Abstract:
An Automatic Gain Control System for voice codecs used in a digital wireless handset or in any other environment where there is a large amount of background noise. The invention utilizes the processing power available in Digital Signal Processor  111  to adaptively adjust the gain of amplifier  202  and filter  205  in the transmit channel to eliminate audio distortion in a “loud talker” environment. The gain correction information is transmitted by the DSP  111  to the digital filter and PCM I/O block  204  in the 3 unused bits of the 16 bit PCM data word thus eliminating any additional connections.

Description:
This application claims priority under 35 U.S.C. §119(e)(1) of provisional application No. 60/109,793, filed Nov. 25, 1998. 
    
    
     TECHNICAL FIELD OF INVENTION 
     The technical field of this invention is communication systems, particularly communication systems as applied in digital wireless telephone handsets. 
     BACKGROUND OF INVENTION 
     In a digital telephone handset, the speaker&#39;s voice is converted to an electric current in a microphone. This current is then amplified in an analog preamplifier, and converted to a digital representation using an analog to digital converter. The output of the analog to digital converter is a pulse code modulated signal that in a wireless telephone application will typically have a dynamic range of 13 bits. This digital signal is then passed through a digital filter to remove aliasing that may be generated in the analog to digital converter, and is then passed on to a Digital Signal Processor for further processing. In order to compensate for the manufacturing tolerances of the various components in this chain, the gain of the analog preamplifier and the digital filter is programmable. During the manufacturing process, the gains of the various amplifiers are adjusted to balance the overall gain of the system to the design requirement, usually by adjusting a PGA (Programmable Gate Array). Once this adjustment is made, the gains become fixed, and can no longer be changed. 
     When the phone is used in a location with a high background noise, a situation is encountered known as the “loud talker environment”. The combination of the high background noise and the louder than normal speech of the user will generate an abnormally large signal from the microphone and preamplifier. This may then exceed the dynamic range capability of the analog to digital converter and the digital filter, resulting in clipping and distortion. In an extreme case, this clipping may result in the PCM signal becoming fully saturated, in which case all of the intelligence content of the signal is lost. In a less serious case, the clipping will result in a distortion of the voice signal. The Digital Signal Processor may attempt to correct the distortion, but it is usually not possible to do that in a satisfactory manner since the damage is done by the time the DSP receives the signal. In the digital phones available today, this is a major problem that seriously limits the usefulness of the phone in a noisy environment. 
     SUMMARY OF THE INVENTION 
     The present invention provides an active Automatic Gain Control (AGC) system that uses the processing power available in the Digital Signal Processor to detect the distortion caused by the clipping in the ADC and the digital filter, and then adjusts the variable gain components to maintain the signal level below the clipping threshold. While the current invention shows this applied in a digital wireless telephone, the same problem exists in voice codecs in general, and the solution applies wherever voice is digitized in a high background noise environment. 
     While Automatic Gain Control circuits are know in the art and are commonly used in analog communication systems, they are difficult to implement in a digital system. The present invention makes the implementation practical by using the processing power available in the Digital Signal Processor to recognize the distortion, and to apply the required correction. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     These and other aspects of this invention are illustrated in the drawings, in which: 
     FIG. 1 shows the block diagram of one possible embodiment of a digital wireless handset 
     FIG. 2 is a block diagram of one embodiment of the invention, showing the audio components of a digital wireless handset; 
     FIG. 3 is a flow chart of the actions the DSP takes during normal operation; 
     FIG. 4 is a flow chart of the actions when the “loud talker” condition is detected; 
     FIG. 5 is a flow chart of the actions taken at the end of the “loud talker” condition; 
     FIG. 6 shows the PCM data format from the DSP to PCM I/O block  204  in FIG. 2, showing the control information format. 
    
    
     DESCRIPTION OF THE PREFERRED EMBODIMENT 
     The present invention includes an active Automatic Gain Control circuit that allows digital wireless telephone systems to operate in a high background noise environment by detecting the audio distortion caused by the noise, and takes corrective action by adjusting the system gain to reduce or eliminate the distortion. 
     FIG. 1 shows a typical embodiment of a digital wireless handset  100  in a block diagram format. The analog baseband function (ABB)  101  contains the transmit and receive channels shown in FIG. 2, and the radio frequency (RF) interface circuits. This function may be implemented using an integrated circuit such as the TCM-4400. The input to ABB  101  is from microphone  201 , and audio is output to earphone  212 . The RF interface connects to the RF section  104 , typically comprising of a receiver  105  such as a TRF-1500, frequency synthesizer  106  such as a TRF20xx and a modulator  107  such as a TRF-3xxx. The designation “xx” shows the availability of multiple parts that are implementation dependent. The power amplifier  108  is typically constructed from discreet components well known in the art. The output of the power amplifier  108  and the input of receiver  105  connect to the antenna  109 . The integrated circuits TCM-4400, TRF-1500, TRF20xx and TRF-3xxx are available from Texas Instruments, Inc. 
     The digital baseband unit  110  may be a single integrated circuit, such as a GEMI12LA68, and will typically contain the DSP  111  used in audio and other signal processing functions, the micro controller  112  used for control and user interface functions, and random logic  113  for miscellaneous functions. User display  114 , keyboard  115  and sim card interface  116  are implementation dependent and are normally constructed from discreet components. The functions of the keyboard and display are self explanatory. The sim card contains subscriber and security information specific to the service being used. Functional block  105  contains power supply, control and management circuits, and may be implemented with an integrated circuit such as the TPS-9120. Circuits GEMI12LA68 and TPS-9120 are also available from Texas Instruments, Inc. 
     FIG. 2 illustrates in a block diagram form the preferred embodiment of the invention. This figure shows the audio portion of a digital wireless handset. The handset is responsive to input from microphone  201 , and generates the audio output in earphone  212 . The signal output of microphone  201  is amplified in variable gain amplifier  202 . The gain of amplifier  202  can be reduced by 6 db, using control line  209 . The output of amplifier  202  is then converted to digital representation by analog to digital converter (ADC)  203 . The output of ADC  203  is then filtered and further amplified by transmit filter  205 , implemented within the digital filter and PCM I/O block  204 . Filter  205  is a variable gain device, and it&#39;s gain may be reduced by 6 db in 2 db steps, under the control of the digital signal processor (DSP)  111 . If more than 6 db gain reduction is required, the filter block  205  activates control line  209  thus reducing the gain of amplifier  202  by 6 db. Functional blocks  201 ,  202 ,  203  and  205  comprise the transmit channel  213  of the digital handset. 
     The audio to be output is supplied to receive filter  207  by the DSP  206 . This audio is in the pulse code modulated (PCM) format shown in FIG.  6 . The transfer is controlled by PCM CLOCK  601 , and is formatted in 16 bit words. In each 16 bit word, the audio information is contained in 13 bit segment  602 . The remaining 3 bit segment  603  contains the gain control information from the DSP to the transmit channel. This information is extracted by receive filter  207 , and is communicated to transmit filter  205  through control line  208 . 
     The output of receive filter  207  is converted to analog format by digital to analog converter (DAC)  210 , amplified by earphone amplifier  211  and then converted to sound by earphone  212 . Functional blocks  207 ,  210 ,  211  and  212  comprise the receive channel  214  of the digital handset. FIG. 7 illustrates a digital wireless handset  100  that implements audio interface  215 , according to a preferred embodiment of the invention. 
     FIG. 3 shows the normal program flow in DSP  111 . Start block  301  initializes the system, and then block  302  reads one data sample in a 16 bit PCM format data word from transmit filter  205 . Test block  303  determines if the  13  significant data bits are all 1&#39;s, indicating a clipping condition. If the result of the comparison is no, test block  304  checks if the loud flag is set. If not, block  305  completes processing of the audio data, and then returns control to block  302  to process the next data sample. 
     If the comparison in test block  303  detects a data word of all 1&#39;s indicating a clipping condition, program flow transfers to block  401  in FIG.  4 . First, the loud flag is set in block  401 , then delay counter A is incremented in block  402 . The purpose of the delay counter is to ignore clipping caused by short transient noise spikes. The value of the delay is implementation dependent, but would typically be in the millisecond range. Test block  403  then determines if the preset delay value has been reached. If not, block  407  replaces the clipped audio sample with the last good sample value, and returns control to block  305 . 
     If test block  403  detects that the delay value has expired, the current gain setting is checked by test block  404 . This is done by reading the gain control register in memory. This register is initialized by start block  301  to a value of  12 , representing the initial gain of the transmit channel. Every time the DSP  111  makes a gain adjustment, this register is updated with the new value. Since the maximum gain reduction is 12 db, a value of 0 in this register indicates that the minimum gain has been reached. Conversely, a value of  12  indicates that initial gain setting has been restored. If the gain is already at the minimum possible value, block  405  substitutes a 0 data word thus effectively muting the audio, and returns control to block  305 . If the gain is not at minimum, block  406  reduces the gain by 2 db, and transfers control to  407  which then replaces the clipped value with the last good sample and than returns control to  305 . 
     If test block  304  determines that the loud flag is set, control transfers to block  501  in FIG.  5 . At this point, we check in  501  if delay time B has expired. The purpose of the delay is to ignore short transients that may be meaningless. The length of this delay is also implementation dependent, but would typically be in the millisecond range. If the delay has not expired, the delay counter B is incremented in block  502 , and control is returned to  305  to complete processing the data. If the delay B has expired, test block  503  checks if the gain is already at the maximum value. If yes, block  504  clears the loud flag, resets delays A and B, and returns control to block  305 . If the gain is not at maximum, block  505  increases gain by 2 db, and then returns control to  305 .