Patent Abstract:
A digital audio transmitter system capable of transmitting high quality, wideband speech over a transmission channel with a limited bandwidth such as a traditional telephone line. The digital audio transmitter system includes a coder for coding an input audio signal to a digital signal having a transmission rate that does not exceed the maximum allowable transmission rate for traditional telephone lines and a decoder for decoding the digital signal to provide an output audio signal with an audio bandwidth of wideband speech. A coder and a decoder may be provided in a single device to allow two-way communication between multiple devices.

Full Description:
CROSS REFERENCE TO RELATED APPLICATIONS 
     This is a continuation of Ser. No. 09/595,521, filed Jun. 16, 2000, issued as U.S. Pat. No 6,373,927 which is a continuation of Ser. No. 08/988,709, filed Dec. 11, 1997, issued as U.S. Pat. No. 6,128,374, which is a continuation of Ser. No. 08/419,199, filed Apr. 10, 1995, issued as U.S. Pat. No. 5,706,335. 
    
    
     FIELD OF THE INVENTION 
     The present invention relates generally to an apparatus and method for transmitting audio signals and pertains, more specifically, to an apparatus and method for transmitting a high quality audio signal, such as wideband speech, through a transmission channel having a limited bandwidth or transmission rate. 
     BACKGROUND OF THE INVENTION 
     Human speech lies in the frequency range of approximately 7 Hz to 10 kHz. Because traditional telephone systems only provide for the transmission of analog audio signals in the range of about 300 Hz to 3400 Hz or a bandwidth of about 3 kHz (narrowband speech), certain characteristics of a speaker&#39;s voice are lost and the voice sounds somewhat muffled. A telephone system capable of transmitting an audio signal approaching the quality of face-to-face speech requires a bandwidth of about 6 kHz (wideband speech). 
     Known digital transmission systems are capable of transmitting wideband speech audio signals. However, in order to produce an output audio signal of acceptable quality with a bandwidth of 6 kHz, these digital systems require a transmission channel with a transmission rate that exceeds the capacity of traditional telephone lines. A digital system transmits audio signals by coding an input audio signal into a digital signal made up of a sequence of binary numbers or bits, transmitting the digital signal through a transmission channel, and decoding the digital signal to produce an output audio signal. During the coding process the digital signal is reduced or compressed to minimize the necessary transmission rate of the signal. One known method for compressing wideband speech is disclosed in Recommendation G.722 (CCITT, 1988). A system using the compression method described in G.722 still requires a transmission rate of at least 48 kbit/s to produce wideband speech of an acceptable quality. 
     Because the maximum transmission rate over traditional telephone lines is 28.8 kbit/s using the most advanced modem technology, alternative transmission channels such as satellite or fiber optics would have to be used with an audio transmission system employing the data compression method disclosed in G.722. Use of these alternative transmission channels is both expensive and inconvenient due to their limited availability. While fiber optic lines are available, traditional copper telephone lines now account for an overwhelming majority of existing lines and it is unlikely that this balance will change anytime in the near future. A digital phone system capable of transmitting wideband speech over existing transmission rate limited telephone phone lines is therefore highly desirable. 
     OBJECTS OF THE INVENTION 
     The disclosed invention has various embodiments that achieve one or more of the following features or objects: 
     An object of the present invention is to provide for the transmission of high quality wideband speech over existing telephone networks, 
     A further object of the present invention is to provide for the transmission of high quality audio signals in the range of 20 Hz to at least 5,500 Hz over existing telephone networks. 
     A still further object of the present invention is to accomplish data compression on wideband speech signals to produce a transmission rate of 28.8 kbit/s or less without significant loss of audio quality. 
     A still further object of the present invention is to provide a device which allows a user to transmit and receive high quality wideband speech and audio over existing telephone networks. 
     A still further object of the present invention is to provide a portable device which is convenient to use and allows ease of connection to existing telephone networks. 
     A still further object of the present invention is to provide a device which is economical to manufacture. 
     A still further object of the present invention is to provide easy and flexible programmability. 
     SUMMARY OF THE INVENTION 
     In accordance with the present invention, the disadvantages of the prior art have been overcome by providing a digital audio transmitter system capable of transmitting high quality, wideband speech over a transmission channel with a limited bandwidth such as a traditional telephone line. 
     More particularly, the digital audio transmitter system of the present invention includes a coder for coding an input audio signal to a digital signal having a transmission rate that does not exceed the maximum allowable transmission rate for traditional telephone lines and a decoder for decoding the digital signal to provide an output audio signal with an audio bandwidth of wideband speech. A coder and a decoder may be provided in a single device to allow two-way communication between multiple devices. A device containing a coder and a decoder is commonly referred to as a CODEC (COder/DECoder). 
     These and other objects, advantages and novel features of the present invention, as well as details of an illustrative embodiment thereof, will be more fully understood from the following description and from the drawings. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a block diagram of a digital audio transmission system including a first CODEC and second CODEC in accordance with the present invention. 
     FIG. 2 is a block diagram of a CODEC of FIG.  1 . 
     FIG. 3 is a block diagram of an audio input/output circuit of a CODEC. 
     FIG. 4 is a detailed circuit diagram of the audio input portion of FIG.  3 . 
     FIG. 5 is a detailed circuit diagram of the level LED&#39;s portion of FIG.  3 . 
     FIG. 6 is a detailed circuit diagram of the headphone amp portion of FIG.  3 . 
     FIG. 7 is a block diagram of a control processor of a CODEC. 
     FIG. 8 is a detailed circuit diagram of the microprocessor portion of the control processor of FIG.  7 . 
     FIG. 9 is a detailed circuit diagram of the memory portion of the control processor of FIG.  7 . 
     FIG. 10 is a detailed circuit diagram of the dual UART portion of the control,processor of FIG.  7 . 
     FIG. 11 is a detailed circuit diagram of the keypad, LCD display and interface portions of the control processor of FIG.  7 . 
     FIG. 12 is a block diagram of an encoder of a CODEC. 
     FIG. 13 is a detailed circuit diagram of the encoder digital signal processor and memory,portions of the encoder of FIG.  12 . 
     FIG. 14 is a detailed circuit diagram of the clock generator portion of the encoder of FIG.  12 . 
     FIG. 15 is a detailed circuit diagram of the Reed-Soloman encoder and decoder portions of FIGS. 12 and 16. 
     FIG. 16 is a block diagram of a decoder of a CODEC. 
     FIG. 17 is a detailed circuit diagram of the encoder digital signal processor and memory portions of the decoder of FIG.  16 . 
     FIG. 18 is a detailed circuit diagram of the clock generator portion of the decoder of FIG.  16 . 
     FIG. 19 is a detailed circuit diagram of the analog/digital converter portion of the encoder of FIG.  12 . 
     FIG. 20 is a detailed circuit diagram of the digital/analog converter portion of the decoder of FIG.  16 . 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
     A digital audio transmission system  10 , as shown in FIG. 1, includes a first CODEC (COder/DECoder)  12  for transmitting and receiving a wideband audio signal such as wideband speech to and from a second CODEC  14  via a traditional copper telephone line  16  and telephone network  17 . When transmitting an audio signal, the first CODEC  12  performs a coding process on the input analog audio signal which includes converting the input audio signal to a digital signal and compressing the digital signal to a transmission rate of 28.8 kbit/s or less. The preferred embodiment compresses the digital using a modified version of the ISO/MPEG (International Standards Organization/Motion Picture Expert Groups) compression scheme according to the software routine disclosed in the microfiche software appendix filed herewith. The coded digital signal is sent using standard modem technology via the telephone line  16  and telephone network  17  to the second CODEC  14 , The second CODEC  14  performs a decoding process on the coded digital signal by correcting transmission errors, decompressing the digital signal and reconverting it to produce an output analog audio signal. 
     FIG. 2 shows a CODEC  12  which includes an analog mixer  20  for receiving, amplifying, and mixing an input audio signal through a number of input lines. The input lines may include a MIC line  22  for receiving an analog audio signal from a microphone and a generic LINE  24  input for receiving an analog audio signal from an audio playback device such as a tape deck. The voltage level of an input audio signal on either the MIC line  22  or the generic LINE  24  can be adjusted by a user of the CODEC  12  by adjusting the volume controls  26  and  28  When the analog mixer  20  is receiving an input signal through both the MIC line  22  and the generic LINE  24 , the two signals will be mixed or combined to produce a single analog signal. Audio level LED&#39;s  30  respond to the voltage level of a mixed audio signal to indicate when the voltage exceeds a desired threshold level. A more detailed description of the analog mixer  20  and audio level LED&#39;s  30  appears below with respect to FIGS. 3 and 4. 
     The combined analog signal from the analog mixer  20  is sent to the encoder  32  where the analog signal is first converted to a digital signal. The sampling rate used for the analog to digital conversion is preferably one-half the transmission rate of the signal which will ultimately be transmitted to the second CODEC  14  (shown in FIG.  1 ). After analog to digital conversion, the digital signal is then compressed using a modified version of the ISO/MPEG algorithm. The ISO/MPEG compression algorithm is modified to produce a transmission rate of 28.8 kbit/s. This is accomplished by the software routine that is disclosed in the software appendix. 
     The compressed digital signal from the encoder  32  is then sent to an error protection processor  34  where additional error protection data is added to the digital signal. A Reed-Solomon error protection format is used by the error protection processor  34  to provide both burst and random error protection. The error protection processor  34  is described below in greater detail with respect to FIGS. 12 and 15. 
     The compressed and error protected digital signal is then sent to an analog modem  36  where the digital signal is converted back to an analog signal for transmitting. As shown in FIG. 1, this analog signal is sent via a standard copper telephone line  16  through a telephone network  17  to the second CODEC  14 . The analog modem  36  is preferably a V.34 synchronous modem. This type of modem is commercially available. 
     The analog modem  36  is also adapted to receive an incoming analog signal from the second CODEC  14  (or another CODEC) and reconvert the analog signal to a digital signal. This digital signal is then sent to an error correction processor  38  where error correction according to a Reed-Soloman format is performed. 
     The corrected digital signal is then sent to a decoder  40  where it is decompressed using the modified version of the ISO/MPEG algorithm as disclosed in the software appendix. After decompression the digital signal is converted to an analog audio signal. A more detailed description of the decoder  40  appears below with respect to FIGS. 7,  16 ,  17  and  18 . The analog audio signal may then be perceived by a user of the CODEC  12  by routing the analog audio signal through a headphone amp  42  wherein the signal is amplified. The volume of the audio signal at the headphone output line  44  is controlled by volume control  46 . 
     The CODEC  12  includes a control processor  48  for controlling the various functions of the CODEC  12  according to software routines stored in memory  50 . A more detailed description of the structure of the control processor appears below with respect to FIGS. 7,  8 ,  9 ,  10 , and  11 . One software routine executed by the control processor allows the user of the CODEC  12  to initiate calls and enter data such as phone numbers. When a call is initiated the control processor sends a signal including the phone number to be dialed to the analog modem  36 . Data entry is accomplished via a keypad  52  and the entered data may be monitored by observation of an LCD  54 . The keypad  52  also includes keys for selecting various modes of operation of the CODEC  12 . For example, a user may select a test mode wherein the control processor  48  controls the signal path of the output of the encoder to input of decoder to bypass the telephone network allows testing of compression and decompression algorithms and their related hardware Also stored in memory  50  is the compression algorithm executed by the encoder  32  and the decompression algorithm executed by the decoder  40 . 
     Additional LED&#39;s  56  are controlled by the control processor  48  and may indicate to the user information such as “bit synchronization” (achieved by the decoder) or “power on”. An external battery pack  58  is connected to the CODEC  12  for supplying power. 
     FIG. 3 shows a lower level block diagram of the analog mixer  20 , audio level LED&#39;s  30  and analog headphone amp,  42  as shown in FIG.  2 . FIGS. 4,  5  and  6  are the detailed circuit diagrams corresponding to FIG.  3 . 
     Referring to FIGS. 3 and 4, line input  210  is an incoming line level input signal while mic input  220  is the microphone level input. These signals are amplified by a line amp  300  and a mic amp respectively and their levels are adjusted by line level control  304  and mic level control  306  respectively. The microphone and line level inputs are fed to the input mixer  308  where they are mixed and the resulting combined audio input signal  310  is developed. 
     Referring now to FIGS. 3 and 5, the audio input signal  310  is to the normal and overload signal detectors,  312  and  314  respectively, where their level is compared to a normal threshold which defines a normal volume level and a clip threshold  318  which defines an overload volume level. When the audio input signal  310  is at a normal volume level a NORM LED  320  is lighted. when the audio input signal  310  is at an overload volume level a clip LED  322  is lighted. 
     Referring now to FIGS. 3 and 6, the audio input signal  310  is fed into the record monitor level control  324 , where its level is adjusted before being mixed with the audio output signal  336  from the digital/analog converter  442  (shown in FIGS.  16  and  20 ). The audio output signal  336  is fed to the local monitor level control  326  before it is fed into the headphone mixer amplifier  334 . The resulting output signal from the headphone mixer amplifier  334  goes to a headphone output connector  338  on the exterior of the CODEC  12  where a pair of headphones may be connected. 
     The audio input signal  310  and audio output signal  336  are fed to record mix control  328  which is operable by the user. The output of this control is fed to a mix level control  330  (also operable by a user) and then to the record output amplifier  332 . The resulting output signal of the record output amplifier  332  goes to a record output  340  on the exterior of the CODEC  12 . 
     FIG. 7 shows a lower level block diagram of the control processor  48  (shown in FIG.  2 ). The encoder  406  (referenced as number  32  in FIG. 2) is further described in FIG. 12 while the decoder  416  (referenced as number  40  in FIG. 2) is refined in FIG.  16 . FIGS. 8,  9 ,  10 ,  11 ,  13 ,  14 ,  15 ,  17 ,  18 ,  19  and  20  are detailed circuit diagrams. 
     Referring to FIGS. 7 and 8 the microprocessor  400  is responsible for the communication between the user, via keypad  412  and LCD display  414 , and the CODEC  12 . The keypad  412  is used to input commands to the system while the LCD display  414 , is used to display the responses of the keypad  412  commands as well as alert messages generated by the CODEC  12 . 
     Referring now to FIGS. 7 and 9, the RAM (random access memory)  402  is used to hold a portion of the control processor control software routines. The flash ROM (read only memory)  404  holds the software routine (disclosed in the software appendix) which controls the modified ISO/MPEG compression scheme performed by encoder DSP  406  and the. modified ISO/MPEG decompression scheme performed by the decoder DSP  416 , as well as the remainder of the control processor control software routines. 
     Referring now to FIGS. 7 and 10, the dual UART (universal asynchronous receiver/transmitter)  408  is used to provide asynchronous input/output for the control processor  48 . The rear panel remote control port  409  and the rear panel RS 232  port  411  are used to allow control by an external computer. This external control can be used in conjunction with or instead of the keypad  412  and/or LCD display  414 . 
     Referring now to FIGS. 7 and 11, the programmable interval timer circuit  410  is used to interface the control processor with the keypad and LCD display. 
     Referring now to FIGS. 7,  8  and  13 , the encoder DSP (digital signal processor)  434  receives a digital pulse code modulated signal  430  from the analog/digital converter  450 . The encoder DSP  434  performs the modified ISO/MPEG compression scheme according to the software routine (described in the software appendix) stored in RAM memory  436  to produce a digital output  418 . 
     The A/D clock generation unit  439  is shown in FIGS. 12 and 14. The function of this circuitry is to provide all the necessary timing signals for the analog digital converter  450  and the encoder DSP  434 . 
     The Reed-Soloman error correction encoding circuitry  438  is shown in FIGS. 12 and 15. The function of this unit is to add parity information to be used by the Reed-Soloman decoder  446  (also shown in FIG. 16) to repair any corrupted bits received by the Reed-Soloman decoder  446 . The Reed-Soloman corrector  438  utilizes a shortened Reed-Soloman GF( 256 ) code which might contain, for example, code blocks containing 170 eight-bit data words and 8 eight-bit parity words. 
     Referring now to FIGS. 7,  16  and  17 , the decoder DSP  440  receives a digital input signal  422  from the modem  36  (shown in FIG. 2) The decoder DSP  440  performs the modified ISO/MPEG decompression scheme according to the software routine (described in the software appendix) stored in RAM memory  444  to produce a digital output to be sent to the digital/analog converter  442 . 
     The D/A clock generation unit  448  is shown in FIGS. 16 and 18. The function of this circuitry is to provide all the necessary timing signals for the digital/analog converter  442  and the decoder DSP  440 . 
     The analog/digital converter  450 , shown in FIGS. 12 and 19, is used to convert the analog input signal  310  into a PCM digital signal  430 . 
     The digital/analog converter  442 , shown in FIGS. 16 and 20 is used to convert the PCM digital signal from the decoder DSP  440  into an analog audio output signal  336 . 
     The Reed-Soloman error correction decoding circuitry  446 , shown in FIGS. 15 and 16, decodes a Reed-Soloman coded signal to correct errors produced during transmission of the signal through the modem  36  (shown in FIG. 2) and telephone network. 
     Another function contemplated by this invention is to allow real time, user operated adjustment of a number of psycho-acoustic parameters of the ISO/MPEG compression/decompression scheme used by the CODEC  12 . A manner of implementing this function is described in applicant&#39;s application entitled “System For Adjusting Psycho-Acoustic Parameters In A Digital Audio Codec” which is being filed concurrently herewith (such application and related Software Appendix are hereby incorporated by reference). Also, applicants application entitled “System For Compression And Decompression Of Audio Signals For Digital Transmission” and related Software Appendix which are being filed concurrently herewith are hereby incorporated by reference. This 
     this invention has been described above with reference to a preferred embodiment. Modifications and variations may become apparent to one skilled in the art upon reading and understanding this specification. It is intended to include all such modifications and alterations within the scope of the appended claims.

Technology Classification (CPC): 7