Patent Publication Number: US-6671325-B2

Title: Wireless infrared digital audio system

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
     This application is a continuation of commonly-owned U.S. patent application Ser. No. 09/425,315 filed on Oct. 25, 1999, now U.S. Pat. No. 6,510,182, issued Jan. 21, 2003 and is incorporated herein by reference. 
    
    
     BACKGROUND OF THE INVENTION 
     RELATED PATENT APPLICATIONS 
     “A Wireless Infrared Digital Audio Transmitting System,” Ser. No.: 09/425,314, Filing Date: Oct. 25, 1999, assigned to the Same Assignee as the present invention. 
     “A Wireless Infrared Digital Audio Receiving System,” Ser. No.: 09/427,020, Filing Date: Oct. 25, 1999, assigned to the Same Assignee as the present invention. 
     1. Field of the Invention 
     This invention is generally related to electronic circuits and systems that transmit and receive digitally sampled analog signals. More particularly, this invention relates to electronic circuits and systems that transmit and receive digital audio signals. 
     2. Description of the Related Art 
     The transmission of analog signals between a source of the analog signals and a reproduction of the analog signals at an output transducer is well known in the art. U.S. Pat. No. 5,596,648 (Fast) describes a wireless infrared audio transmission system where infrared LED emitters are activated by a frequency modulated pulse wave transmitted as light to a receiver. The audio analog signal modulates the frequency modulated pulse wave. 
     U.S. Pat. No. 5,596,603 (Haupt, et al.) illustrates another device for wireless transmission of audio signals. Refer now to FIG. 1 for an overview of this structure. The analog source  5  provides a left channel L and a right channel R. The analog source  5  would be microphones, a FM tuner/receiver, or an analog recording media. The left channel L and the right channel R are inputs to analog-to-digital converters  15  and  20 . It is well known in the art that the analog sources can provide any number of channels. The left channel L and right channel R are chosen for illustration purposes. 
     Additionally, the analog signals from the analog source  5  can have been previously converted to digitized samples and then provided by the digital source  10 . The digitized samples of the analog signals are retained in a data buffer  25 . The digitized samples are then formatted in data frames in the data formatting unit  30 . In Haupt, et al. a data frame is 128 bits in length for each channel (left channel L or right channel R). The data frames are then transferred to the date modulator  35 . A carrier signal is then modulated with the data frames. 
     In the case of Haupt, et al., the data frames are changed from a 4 bit audio data to a 5 bit transmission data, which is used to activate and deactivate an infrared light emitting diode. The modulated carrier signal is transferred to a transmitter and then conveyed to the communication medium  45 . The infrared light is then radiated through the open atmosphere to a receiving light sensitive diode. In this case, the communication medium  45  is the open atmosphere. 
     It is well known that the transmitter  40  can produce radio frequency waves in addition to light. Further, the communication medium  45  can be either wire such as coaxial cable, twisted-pair cable or other forms of metallic (copper) inter-connection. Additionally, the communication medium  45  may be a fiber optic cable. 
     The receiver  50  will recover the modulated carrier signal from the communication medium  45 . Typically, a clocking or timing signal is included in the data frame and the modulated carrier signal. A clock extraction circuit  55  will develop the embedded clocking or timing signal and synchronize the receiving subsystem  100  with transmitting subsystem  95 . Classically, the clock extraction circuit  55  incorporates a phase locked oscillator, which can malfunction if there are errors in the transmitted modulated carrier signal. 
     The recovered modulated carrier signal is transferred to the demodulator  60  to extract the data frames. The data frames are then reformatted in the receive data formatter  65  to recreate the digitized samples of the analog signals. The recreated digitized samples are then transferred to the digital-to-analog converters  70  and  75  to reproduce the analog signals  80  and  85 . Alternately, the digitized samples of the analog data can be transferred  95  to external circuitry for further processing. 
     The wireless transmission as shown in FIG. 1 is subject to corruption of the digitized samples during transmission. For instance, noise from an electronically ballasted halogen lamp would completely breakdown recovery of the transmission of the modulated carrier signal. 
     A solution to the corruption of the modulated carrier signal is to provide a level of redundancy for the digitized samples. U.S. Pat. No. 5,832,024 (Schotz, et al.) shows the use of forward error correction codes using the well known Reed-Solomon Coding. This will eliminate errors of relatively short duration, but will not prevent disruption of the output analog signals  80  and  85  due to long term digitized sample corruption. 
     To eliminate longer corruption of the digitized samples Schotz, et al. employs a convolutional interleaving circuit to separate the digitized samples of the analog signal that would normally be transmitted together. This allows the greater probability that a longer term error can be to be corrected 
     If the error correction coding and the convolutional interleaving of the digitized samples cannot insure corrected digitized samples of the analog signals, the analog signal will be reproduced (especially in audio signals) as annoying cracks and pops in a speaker. To eliminate the cracks and pops, Schotz, et al. suggests that the digitized samples can be brought to a null level or muted. However, if the muting is activated suddenly, it is distracting and is annoying to the listener in an audio application. 
     U.S. Pat. No. 5,602,669 (Chaki) provides a digital transmitter-receiver that transmits a digital audio signal within a specified frequency band, and receives the specified frequency band. Chaki modulates a fundamental frequency using Quadrature Phase Shift Keying (QPSK). The QPSK modulated signal is transferred to an infrared emitter for transmission. 
     U.S. Pat. No. 5,420,640 (Munich, et al.) describes a memory efficient method and apparatus for synchronization detection within a digital data stream over a communication path. The digital data is arranged as a sequence of frames, each frame including a plurality of lines of data. The beginning of each frame is indicated by a frame synchronization word. The beginning of each line is indicated by a horizontal synchronization byte. An encoder, before transmission, interleaves the data. The decoder contains circuitry for locating the horizontal and frame synchronization data and contains circuitry for deinterleaving the digital data. Both the synchronization locating circuitry and the deinterleaving circuitry require access to a memory, but not at the same time. Therefore, a single memory is used with the synchronization recovery circuitry and deinterleaving circuitry alternately addressing the memory. The digital data stream of Munich, et al. pertains to video, audio and other related services of subscriber based television systems. 
     U.S. Pat. No. 5,745,582 (Shimpuku, et al.) teaches an audio signal transmitting and receiving system which can perform optical transmission of a digital format audio signal with small deterioration of the sound quality over the transmission path. The audio signal transmitting system has circuits to add an error correction signal to a digital audio signal. The digital audio signal with the error correction signal is then encoded and interleaved to generate an audio transmission signal. Repeating a digital control signal, which is to be used for the reproduction of the digital audio transmission signal, generates a continuous signal. A multiplexer combines the audio transmission signal and the continuous signal to generate a multiplexed signal. A modulation circuit then modulates a carrier signal similar to that described above with the multiplexed signal by a predetermined digital modulation method to generate a modulated signal within a predetermined frequency band. The modulated signal is transmitted by an optical transmission signal. A differential type QPSK modulation method creates the modulated signal preferably. Shimpuku, et al. further describes an audio signal receiving circuit for reproducing a digital audio signal and a digital control signal from the optical transmission signal. The audio signal receiving circuit has an optical receiver to convert the optical transmission signal to an electric reception signal. The modulated signal is then reproduced to permit demodulation of the reception signal by a digital demodulation method corresponding to the predetermined digital modulation method to reproduce the multiplexed signal. A separating circuit separates the audio transmission signal and the continuous signal from the multiplexed signal. The audio transmission signal is then deinterleaved and error correction based on the added error correction signal is performed to reproduce the digital audio signal 
     The digital source  10  is often a compact audio disk (CD), a Moving Picture Expert Group Audio Layer  3  (MP3) data file, a digital audio tape (DAT), a Digital Video Disk (DVD), or a digital satellite receiver (DSR). The format of the digitized samples from a digital source  10  commonly complies with the Sony/Phillips Digital Interface (S/PDIF). International standards that have developed from this standard are the Audio Engineering Society (AES) AES-3, the European Broadcasters Union (EBU) Tech. 3250-E, the Japanese Electronic Industries Alliance (EIAJ) CP-340, and the International Electronic Commission (IEC) IEC60958. While these standards are similar, they are not necessarily identical. However, the data format as shown in FIG. 3 is common for each standard. The allowed sampling frequencies or sampling rates of the audio analog signals to create the digitized samples are the 44.1 kHz for CD and MP3, 48 kHz for DAT and DVD, and 32 kHz for DSR. 
     Refer now to FIG. 2 to discuss the data format of the S/PDIF family of international standards. A frame consists of two subframes  200  and  205  containing the samples from an A channel or left channel and a B channel or right channel. Each subframe has a synchronizing preamble A SYNC and B SYNC. The synchronizing preamble identifies the contents of the subframe as being either a word containing a sample of the A channel at the beginning of a block  215 , the A channel within a block, or the B channel. 
     The digitized audio samples for channel A and channel B can contain up to 24 bits representing the amplitude of a sample of the analog audio signal. Normally for CD applications, only the 16 bits A 8  through A 23  are employed to convey the digitized audio samples. The bits AV and BV are the validity indicating if the digitized audio sample is in error. The bits AU and BU are user defined bits which when collected from many samples indicate running time, track number, etc. The bits AC and BC are channel status bits defining such information as emphasis, sampling rate, and copy permit. The bits AP and BP are parity bits for error detection to verify reception of the data samples. 
     The digitized audio samples are encoded using a commonly known biphase mark or Manchester coding technique. The samples are transferred serially at a rate of 2.8 MHz for a sampling rate of 44.1 kHz, 2 MHz for a sampling rate of 32 kHz, and 3.1 MHz for a sampling rate of 48 kHz. 
     A block of the digitized audio samples consists of 192 frames concentrated together. 
     U.S. Pat. No. 5,889,820 (Adams) describes a circuit for SPDIF-AES/EBU digital audio data recovery. The circuit decodes an input signal. The circuit includes a measurement sub-circuit having an input to receive a timing clock signal that is asynchronous with clocking of the input signal. The asynchronous timing clock signal measures the duration of a plurality of pulses received on the input signal in relation to frequency of the timing clock signal. A decode circuit decodes the input signal into digital data. The invention of Adams permits use of all digital components for decoding digital audio data encoding using biphase-mark encoded data according to the S/PDIF or AES/EBU standards. 
     SUMMARY OF THE INVENTION 
     An object of this invention is to provide a system for transmitting, receiving, recovering, and reproducing digitized samples of analog signals. 
     Another object of this invention is to conceal unrecoverable digitized samples of analog signals to maintain a level of fidelity in reproducing the digitized samples of the analog signal. 
     Further, another object of this invention is to transmit the digitized samples of the analog signals such that the probability of interference with the transmission and thus corruption of the digitized samples of the analog signals is minimized by transmitting the digitized samples as bursts shorter period than the time of the analog signals represented by the transmitted digitized samples. 
     Still further, another object of this invention is to receive the digitized samples of analog signal without synchronizing a receiving clock with a transmitting clock to capture the digitized samples of the analog signals. 
     Even still further, another object of this invention is to convert the digitized samples of the analog signals from an external source that has various sampling rates to digitized samples of the analog signals having a rate. 
     Again, another object of this invention is to softly mute the digitized samples of the analog signals when large groups of the digitized samples can neither be recovered nor concealed. 
     Another object of this invention is to track the long term difference between a transmit clock and a receive clock and to interpolate and decimate any underrun or overrun of the digitized samples of the analog signals within a group of digitized samples of the analog signals. 
     To accomplish these and other objects, a digital communication system for transmitting and receiving digitized samples of analog signals is comprised of a transmission subsystem for transmitting the digitized samples a communication medium to convey the transmitted digitized samples having various sampling rates, and a receiving system to receive and reproduce the transmitted digitized samples. The transmission subsystem receives the digitized samples having a variable sampling rate from an external signal source and then converts the digitized samples having a variable sampling rate to digitized samples with a fixed rate. The digitized samples have error correction codes generated to allow correction of any errors in the fixed digitized samples that may occur during transmission of the digitized samples. The digitized samples are formatted into groups of interleaved digitized samples with appended error correction codes. A preamble timing signal and a start signal is then appended to the group of interleaved digitized samples to form a transmit frame. A carrier signal is then modulated with the transmit frame and the modulated carrier signal is then transmitted to the communication medium. 
     The receiving subsystem is connected to the communication medium to receive and recover the modulated carrier signal. The modulated carrier signal is then demodulated to recover the transmit frame and to extract the group of interleaved digitized samples and the error correction codes from the transmit frame. The group of interleaved digitized samples with the error correction codes are checked, and the group of interleaved digitized samples with errors are corrected. If any of the group of interleaved digitized samples are uncorrectable, an estimated sample value of those uncorrectable digitized samples is created by interpolating from adjacent interleaved digitized samples to conceal any effect of the uncorrectable digitized samples. Any of the digitized samples that are not concealable or unrecoverable or are invalid are soft muted. The digitized samples are then transferred to a digital-to-analog converter for restoration of the analog signal. 
     The transmission subsystem has a sampled data receiver to receive the digitized samples of the analog signals from the external source of the digitized samples of the analog signals. A variable sampling rate converter is connected to the sampled data receiver to convert the digitized samples of the analog signals that are sampled at the one rate of the various sampling rates to digitized samples of the analog signals sampled at a fixed rate. A plurality of the digitized samples of the analog signals is retained in a data buffer. A data buffer controller is connected to the variable sampling rate converter and the data buffer to control the placement and removal of the plurality of digitized samples of the analog signals within the data buffer. An error correction code generator is connected to the data buffer controller to receive multiple digitized samples of the analog signals through the data buffer controller from the data buffer. The error correction code generator generates an error correction word that appended to the multiple digitized samples of the analog signals, and then returns the multiple digitized samples of the analog signals with the appended error correction word through the data buffer controller to the data buffer. The error correction code generator creates a Reed-Solomon error correction code with a the error correction code word that has a data block size of 238 bytes and one control byte and 16 parity bytes. A frame formatter is connected to the data buffer controller to receive an interleaved group of the multiple digitized samples of the analog signals with the appended error correction codes and appends a preamble timing signal and a start signal before the interleaved group of the multiple digitized samples of the analog signals to form a transmit frame. A pulse position modulator is connected to the frame formatter to receive the transmit frame and modulate a carrier signal according to a pulse position modulation with the transmit frame. A burst transmitter is connected between the pulse position modulator and the communication medium to convey a modulated carrier signal to the communication medium. The modulated carrier signal is transmitted as a burst within a short time period to minimize probability of interference on the communication medium. 
     The communication medium may be either wireless or wired and the modulated carrier signal may be transmitted as light or as Radio Frequency energy. The wired communication media may be either fiberoptic cable, coaxial cable, or two wire twisted pair cable. 
     The receiving subsystem has a receiver connected to the communication medium to sense and amplify the modulated carrier signal and to recover the transmit frame. A demodulator is connected to the receiver to demodulate the modulated carrier signal and extract the groups of interleaved multiple digitized samples of the analog signals with the appended error correction code. The demodulator is connected to a received data buffer to retain the group of interleaved multiple digitized samples of the analog signals with the appended error correction code. A received data buffer controller is connected to the demodulator and the received data buffer to control transfer of the groups of interleaved multiple digitized samples of the analog signals with the appended error correction code from the demodulator to the received data buffer. An error check and correction circuit is connected to the received data buffer controller to receive one group of the multiple digitized samples of the analog signals with the appended error correction code. The error check and correction circuit checks and corrects any errors that occur in transmission in the one group of the multiple digitized samples of the analog signals and then replaces the corrected group of the multiple digitized samples of the analog signals to the received data buffer. Any non-correctable digitized samples of the multiple digitized samples of the analog signals are identified for concealing. A block recovery circuit is connected to the received data buffer controller to receive and interpolate the non-correctable digitized samples of the analog signals to conceal an effect of the non-correctable digitized samples analog signals. A soft muting circuit is connected to the received data buffer controller to access those groups of the multiple digitized samples that were not correctly received and declared invalid and those of the multiple digitized samples of the analog signals with non-recoverable and non-concealable errors. The soft muting circuit, further, accesses those of the multiple digitized samples of the analog signals that are correct and adjacent to those of the multiple digitized samples of the analog signals that are invalid or with non-correctable and non-concealable errors. The soft muting circuit then applies a smoothing function to the multiple digitized samples of the analog signals to bring those of the multiple digitized samples of the analog signals that are invalid or with the non-correctable and non-concealable error to a null value. 
     The receiving subsystem has a jitter tracking circuit to compare the block transmission timing signal with a clock signal of the receiver subsystem to determine overrun and underrun of the contents of the group of interleaved multiple digitized samples of the analog signals with the appended error correction code. The block transmission signal indicates a boundary of groups of interleaved multiple digitized samples of the analog signals with the appended error correction codes. The number of words within each group of interleaved multiple digitized samples of the analog signals with the appended error correction codes must contains the correct number of digitized samples of analog signals. An interpolation and decimation circuit is connected to the jitter tracking circuit and the received data buffer controller to generate or eliminate digitized samples of the analog signals if the jitter tracking circuit indicates overrun or underrun of the contents of the group of interleaved multiple digitized samples analog signals. 
     An interface circuit is connected to the received data buffer controller to translate the digitized samples of the analog signals to a format acceptable by subsequent circuitry. 
     The digitized samples having variable sampling rates are sampled at sampling rates of 32 kHz, 44.1 kHz, and 48 kHz. 
     The transmission subsystem may have at least one analog-to-digital converter connected between the external source and the data buffer controller to receive the analog signals and to generate the digitized samples analog signals. The sampling rate of the analog-to-digital converter is approximately 48 kHz. An alternate sampling rate for the analog-to-digital converter is 44.1 kHz. 
     The interleaved group of the multiple digitized samples is comprised of a plurality of least significant bytes of the even designated digitized samples of the group of multiple digitized samples, a plurality of most significant of the even designated digitized samples, a first command byte, a first plurality of error correction parity bytes, a plurality of least significant bytes of the odd designated digitized samples, a plurality of most significant bytes of the odd designated digitized samples, a second command byte, and a second plurality of error correction parity bytes. 
     The carrier signal is modulated with a pulse positioned modulation by positioning of a pulse of the carrier signal within a period of the carrier signal according to a binary value of a plurality of bits within the transmit frame. The plurality of bits of this invention is two bits. 
     The digitized samples are encoded in a non-return to zero (NRZ) format. 
     The burst transmitter includes an infrared light emitting diode and a diode switching circuit connected between the pulse position modulator and the infrared light emitting diode to activate and deactivate the infrared light emitting diode with the modulated carrier signal. 
     The receiver comprises a light sensitive diode that receives light radiated from the infrared light emitting diode. 
     The demodulator demodulates the modulated carrier signal by oversampling the modulated carrier signal to determine an evaluation point of the modulated carrier signal to recover the transmit frame. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a block diagram of a transmitting and receiving system for digitized samples of analog signals of the prior art. 
     FIG. 2 is a diagram of the format of the S/PDIF interface. 
     FIG. 3 is a block diagram of a transmitting and receiving system for digitized samples of analog signals of this invention. 
     FIG. 4 is a block diagram of the transmitting system for digitized samples of analog signals of this invention. 
     FIG. 5 is a block diagram of the receiving system for digitized samples of analog signals of this invention. 
     FIG. 6 is a diagram of the format of a block of frames of the digitized samples of analog signals to be transmitted and received by the transmitting and receiving system of this invention. 
     FIG. 7 is a diagram illustrating the conversion of the non-return to zero coding to the pulse position modulation of this invention. 
     FIG. 8 is a diagram of the buffer addressing of the transmitter and receiver buffer of this invention. 
     FIG. 9 is a timing diagram of the operation of the transmitter buffer of this invention. 
     FIG. 10 is a timing diagram illustrating the over-sampling recovery of the received digitized samples of analog signals of this invention. 
     FIG. 11 is a flow diagram illustrating the method of transmitting digitized samples of analog signals of this invention. 
     FIG. 12 is a flow diagram of the method of receiving, recovering, concealing, and reproducing digitized samples of analog signals of this invention. 
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
     Refer now to FIG. 3 for a discussion of the structure of the transmitting and receiving system of this invention. A source  305  of an analog signal generates analog signals to the left channel L and the right channel R. The analog source  305  may be microphones, a radio receiver/tuner, or analog recording media. The analog signals from the left channel L and the right channel R respectively are the inputs to the analog-to-digital converters  315  and  320 . The analog-to-digital converters  315  and  320  sample the analog signals at a fixed rate. The fixed rate in the preferred embodiment is 48 kHz and in an alternate embodiment the fixed rate is 44.1 kHz. 
     The digital source  310  provides digitized samples of the analog signal that have been previously sampled and recorded or stored on a media such as a compact disk, digital audio tape, or other digital storage medium. The sampling rates of the digitized sample may differ from the fixed rate of the preferred embodiment. In the case of digital source  310 , complying with the industry standards that implement the S/PDIF format, the sampling rates will be, as described above, 44.1 kHz, 48 kHz, and 32 kHz. 
     The digitized samples of analog signals are transferred to the transmitter  395 . The transmitter  395  converts those digitized samples generated from differing sampling rates to digitized samples generated from the fixed rate. 
     The digitized samples are reorganized to interleave the digitized samples such that those digitized samples that are adjacent are separated such that they will be transmitted at different times. This separation minimizes the probability that any interference with the transmission prevents correction and reconstruction of the digitized samples. 
     Refer now to FIG. 8 for a discussion of the data structure of the digitized samples of this invention. In this invention, the data samples consist of two bytes (16 bits) of the possible data samples. The remaining byte (8 bits) of the S/PDIF format is discarded. The first group  810  of 238 bytes consists of the alternating least significant bytes of the left channel L (A channel) and the least significant bytes of the right channel R (B channel) of the even designated samples. The second group  815  of 238 bytes consists of the alternating most significant bytes of the left channel L and the most significant bytes of the right channel R of the even designated samples. The third group  820  of 238 bits consists of the alternating least significant bytes of the left channel L and the least significant bytes of the right channel R of the odd designated samples. The fourth group  825  consists of the alternating most significant bytes of the left channel L and the most significant bytes of the right channel R of the odd designated samples. As can be seen, the bytes of one digitized sample are separated by 238 bytes during a transmission. Further, two adjacent of the digitized samples are separated by at least 238 bytes. While this structure is adequate for the preferred embodiment, other interleaving patterns are possible and still are in keeping with the intent of this invention. 
     Returning to FIG. 3, the transmitter  395  generates error correction codes (ECC)  825  and  845  to be appended to the group of most significant bytes  810  and the group of most significant bytes  820  of the odd designated samples  805 . The ECC codes  835  and  845  are forward error correction codes employing the Reed-Solomon method of error correction encoding. The ECC codes of this invention  835  and  845  have a symbol size of 8 bits or one byte and will have a Galois field GF(2 m ) or GF(2 8 ). The block length (n) of the ECC codes  835  and  845  of this invention is 255 bytes (2 8 ). The number of correctable errors is chosen to be 8 and therefore the message size is 239 bytes, which is 238 of the most significant bytes  815  and  825  of the even samples  800  and the odd samples  805  and one command and control byte  830  and  840 . 
     The preferred embodiment allows correction of only the most significant bytes  815  and  820  of the digitized samples. It would be apparent to one skilled in the art that a different Galois field GF (2 m ) could be chosen and that a different number of correctable errors could further be chosen and still be in keeping with the intent of this invention. 
     The interleaved digitized samples with the appended ECC code of FIG. 8 are formed for serial transmission as shown in FIG.  6 . Each frame is divided into two subframes, an even subframe  610  and an odd subframe  615 . The even subframe  610  is structured as described in FIG. 8 as the group of even designated digitized samples  800  and the odd subframe  615  is structure as described in FIG. 8 as the odd designated digitized samples  805 . The ECC code  835  is appended to the even designated digitized samples to complete the subframe  610  and the ECC code  845  is appended to the odd designated digitized samples  805 . 
     A timing preamble  620  and  630  and a start flag signal  625  and  635  are appended respectively before the even subframe  610  and the odd subframe  615 . The timing preambles  620  and  630  each consist of up to 16 bytes of a unique pattern which identifies the beginning of each subframe  610  and  620 . The timing preamble  620  and  630  pattern is a unique pattern that violates the normal pulse position modulation hereinafter described. The unique pattern of the timing preamble  620  and  630  is sufficiently long to allow the receiver to identify the timing preamble  620  and  630  and “lock” onto the timing preamble  620  and  630 . The start flag signals  625  and  635  each consist of two bytes of a unique pattern indicating the beginning of the even subframes  610  and  615 . 
     The formatted groups of the digitized samples of the analog signals are then used to modulate a carrier signal within the transmitter. In the preferred embodiment, the modulation is a pulse position modulation (PPM) as shown in FIG.  7 . The plot  725  shows the possible combinations of voltage levels that comprise a basic raw non-return to zero encoded data of the digitized samples. Each bit time T bpm    700 ,  705 ,  710 , and  715  is divided into four phases φ 1 , φ 2 , φ 3 , φ 4 . The plot  720  shows the resulting pulse position modulation codes of this invention. The NRZ code (00) illustrated in the bit time  700  places a logic level 1 in the phase slot φ 1 ; NRZ code (01) illustrated in the bit time  705  places a logic level 1 in the phase slot φ 2 , the NRZ code (10) illustrated in the bit time  710  places a logic level 1 in the phase slot φ 3 ; and the NRZ code (11) illustrated in the bit time  715  places a logic level in the phase slot φ 4 . 
     The modulated carrier signal is then burst transmitted to a communication medium  345 . The modulated carrier signal is transmitted as light or a radio frequency (RF) radiated into the atmosphere in wireless communication or is transmitted also as light or a radio frequency transferred to a transmission cable in wired communication. If the modulated carrier signal is transmitted as light, it is transmitted on a fiberoptic cable. If the modulated carrier signal is transmitted as RF, it is transmitted on a cable such as coaxial cable or simple twisted pair two wire cable. 
     The  476  digitized samples that compose a frame of the digitized samples represent 119 samples of a stereo music presentation or 2.479 milliseconds of the presentation. The transmission frequency of the modulated carrier signal in the preferred embodiment is at 2.0 MHz. Thus the transmission of  476  digitized samples coupled with the timing preamble, the start flag signal, the ECC code word, and the command and control byte (493) byte) requires 1.972 milliseconds to complete. The transmitter will not be transmitting or will be idle for 20.5% of the time. This will minimize the probability that any burst of noise will interfere with the transmission. If the transmission frequency is increased or the period of a burst of the digitized samples decreased, the time of transmission decreases and the idle time increases, further improving the probability of having no interference. 
     The modulated carrier signal that is transmitted on the communication medium  345  is conveyed to the receiver  300 . The receiver will recover the modulated carrier signal by converting the light or RF signal to electrical signals. The modulated carrier signal is then demodulated to recover the transmitted frame of the groups of digitized samples of the analog signals. 
     The receiver of this invention does not extract a transmitted clock from the modulated carrier signal to synchronize the receiver  300  with the transmitter  395 . The receiver  300  has a clock that is independent of the transmitter clock. The receiver clock is specified to be nominally identical with the transmitter clock, but because of tolerancing and phasing differences between the two clocks, the received modulated carrier signal is not always received correctly and causes errors. 
     To minimize the errors, the receiver recovers the modulated carrier signal to extract the transmit frame of the digitized samples of the analog signal by oversampling the modulated carrier signal. Oversampling is sensing the logic level at a rate that is a relatively large multiple of the frequency of the carrier signal. When a relatively large number of oversampling results indicate a series of samples having one logic level after an opposite logic level in the received modulated carrier signal, a starting evaluation point is arbitrarily assumed and each successive evaluation occurs at the frequency of the modulated carrier signal after the arbitrarily assumed evaluation point. The evaluated data is compared with the expected timing preamble. If a mismatch occurs between the evaluated data and the expected timing preamble, the starting evaluation point is adjusted and the data reevaluated. This process is repeated until the timing preamble is recovered from the received modulated carrier signal. 
     If the total timing preamble is then detected correctly, the timing of the recovering is determined to be locked. Once the receiving timing is locked to the received modulated carrier signal, the start flag signal is sought from the received modulated carrier signal. If the start flag signal is not found, the frame is declared invalid and brought to a null level. If the start flag signal is recovered from the received modulated carrier signal, the transmit frame is then recovered from the received modulated carrier signal. 
     Refer to FIG. 11 for a more complete overview of the selection of the evaluation point to determine the locking of the timing of the receiver  300  with the received modulated carrier signal  750 . The recovery clock  755  has a frequency or rate of sampling that is, for the preferred embodiment, a factor of six times the frequency of the carrier signal  760 . The timing clock  755  sets the sampling time of the modulated carrier signal  750 . The recovered data  765  is the logic level of the modulated carrier signal  750  at the sampling time of the timing clock  755 . As described above, the first logic level after an opposite logic level (logic level 0 in this case) starts the count to establish the arbitrary evaluation point  750 . The successive evaluation points will be at the rate of the carrier signal  760  or in this case every sixth occurrence of the timing clock  755 . 
     It is apparent to those skilled in the art that the frequency of the timing clock  755  can be modified to a multiple factor of the frequency of the carrier signal and still be in keeping with the intent of this invention. 
     The recovered transmit frame is then demodulated in the receiver  300  to extract the raw non-return encoded date of the digitized samples from the pulse position modulated code of the transmit frame. 
     The pulse position modulated code of the transmit frame is reconverted as explained in FIG. 7 to the raw non-return encoded data of the digitized samples to recover the transmit frame. 
     The interleaved digitized samples of the analog signals and the ECC codes are then checked, and any of the interleaved digitized samples that have been corrupted during transmission are corrected. 
     The ECC check and correction employs Reed-Solomon Error Correction methods that are well known in the art. In the preferred embodiment, 8 symbols or bytes are correctable in each of the group of 238 most significant bytes  810  and  820  of FIG. 8 plus one command and control byte. The method to accomplish the ECC checking and correcting begins with calculating the syndromes indicating any errors in the groups of most significant bytes  810  and  820 . For those groups of most significant bytes  820  and  820  with error, Berlekamp&#39;s iterative algorithm is invoked to find the error locator polynomial. The Chien search method is then used to locate the roots of the error locator polynomial and the Forney algorithm is used to compute the error magnitudes to reconstruct the correct data. If there are more errors than are correctable by the Reed-Solomon Error Correction Method, in the case of the preferred embodiment 8 bytes, those non-correctable groups of the most significant bytes  810  and  820  are identified for further processing to conceal those errors. 
     The digitized samples of analog signals that are identified as having uncorrectable errors and their adjacent correct digitized samples are then combined to interpolate an estimate of the magnitude of the sample of the analog signal. It is well known that generally adjacent samples do not vary greatly in magnitude and that interpolating between adjacent samples will conceal any non-recoverable errors. In audio applications, the perception of the sound will not be affected, even though there may be some high frequency distortion. 
     If any frames of the digitized samples of the analog signals have uncorrectable and non-concealable errors, or if frames of the received modulated carrier signal are declared invalid, those frames are soft muted to prevent any “clicks” from unrecoverable errors. The soft muting employs a Hanning window to weight the frames of the digitized samples of analog signals adjacent to those frames that have uncorrectable errors and non-concealable errors or that are invalid to weight the amplitude of the adjacent frames to slowly decrease the amplitude of the analog signal to a null value. 
     Further, muting is achieved by programming a delay to null subsequent digitized samples of the analog signals. The extended nulling of the subsequent digitized samples of the analog signals avoid repetitive interference from such sources as infrared remote controls for television, video tape recorders, and audio systems. The infrared remote controls send a burst of data that can last from 0.2-1.0 seconds. 
     Each frame of the digitized samples of the analog signal must maintain a constant number of digitized samples. Even with the method of oversampling as described above, the differences between the clock of the transmitter  395  and the receiver  300  causes there to be an “overrun” (more digitized samples received in a frame) or “underrun” (fewer digitized samples received in a frame) of the digitized samples received in a transmit frame. An overrun causes more digitized samples to be in a transmit frame while an underrun causes fewer digitized samples to be received in a transmit frame. The receiver tracks the jitter or misalignment between the frequency of the carrier signal and the frequency of the transmit frame to detect any of the frames of the digitized samples having either an overrun or underrun. When an overrun or underrun occurs, the receiver  300  interpolates or decimates the frame of digitized samples to insure the constant number of digitized samples in a frame. 
     The frames of the digitized samples of the analog signals are transferred  390  to an external circuitry for further processing or are formatted to the appropriate order to be transferred to the digital-to-analog converters  370  and  375 . Those digitized samples of the analog signal  370  of the left channel L are transferred to the digital-to-analog converter  370  that in turn reproduces the analog signal  380  for the left channel L. Those digitized samples of the right channel R are transferred to the digital-to-analog converter  375  that in turn reproduces the analog signal  385  for the right channel R. 
     Refer now to FIG. 4 for a discussion of the transmitter system  395  of this invention. The analog source  305 , as described above, provides the left channel L and right channel R analog signals. The analog-to-digital converters  315  and  320  respectively sample the left channel L and right channel R analog signals and generate the digitized sample of the analog signals. Additionally, as described above, the digital source  310  provides digitized samples of the analog signal. 
     The digitized samples of the analog signals from the digital source  310  are transferred to the digital receiver  400 . The digital receiver  400  acquires the digitized samples at the sampling rate at which they are retained within the digital source (e.g. 44.1 kHz for CD and MP3, 48 kHz for DAT and DVD, and 32 kHz for DSR). The digitized samples of the analog signals are transferred to the variable sampling rate (VSR) converter  405 . The VSR converter  405  modifies the digitized samples of the analog signals to digitized samples of the analog signals sampled at a fixed rate. In the preferred embodiment, the fixed rate is 48 kHz. Thus, the digitized samples for CD are modified from a sampling rate of 44.1 kHz to that of 48 kHz. An alternate embodiment employs a fixed rate of 44.1 kHz. 
     A method to accomplish this is to send the digitized samples of the analog signals to a digital-to-analog converter to reproduce the original analog signal. The reproduced analog signal is then an input to an analog-to-digital converter that samples the reproduced analog signal at the fixed rate or 48 kHz for the preferred embodiment. In the alternate embodiment, the fixed rate may be another frequency such as 44.1 kHz. 
     The digitized samples of the analog signal are then transferred from the analog-to-digital converters  315  and  320  or the variable sampling rate converter through the transmit data buffer controller  410  to the transmit data buffer  415 . 
     The transmit data buffer controller  410  controls the access to and the retrieving from the transmit data buffer  415  of the digitized samples of the analog signals. The transmit data buffer  415  is a random access memory (RAM). In the preferred embodiment, the transmit data buffer  415  is a static RAM. However, other memory structures such as dynamic RAM (DRAM) or synchronous DRAM can be used for the transmit data buffer  415  and still be in keeping with the intent of this invention 
     Refer now to FIG. 8 to discuss the structure of the transmit data buffer  415  and the assignments of the component bytes of the digitized samples of the analog signals. The transmit data buffer  415  is segmented into groups of buffer blocks. In the preferred embodiment, there are three buffer blocks, buffer  1   850 , buffer  2   870 , and buffer  3   890 . The digitized samples of the analog signals placed in each buffer, as described above, with 238 least significant bytes of the even designated digitized samples  810  of the analog signals occupy the first address segment  852  of the first buffer block  850 . The 238 most significant bytes of the even designated digitized samples  815 , the command and control byte  830 , and the ECC code  840  occupy the second address segment  854  of the first buffer block  850 . The 238 least significant bytes of the odd designated digitized samples  820  occupy the third address segment  856  of the first buffer block  850 . The most significant bytes of the odd designated digitized samples  825  occupy the fourth address segment  858  of the first buffer block  850 . This structure enables the interleaving of the digitized samples to allow the correction and concealment of errors during transmission. 
     Additional frames of the digitized samples are likewise retained in the buffer block  2   870  and buffer block  3   890  in the same organization as described for buffer block  1   850 . This allows operations (receiving from the external sources  305  and  310 , ECC encoding, and transmitting) to be accomplished in parallel. 
     Each of the groups of most significant bytes  815  and  820  and the command and control bytes  830  and  840  are read from the transmit data buffer  415  transferred through the transmit data buffer controller to the Reed-Solomon Error Correction Encoder  420 . The ECC encoder  415  generates the symbol code work with a Galois Field GF (2 8 ). Each symbol is a byte, as above described, and the code allows 8 bytes to be corrected. 
     The ECC codes  840  and  845  are each appended respectively to the most significant bytes  815  and  825  of the even and the odd designated digitized samples  800  and  805  in the second address segment  854  and the fourth address segment  858 . 
     The interleaved digitized samples are retrieved from each buffer block  850 ,  870 , and  890  of the transmit data buffer  415  individually transferred to through the transmit data buffer controller  410  to the frame formatter  425 . The frame formatter appends the timing preambles  620  and  630  and the start flag signals  625  and  635  of FIG. 6 to assemble the subframes of each transmit frame  605  of the interleaved group of digitized samples of the analog signals. 
     FIG. 9 shows the access and retrieval pattern of the transmit data buffer  415  of FIG. 4 over time. The first group of digitized samples from the analog-to-digital converter  305  or the digital source  310  are stored in the first buffer block  850  at the first time segment  900 . The second group of digitized samples from the analog-to-digital converter  305  or the digital source  310  are stored in the second buffer block  870  during the second time segment  905 . During the second time segment  905 , the ECC encoder  420  accesses the first buffer block  850 , generates the ECC codes  840  and  845 , and stores the ECC codes  840  and  845  in the first buffer block  850 . In the third time segment  910 , a third group of digitized samples is stored in the third buffer block  890 , the ECC codes are generated and stored in the second buffer block  870 , and the first group of digitized samples with the appended ECC codes are transferred to the frame formatter  425  and on for transmission. During the fourth time segment  915 , a fourth group of digitized samples are placed in the first buffer block  850 , the group of digitized samples in the second buffer block  870  are sent on for transmission, and the ECC codes for the group of digitized samples in the third buffer block  890  are generated and stored back to the third buffer block  890 . 
     This pattern or storing and accessing the transmit data buffer  415  continues for all following segments  920 ,  925  and forces simultaneous access and storing to the buffer blocks  850 ,  870  and  890 . The transmit data buffer controller  410  must appropriately arbitrate the accesses and storing to ensure there are no conflicts. 
     From the frame formatter  425 , the formatted transmit frame is transferred to the pulse position modulator  430 . The pulse position modulator  430  encodes each pair of bits of the transmit frame as described above in FIG.  7 . In the preferred embodiment, the frequency of the carrier signal modulated by the transmit frame is at least 2.0 MHz. 
     As described above, the period of time of the audio analog signal represented by the transmit frame is 2.479 milliseconds and with the carrier signal being 2 MHz, each transmit frame has a duration of 1.972 milliseconds. As described above, the burst transmitter  435  is activated to transmit the modulated carrier signal. In the preferred embodiment of this invention, the burst transmitter  435  is a switching circuit that activates and deactivates a light emitting diode  445  to send light to the atmosphere which acts as the communication medium  345 . 
     As stated above, the communication medium  345  may be wired or wireless. The wired communication medium may be fiber optic cable, coaxial cable, or a two wire twisted pair cable. The burst transmitter  435  may transmit either light or RF signals into the wired or wireless communication medium  345 . 
     The clock circuit  440  provides the necessary timing signals to the analog-to-digital converters  315  and  320  to ensure the fixed rate of 48 kHz (or the alternate fixed rate of 44.1 kHz). The clock circuit  440  provides the necessary timing signals to convert those digitized samples of the analog signal taken at rates other than the fixed rate to be converted to the digitized samples of the analog signal sampled at the fixed rate. 
     The clock circuit  440  generates the 2 MHz carrier signal and transfers it to the pulse position modulator  340  to create the modulated carrier signal that is the input to the burst transmitter  435 . 
     The clock circuit  440  is fixed frequency timing generation circuit that eliminates the need for a phase locked loop oscillator at either the transmitter  395  or the receiver  300  of FIG.  4 . 
     Refer now to FIG. 5 for a description of the receiving system  300  of this invention. The modulated carrier signal is transferred onto the communication medium  345  to a receiver  505  that recovers the modulated carrier signal. In the preferred embodiment, the light transferred through the atmosphere impacts upon a light sensitive diode  500 . Changes in the light sensitive diode  500  are sensed and amplified in the receiver  505  to recover the modulated carrier signal. The receiver  505  locks on to the modulated carrier signal and completes the recovery by the oversampling of the modulated carrier signal to capture the groups of the digitized samples of the analog signals as above-described in FIG.  11 . The demodulator  510  converts the pulse position modulated digitized samples to the raw non-return to zero encoded data of the digitized samples as above-described in FIG.  7 . 
     The demodulated digitized samples now have the frame format of the interleaved digitized samples of FIG.  8  and are transferred from the demodulator  510  through the receive data buffer controller  515  to the receive data buffer  520 . 
     The receive data buffer  520  is structured as the transmit data buffer  415  of FIG.  4 . The receive data buffer  520  is a random access memory, which, in the preferred embodiment of this invention, is a static RAM. However, other memory structures such as DRAM or synchronous DRAM can be used as the receive data buffer  520  and still be in keeping with the intent of this invention. 
     The structure of the receive data buffer  520  is identical to that of the transmit data buffer  415  as described in FIG.  8 . The receive data buffer  520  has the groups of buffer blocks  850 ,  870 , and  890 . Each of the buffer blocks  850 ,  870 , or  890  has address segments  852 ,  854 ,  856 , and  858  to retain the frame of the interleaved digitized samples of the analog signal. 
     The most significant bytes of the even designated samples of the analog signal  815  and the most significant bytes of the odd designated samples of the analog signal  825  with the respective appended ECC codes  840  and  845  are retrieved by the receive data buffer controller  515  from the receive data buffer  520  and transferred to the Reed-Solomon EEC decoder  525 . The Reed-Solomon EEC decoder  525  checks and corrects the most significant bytes of the even and odd designated digitized samples  815  and  825  using the Reed-Solomon ECC method as described above. The corrected most significant bytes of the even and odd designated digitized samples  815  and  825  are stored through the receive data buffer controller  515  to the receive data buffer  520 . Those of the even and odd most significant bytes of the digitized samples having uncorrectable errors (more than 8 bytes in error as described above) are identified for further processing to conceal the effect of the errors on the reproduced analog signal. 
     Those of the digitized samples identified as having uncorrectable errors are retrieved with their adjacent correct digitized samples of the analog signals from the receive data buffer  520  and transferred through the receive data buffer controller  515  to the block recovery circuit  530 . The block recovery circuit  530  interpolates an estimate of the uncorrectable digitized sample from the adjacent digitized samples of the analog signal to conceal the uncorrectable error in the digitized sample. The preferred embodiment employees a linear interpolation between adjacent digitized samples to insure an ease of hardware implementation. It is in keeping with the invention to use more complex interpolation methods. 
     The interpolated estimate of a correct magnitude of the uncorrectable digitized sample is transferred from the block recovery circuit  530  through the receive data buffer controller  515  to be stored in the location of the uncorrectable digitized samples with the receive data buffer  520 . 
     Any of the digitized samples that have errors that are neither correctable nor concealable are identified for soft muting. 
     Those buffer blocks segments containing invalid frames or uncorrectable and non-concealable digitized samples are transferred to the soft muting circuit  535  with a number of adjacent frames of the digitized samples of the analog signal. The soft muting circuit  535  applies a Hanning window to the frames of the digitized samples to weight the frames adjacent to the invalid frames or frames with uncorrectable and non-concealable errors to eliminate any “clicking” noise as described above. 
     The jitter tracking circuit  545  compares a block transmission timing signal with sampling timing of the receiver  505  to identify any misalignment between the block transmission timing signal and the sampling timing of the receiver  505  that indicates the above-described overrun or underrun of the digitized samples with a frame. The block transmission timing indicates a boundary between groups of adjacent frames of the digitized samples of the analog signal. The synchronization circuit  540  provides any adjustment of the sampling timing within the receiver  505  if necessary. The synchronization circuit  505  further retrieves a frame of the digitized samples and interpolates or decimates any underrun or overrun of digitized samples with the frame to provide the correct number of digitized samples of the analog signal within the frame. 
     Each frame of the digitized samples of the analog signal is then transferred from the receive data buffer  520  through the receive data buffer controller  515  to the data out interface  550 . In the preferred embodiment, the data out interface is the I 2 S interface that is known in the art for digital audio transmission. The data out interface  550  transfers the digitized samples of the analog signals to the digital-to-analog converters  370  and  375 . The digital-to-analog converters  370  and  375  reproduce the analog signal out  380  for the left channel L and the analog signal out  385  for the right channel R. The data out interface also provides the frame of digitized samples in a digital format  390  for further processing by external circuitry. 
     One implementation of the preferred embodiment of the receiving system  300  is as a portable remote system for headphones to reproduce audio signals. In this application, the receiving system must be disabled when not in use. The power management circuit  565  will sense when no frames of digitized samples of the analog signal are received for a relatively long period. The power management circuit  565  then removes the power supply voltage source from the receiving system. When this happens, the soft muting circuit  535  is engaged to prevent noise during the disabling of the power supply voltage source. 
     The receiving clocking circuit  560  provides the oversampling timing signal to the receiver  505  to recover the modulated carrier signal and to lock the receiving system  300  to ensure the recovery of the modulated carrier signal. As described prior, the receiving clock circuit  560  is nominally identical to the clock of the transmitting system  440  of FIG.  4 . Any difference in the receiving clock circuit  560  and the transmitting clock circuit  440  are the tolerance and phase differences of the two circuits and is tracked and corrected in the jitter tracking circuit  545  and the synchronization circuit  540 . 
     The command and control byte  830  and  840  of FIG. 8 contains command and control data that is accessible in the receive buffer  520  of FIG.  5 . The command and control circuit  555  has an external connection (not shown) to an interface to external circuitry. The command and control circuit accesses the command and control circuit bytes in the receive buffer, then decodes the command and control byte and transfers appropriate command and control signals to the external circuitry. In the preferred embodiment the external connection is a serial interface to provide volume control to a remote speaker or for text transmission to a panel display. 
     As is known by one skilled in the art, many of the functions of the transmitting system  395  of FIG.  4  and the receiving system  300  of FIG. 5 could be incorporated as methods executed within a computer system such as a microprocessor, a digital signal processor, or a microcontroller. Refer now to FIGS. 11 and 12 for a review of the method for transmitting digitized samples of an analog signal, receiving the digitized samples of the analog signal and reproducing the analog signal. 
     The method of transmitting begins receiving  1100  of the digitized samples of the analog signal. The digitized samples have the format as described for the international standards that define the S/PDIF format. Those digitized samples that are collected at a sampling rate other than the fixed rate of this invention must be converted  1115  to digitized samples collected at the fixed rate. In the preferred embodiment the fixed rate is 48 kHz and in the alternate embodiment the fixed rate is 44.1 kHz. 
     Alternative to receiving  1100  the digitized samples from a digital source such as a CD, MP3, DAT, DVD, etc., is receiving  1110  the analog signal and performing an analog-to-digital conversion  1120  at the fixed rate to generate the digitized samples of the analog signal. The analog signal may have multiple channels. For instance, the analog signal of stereo audio has a left channel and a right channel. Each channel is separately converted  1120  from analog-to-digital digital to form a left channel set of digitized samples and a right channel set of digitized samples. 
     The digitized samples of the analog signal are stored  1125  in a buffer and organized into the interleaved structure of FIG.  8 . As described in FIG. 8, the least significant bytes of the even designated digitized samples are placed in a first segment of the buffer. The least significant bytes of the left channel set of digitized samples are alternated with the right channel set of digitized samples within the first segment. The most significant bytes of the alternated left channel and right channel set of even designated digitized samples are placed in the second segment of the buffer. The least significant bytes of the alternated left channel and right channel set of the odd designated digitized samples are placed in a third segment of the buffer. The most significant bytes of the alternated left channel and right channel of the odd designated digitized samples are placed in a fourth segment of the buffer. This structure interleaves the digitized samples such that adjacent samples are separated to reduce the probability of an error condition totally disrupting groups of digitized samples and the errors that occur can be corrected or concealed. To allow parallel processing of the digitized samples, the buffer has multiple segments to retain multiple frames of the interleaved groups of digitized samples. 
     An error correction code is generated  1130  for the group of digitized samples and appended to the groups of digitized samples within the buffer. In the preferred embodiment of this invention, the ECC code is generated  1130  for the second segment of the buffer or the most significant bytes of the even designated samples and for the fourth segment of the buffer or the most significant bytes of the odd designated samples. 
     The ECC generation  1130  employs a Reed-Solomon method on a codeword of 255 bytes have a Galois Field GF (2 8 ) where 238 of the most significant bytes of the digitized samples and one command and control byte are the data to be protected. The ECC code is 16 bytes and provides correction up to 8 bytes within the code word. 
     Each frame of the groups of interleaved digitized samples are formatted  1135  to form the transmit frame. The transmit frame is structured as shown in FIG.  6 . As described in FIG. 6, a first timing preamble and a first start flag signal are appended before the even designated digitized samples to form a first subframe. A second timing preamble and a second start flag signal are appended before the odd designated digitized samples to form a second subframe and to complete a frame. 
     The transmit frame then pulse position modulates  1145  a carrier signal. The pulse position modulation  1145  converts the raw non-return to zero encoding of the digitized samples to the pulse position coding as shown in FIG.  7 . In the preferred embodiment, the carrier signal has a frequency of at least 2 MHz. The positioning of the pulse is determined by the value of pairs of bits of the transmit frame. 
     The modulated carrier signal then activates a driver circuit to transmit  1150  the modulated carrier signal on a communication medium. As described above, the communication medium may be wired or wireless, transmitting either light or RF energy. The wired communication medium may be fiber optic cable, coaxial cable, or common two wire twisted pair cable. 
     The burst transmitter  1150  is transmitting the entire modulated carrier signal for a relatively short period when compared with the period of the analog signal included in the frame of the digitized samples. In the preferred embodiment, 238 samples of the left channel and the right channel are included in each frame. Since the samples of the left channel and the samples of the right channel are created simultaneously, 119 samples of the analog signal or 2.479 milliseconds of the analog signal are transmitted. With the frequency of the carrier signal being 2 MHz, each frame is transmitted in 1.972 milliseconds. The transmission is then idle 20.5% of the time. This minimizes any effect of burst noise in the transmission medium and improves the probability that the transmission is received  1155 . If the frequency of the carrier signal increases or the period of the burst of the digitized signals decreases, the idle time will increase accordingly. 
     The modulated carrier signal is received  1155  by sensing and amplifying the modulated carrier signal. In the preferred embodiment, the modulated carrier signal is transmitted  1150  as infrared light created by activating and deactivating a light emitting diode to radiate the infrared light to the atmosphere. A light sensitive diode receives  1155  the infrared light, converts the infrared light to electrical impulses that are sensed and amplified. The received modulated carrier signal is recovered by oversampling the received modulated carrier signal to determine a position to lock onto to capture the modulated carrier signal. The timing preamble is tested to find the correct lock position to correctly retrieve the timing preamble, the start flag signal sensed, and the frame of the groups of interleaved digitized samples are recovered. 
     The recovered modulated groups of the interleaved digitized samples are demodulated  1160  to recover raw non-return to zero encoding of the digitized samples. 
     The frames of the digitized samples are stored  1170  in a buffer as described in FIG.  8 . The most significant bytes of the even designated digitized samples and the most significant bytes of the odd designated digitized samples and their respective ECC codewords have an ECC check and correction  1175  performed to find and repair any errors to the digitized samples within a frame. The ECC check and correction  1175  is the known Reed-Solomon method as described above. 
     Any of the groups of most significant bytes of the even designated or odd designated digitized samples that cannot be corrected are identified to be recovered  1185  by concealing the error. The groups are tested  1180  for having errors. If there are errors, the digitized samples with errors and the adjacent digitized samples that are error-free are examined and an estimate of the digitized sample in error is generated by linearly interpolating between the adjacent digitized samples that are error-free. 
     The corrected or recovered digitized samples are then stored  1190  in the buffer in the location containing errors. Any unrecoverable errors are identified for soft muting. 
     The frames of the digitized samples are checked  1195  for invalid data, unrecoverable errors  1200 , and for synchronization  1215  and conveyed to a digital-to-analog converter to reproduce  1225  the analog signal. If the frames of the digitized samples are invalid  1195  or unrecoverable  1200 , the invalid or unrecoverable frames and adjacent correct frames of the digitized data are soft muted  1205 . Soft muting  1205  applies a Hanning window to the invalid or unrecoverable frames of digitized samples and to the adjacent correct frames to weight the adjacent correct frames to slowly bring to null the reproduced analog signal. This eliminates an annoying effect that occurs if only the invalid or unrecoverable frame is muted in an audio application. 
     During the recovery  1155  and demodulation  1160  of the modulated carrier signal, the difference between the frequency of the evaluation point of the oversampling clock and the carrier signal is tracked to determine synchronization. The jitter tracking  1210  is tested  1215  to ensure there is no overrun or underrun of the number of samples. If there is an overrun or underrun, the frames of digital data are decimated or interpolated to synchronize  1220  the digitized samples before they are converted  1225  to reproduce the analog signals. The decimated or interpolated digitized samples insure that there are the correct number of digitized samples in each frame. 
     While this invention has been particularly shown and described with reference to the preferred embodiments thereof, it will be understood by those skilled in the art that various changes in form and details may be made without departing from the spirit and scope of the invention.