a communication system by requiring additional steps in the transmission and reception of communicated signals. The increased complexity leads to a higher implementation cost for products employing these techniques. Further, when these techniques are used to reduce the required data rate, they are inherently “lossy” coding schemes, which makes them less useful for transferring “white” signals such as higher-speed modem signals. 
     In contrast, simple PCM voice coding without compression offers a number of advantages. PCM voice quality is excellent, typically even better than the voice quality of ADPCM. PCM coding is used in many places in the wired telephone networks, thus its quality is by definition at the toll-quality level of a wired telephone. PCM also has the advantage over ADPCM in that it can successfully pass higher-speed MODEM signals. 
     The quantization of the PCM coder is preferably logarithmic, in consideration of the logarithmic sensitivity of the human ear to acoustic signals. The A-law μ-law quantization schemes reflect this sensitivity and provide a lossless coding with high fidelity for human voice signals. Linear PCM coding, in contrast, is a simpler scheme that uses evenly-spaced quantization levels. Linear PCM coding may be preferable in some applications for modem signals, especially higher bit-rate modem signals. 
     The implementation cost of PCM is less than that of ADPCM. This consideration is important in the design of a cordless telephone for residential or other “low-end” customers. It enables a cost reduction over cordless telephones using ADPCM, and also provides better voice quality. The decreased cost and higher quality come at the penalty of a higher required data rate. As described above, PCM encoding of voice signals typically requires a data rate of 64 kb/s. The inverse relationship between data rate and range (for a fixed signal-to-noise ratio) results in a range that will be less than that of an ADPCM or CVSD system. This is the trade-off required to lower the cost of the digital telephone with high voice quality. 
     The decreased range, however, is an acceptable trade-off for many or most consumers. The range of current digital narrow band cordless telephones is generally much greater than a typical user in, for example, an apartment can take advantage of. A reduction in range on a lower priced telephone with excellent voice quality is thus not a hindrance to many end users. Indeed, such a system can offer a high quality digital cordless telephone to consumers who previously could not afford one. 
     The design of a digital cordless telephone using coding techniques such as A-law or μ-law enables a low cost digital telephone with excellent voice quality and a range that is acceptable for a low cost digital cordless telephone. 
     FIG.  1 : Digital Communication System 
     A representative digital communications system  100  is shown in FIG.  1 . Pictured here are a handset transceiver  110  and a base unit transceiver  120  that communicate through a wireless PCM link  165 . System  100  is preferably used in a cordless telephone system, though other communication systems, such as mobile radio units and links to remote instrumentation may also embody the design considerations described herein. 
     Handset  110  and base unit  120  each comprise a transmitter and receiver for wireless PCM signals communicated on wireless PCM link  165 . Base unit  120  receives an incoming telephone signal from telephone connection  122  and transmits information from the incoming telephone signal to handset  110  as an incoming wireless PCM signal. Handset  110  is a portable unit that re-creates an incoming audio signal (which may be a voice signal, a modem signal, or some other signal) from the incoming wireless PCM signal. In the opposite direction, handset  110  generates an outgoing wireless PCM signal in response to an outgoing audio signal. Base unit  120  receives the outgoing wireless PCM signal and converts it to an outgoing telephone signal for telephone connection  122 . 
     In one embodiment, handset transceiver  110  and base unit transceiver  120  also communicate with other transceiver units (not shown). Handset  110  preferably includes a switch  117  that toggles between two operation modes for handset  110 . In a “voice” mode, handset  110  communicates acoustic signals received through a microphone and produces by a speaker. In a “data” mode, handset  110  communicates data signals through a modem port  115 . 
     FIG.  2 : Handset Transceiver—Block Diagram 
     FIG. 2 is a block diagram of handset transceiver  110 . A transmitter  200 T receives an analog transmit audio signal  215 T from a microphone  210 T or from modem port  115  and converts transmit audio signal  215 T to an RF transmit signal  265 T. A receiver  200 R performs the inverse of this process: it receives an RF received signal  265 R and processes received signal  265 R to generate a received analog audio signal  215 R. The received audio signal  215 R is provided either to a speaker  210 R or to modem port  115 . RF transmit and receive signals  265 T and  265 R are communicated to base unit transceiver  120  through wireless PCM link  165  (from FIG.  1 ). 
     Transmit audio signal  215 T is preferably an analog signal with frequency components in the range 30 Hz-3 kHz. Thus, transmit audio signal  215 T is well-suited for carrying modem tones as well as human voice. A selector  225 T coupled to microphone  210 T, modem port  115 , and transmitter  200 T determines whether audio signal  215 T is received from microphone  210 T or from modem port  115 . Transmit audio signal  215 T also includes any dialing signals such as rotary-dial interrupts or DTMF (“touch-tone”) signals from a dialing unit (not shown). Another selector  225 R coupled to speaker  210 R, modem port  115 , and receiver  200 R determines whether received audio signal  215 R is provided to speaker  210 T or to modem port  115 . Selectors  225 T and  225 R comprise linked switches that connect transmitter  200 T and speaker  200 R either to microphone  210 T and speaker  210 R or to modem port  115 . The switching is preferably controlled a detector (not shown) that determines when an external modem is connected to the modem port. In another embodiment, selectors  225 T and  225 R are switched by a user-actuated switch or pushbutton. 
     Transmit audio signal  215 T is processed in transmitter  200 T by several circuit blocks coupled in sequence: a PCM coder  220 T, a scrambler  230 T, a line coder  250 T, and an RF transmitter  260 T. The first block of the transmitter is PCM coder  220 T that samples transmit audio signal  215 T to generate a digital signal  225 T representing the audio signal. 
     PCM coder  220 T is a lossless coder, that is, it digitizes transmit audio signal  215 T and uses a coding scheme to generate a PCM data stream  225 T that completely describes the sampled signal, to within the limits of the digital sampling and quantization. In one embodiment, PCM coder  220 T generates 8-bit samples of transmit audio signal  215 T at an 8 kHz sample rate so that PCM data stream  225 T is a 64 kbps digital signal of PCM bytes. The quantization levels used by PCM coder  220 T for digitizing transmit audio signal  215 T may be chosen according to a variety of protocols. For example, the levels evenly spaced in signal amplitude (linear PCM coding) or logarithmically spaced (μ-law or A-law PCM coding). In one embodiment, a user may switch between quantization schemes, providing flexibility for the transceiver to better communicate human voice or certain types of modem signals. 
     PCM data stream  225 T is preferably sent to scrambler  230 T, whose principal function is to smooth or “whiten” the spectrum of transmit signal  265 T, preferably by XORing the bits in PCM data stream  225 T with the output of a scrambling pattern generator (not shown). Scrambler  230 T preferably also buffers PCM data stream  225 T so that the scrambler output  235 T is grouped into transmit frames. This signal  235 T is then provided to a line coder  250 T that maps the scrambled digital signal into analog waveforms appropriate for the selected modulation technique, thereby producing a line-coded baseband transmit signal  255 T. The baseband signal  255 T is provided to an output stage  260 T. This output stage  260 T is an RF transmitter in which baseband transmit signal  255 T is upconverted to an RF transmit frequency, amplified, and radiated as transmit signal  265 T. Transmit signal  265 T may use amplitude-shift keying, frequency-shift keying, phase-shift keying, or combinations of these to convey PCM data stream  225 T. The implementation of line coder  250 T and output stage  260 T are designed in consideration of the modulation technique chosen for transmit signal  265 T. 
     Receiver unit  200 R comprises components that reverse the functions of the blocks in transmitter unit  200 T. The input stage  260 R is an RF receiver that receives received signal  265 R and downconverts it to produce a baseband received signal  255 R. A line receiver  250 R samples and decodes the baseband received signal. The sampling is preferably done at a high sample rate, with the resulting digital data stream decimated to generate a received digital signal  235 R with the same overall bit rate as the PCM data stream  225 T. Received digital signal  235 R is provided to a descrambler  230 R, which XORs it with the same whitening sequence that is used in scrambler  230 T, thereby recovering a received PCM data stream  225 R. Descrambler  230 R provides PCM data stream  225 R to a PCM decoder  220 R, which reconstructs received audio signal  215 R from the digital audio signal  225 R. 
     In another embodiment, the audio signal is not scrambled, and scrambler  230 T and descrambler  230 R are not included in the transmitter and receiver, thereby further simplifying the implementation of the system. 
     In another embodiment, handset transceiver  110  is an infrared (IR) or other optical transceiver that communicates with other units by IR or visible-light signals transmitted either through open space or optical fibers. In this embodiment of the transceiver, transmitter output stage  260 T is an optical source, such as an LED or a diode laser, that generates an optical transmit signal  265 T modulated with the baseband transmit signal  255 T. The optical modulation may be performed by binary on-off keying (OOK), amplitude-shift or frequency-shift keying, or phase-shift keying of a coherent optical signal. Receiver input stage  260 R is an optical detector, such as a photodiode, that receives an optical receive signal  265 R and in response generates an information-modulated baseband signal  255 R that is sampled and decoded in line receiver  250 R. 
     FIG.  3 : Base Unit—Block Diagram 
     FIG. 3 is a block diagram of base unit transceiver  120 . Base unit transceiver  120  comprises a transmitter  300 T and a receiver  300 R that function in a manner similar to that of handset transceiver  110  described above, except that instead of receiving and generating acoustic signals in a microphone and speaker, base unit transceiver  120  communicates on telephone connection  122 . 
     Transmitter  300 T receives an incoming audio signal  315 T from telephone connection  122  through a telephone port  310 . Transmitter  300 T converts incoming audio signal  315 T to an RF transmit signal  365 T (which is received by handset  110  as signal  265 R). Receiver  300 R receives an RF received signal  365 R (which is generated by handset  110  as signal  265 T) and processes received signal  365 R to generate an outgoing audio signal  315 R. Outgoing audio signal  315 R is provided to telephone connection  122  through telephone port  310 . RF transmit and receive signals  365 T and  365 R are communicated to handset  110  through wireless PCM link  165  (from FIG.  1 ). If telephone connection  122  is configured to carry a digital signal, telephone port  310  converts incoming and outgoing audio signals  315 T and  315 R from and to the appropriate digital format. 
     Transmitter  300 T preferably comprises components (such as a PCM coder  320 T, a scrambler  330 T, a line coder  350 T, and an RF transmitter  360 T) that perform the same operations as the corresponding components of transmitter  200 T in handset  110 . Similarly, receiver  300 R preferably includes components (such as a PCM decoder  320 R, a descrambler  330 R, a line receiver  350 R, and a receiver input stage  360 R) that function in the same manner as the corresponding components of receiver  200 R in handset  110 . 
     In one embodiment, telephone port  310  is configured to receive an analog telephone signal, such as a POTS (“plain-old telephone service”) or wireless local loop signal, from telephone connection  122 . In other embodiments, the telephone signal from telephone connection  122  is a digital signal or is included in a digital signal. The digital signal may be a DSL, ADSL, HDSL, HDSL 2 , other xDSL, ISDN, or T 1  signal, among others. Telephone port  310  is configured to convert the digital signal, or an audio portion of the digital signal, to incoming audio signal  315 T, and to likewise convert outgoing audio signal  315 R into the appropriate digital format for telephone connection  122 . 
     If telephone connection  122  is a digital signal, then incoming and outgoing audio signals  315 T and  315 R are preferably digital audio signals, converted to and from the PCM format by PCM coder  320 T and PCM decoder  320 R. 
     It is noted that in certain cases, such as when the telephone signal is an ISDN signal, the audio signal in the telephone signal is already in a PCM digital format. In such cases, the PCM audio signal may be directly used, obviating the need for the PCM coder  320 T and decoder  320 R. In these cases, telephone port  310  is coupled directly to scrambler  330 T and to descrambler  330 R (or to line coder  350 T and line receiver  350 R, if the scrambler is not implemented), and telephone port  310  provides and receives PCM audio signals without performing any analog-to-digital or digital-to-digital conversions. 
     In another embodiment, telephone connection  122  is a dedicated computer line, such as an Ethernet line. Telephone port  310  is then configured to extract a digital audio signal from data received on telephone connection  122  and to convert the digital audio signal into incoming audio signal  315 T. In this embodiment, telephone port  310  may be comprised in a separate conversion unit, such as a plug-in card for a home computer. 
     In one embodiment, the system is incorporated in a wireless local loop (WLL) network. In this embodiment, telephone connection  122  is a WLL link to a remote transceiver in a central telephone office (CO). Base unit transceiver  120  receives the WLL link from the CO and serves as a distribution point for a local area, such as an apartment building or a block of houses. Each apartment, house, or other “customer” in the local area has a handset transceiver  110  that is linked to base unit transceiver  120  through a wireless PCM link  165  (from FIG.  1 ). 
     In yet another embodiment, telephone connection  122  is an analog or digital cellular telephone link, and telephone port  310  receives and transmits audio signals  315 T and  315 R on the cellular link  122 . Such a system may be particularly useful in an mobile communications system, one embodiment of which is shown in FIG.  4 . 
     FIG.  4 : Mobile Communications Application 
     FIG. 4 illustrates an embodiment of the PCM transceiver in a mobile communications system. This embodiment enables a “hands-free” telephone system in an environment where a user may prefer not to carry a cellular-telephone handset. In this embodiment, the telephone signal is a cellular-telephone link  122 A received by a local base unit  120 A in an automobile. A hands-free unit  110 A is coupled to local base unit  120 A by a wireless PCM link  165 A. Hands-free unit  110 A includes a microphone and a speaker to performs the acoustic/electronic transduction for a user. 
     Hands-free unit  110 A may be a headset, as shown in the figure, or a speakerphone, or a combination speaker and portable microphone, or another unit that enables hands-free communication. Base unit  120 A may be placed under the driver&#39;s seat of the automobile, or in the automobile&#39;s trunk, thereby providing a measure of security against theft (especially if base unit  120 A and hands-free unit  110 A are “paired” units, restricted by electronic signature to operate only with each other). 
     The figure illustrates the system being used by a driver in an automobile, but it may also be adapted for use in trucking, in boating/marine environments, in aviation, and in business, residential, industrial, military, and other settings where a user may prefer not to be encumbered by a cellular handset.