WO1993015559A1

WO1993015559A1 - Bit error rate controlled squelch

Info

Publication number: WO1993015559A1
Application number: PCT/US1993/000511
Authority: WO
Inventors: Tony R. Branch; Chin P. Wong
Original assignee: Motorola, Inc.
Priority date: 1992-02-03
Filing date: 1993-01-21
Publication date: 1993-08-05

Abstract

A radio communication device (100) includes an audio output device (116). The communication device (100) also includes a receiver (104) for receiving a signal and a digital signal processor (110) for calculating the bit error rate (BER) of the received signal. Also included in the communication device (100) is a comparator (208) for comparing the calculated BER with a threshold value and determining if the calculated BER is above the threshold value. The DSP (110) is coupled to audio gates (114) where the received audio is prevented from reaching the audio output device (116) when the BER is above the threshold value.

Description

BIΓERROR ATECONTROLLEDSQUELCH

Technical Field This invention relates generally to communication devices and more particularly to squelch circuits used in communication devices.

Background Modern radio receivers are typically equipped with a squelch circuit to disable the receiver's audio amplifier after the radio frequency (RF) carrier is no longer being received by the radio. The squelch circuit prevents noise from being heard at the receiver's speaker during normal radio operation. The basic purpose of a noise squelch circuit is to detect the change in the discriminator output noise as the received carrier strength changes. As the RF carrier strength increases, the discriminator noise decreases. Because the higher frequency noise falls or "quiets" more rapidly than low or mid-range noise and provides faster response time, it is used in most noise squelch circuits to determine the squelch decision (whether to squelch or unsquelch the receiver). In digital receivers, however, the received signal may have an acceptable high frequency noise floor yet be low in audio quality, because of digital bit corruption. In this application the use of typical squelch circuits result in diminished performance and would permit low quality voice signal to reach the receiver's audio amplifier. A need is therefore felt for a squelch circuit that can be effectively used with digitized voice receivers in preventing low quality voice signals from reaching the audio amplifier, and hence the speaker. Summary nf the Invention

Briefly, according to the invention, a radio communication device having an audio output device is provided. The communication device also includes a receiver for receiving a signal and a means for calculating the bit error rate (BER) of the received signal. Also included in the communication device is a comparator means for comparing the calculated BER with a threshold value and determining if the calculated BER is above the threshold value. A means for muting the received signal is also provided for muting the audio output device when the BER is above the threshold value.

Brief Description of the Drawings

FIG. 1 shows a block diagram of a communication device in accordance with the present invention.

FIG. 2 shows a block diagram of the components of a Digital Signal Processor (DSP) in accordance with the present invention.

FIG. 3 is an information packet received by the communication device of the present invention.

FIG. 4 is a flow chart of the operation of the communication device in accordance with the present invention.

Detailed DesrrintioTi of the Preferred TCmhndiment

Communication devices used in the communication of analog or digital signals utilize a variety of squelch techniques to prevent undesired noise from reaching the speaker. In general, this occurs when the quality (e.g., received signal strength) of the received signal is of such low level that would impair the receivers' ability to properly decode the received information. These squelch techniques, however, are met with a predicament, when apphed to digitized voice receivers. Since the quality of a received digitized voice signal depends both on the signal strength and the quality of the bit stream, the present squelch techniques find shortcomings in successfully distinguishing between desired and undesired received signals. The principles of the present invention provide for a squelch technique that prevents undesired signals in a digitized voice communication system from reaching the speaker.

These principles are better described by referring to the drawings, in particular to FIG. 1. This figure shows a block diagram of a communication device 100 in accordance with the present invention. The communication device 100 includes an antenna 102 where radio frequency signals are received and applied to a receiver 104. Received signals are coupled to an Analog to Digital Converter (ADC) 106 where they are converted from analog to digital and presented to a digital signal processor (DSP) 110 where they are further processed. In the preferred embodiment, the received signal comprises digitized voice. The operation of DSPs is well known in the art. In the preferred embodiment, a DSP 56001 is used to provide the demodulation and decoding of the received signal along with a host of other functions. The DSP 110 is in constant communication with a controller 112, which among other things, receives data components of the received signal from the DSP 110 and upon further decoding presents them to the user on a display 108. Voice components, once again from the DSP 110, are coupled to an audio gate circuit 114 via a Digital to Analog Converter (DAC) 118. The DAC 118 converts the error-corrected digital signal to analog and couples it to the audio gates circuit 114. The analog signals are then gated from the audio gates 114 to a speaker 116 under the control of the controller 112. The DSP 110 performs Bit Error Rate (BER) measurements in order to determine the quality of the decoded digitized voice. Once a decision has been made on the quality of the signal by evaluating its BER, a signal is communicated to the controller 112, informing it of the BER calculation. The controller 112 compares the calculated BER with a threshold and unmutes the audio gates 114 when the BER is below that threshold. Methods of calculating BER are well known in the art and are available in several varieties. The preferred BER measurement technique uses BER count with multiple error correction codes. In other words, a variety of error detection techniques may be employed to correct the errors in the received bit stream which includes different segments each being possibly coded with a different error correction routine.

In the preferred embodiment, the received signal includes an information packet having several portions, segments, or words. FIG 3 shows one such information packet 300. A preamble 302 generally precedes the received information packet followed by a sync word 304. A network ID word 306 follows the sync word and is preceded by a link control segment 308. The final component of the preferred frame is a user voice data segment 310 which is the actual intelligence desired to be received by the receiver 100. In the preferred embodiment, each of the segments of the information packet 300 are encoded with different error detection and correction information. The DSP 110 therefore includes circuitries to determine the type of error correction it should use in decoding and correcting each segment of the information packet 300, accordingly.

Referring now to FIG. 2, a block diagram of a portion of the elements of the DSP 110 in accordance with the present invention is shown. Signals from the demodulator 106 are coupled to a frame dis-assembler 202 where the information packet is disassembled into different words and a decision is made as to what segments of the DSP should be retrieved in order to decode and correct the particular frame under evaluation. Upon disassembly, the demodulated signal is coupled to a buffer 204 and a decoder with error corrector 206. The error corrector 206 proceeds to correct any errors that it can correct that may have occurred during the transmission of the signal. The result of this error corrected signal and the buffered signal of the buffer 204 are compared at a bit comparator 208. The information on the bit comparison is fed into a BER counter 210 where the BER is counted. The BER counts are then coupled to the controller 112 where a decision is made as to whether audio gates 114 should be muted or unmuted. The BER counter 210 is reset periodically to prevent BER overflow. An operating window controls the period of time the counter 210 continues to count. This operating window may be controlled via the DSP 110 or the controller 112. The combination of the buffer 204, the error corrector 206, the bit comparator 208 and the BER counter 210 are repeated for the different frame variations that may be used in the communication of the packet 300. To illustrate the number of components, a buffer 212, an error corrector 214 are shown to receive the signal at the output of the frame dis-assembler 202. The output signals of 212 and 214 are compared at a bit comparator 216 and the BER is counted as a counter 218. The result of this count is once again coupled to the controller 112. Elements 212, 214, 216, and 218 represent the Nth circuit of the error detection and correction circuits needed to process the received packet 300.

FIG. 4 shows a flow chart of the operation of the DSP 110 and the controller 112. From a start block 402, the DSP calculates the BER, block 404. The output of block 404 is coupled to a condition block 406 where the BER is compared to a threshold. The NO output indicates that the error rate is too high thereby the audio gates 114 are muted, block 408. The output of block 408 is returned to the calculate BER, block 404. The YES output of the condition block 406 which indicates that the BER is lower than the threshold and hence a valid signal is coupled to an unmute audio block 410 where the controller 112 proceeds to unmute the audio gates 114. With the audio gates unmuted, the BER is once again calculated, block 412. The output of block 412 is coupled to another condition block 414 where the BER is once again compared to a threshold and a decision is made as to whether it is higher than the threshold value. A NO output returns the operation to the calculate BER block 412. Note that in this loop the audio remains unmuted until the BER count increase to a level above the threshold. As soon as the BER creeps above the threshold, hence resulting in the condition block 414 having a YES output, the audio gates 114 are muted via a return to block 408.

In summary, it has been shown that in a digitized voice environment, the BER can be used to squelch the receiver. The DSP 110 in combination with the controller 112 decode the demodulated audio and correct the errors in the received frame. The BER is compared to a threshold to determine the quality of the received signal. As the result of the BER comparison the controller 112 proceeds to mute or unmute the audio gates 114, accordingly. The principles of the present invention, as have been demonstrated, provide for a squelch system for use in digitized voice environments. Thereby overcoming the deficiencies of present squelch systems that work with analog signals.

A benefit of the present invention is that the BER squelch may be used with any modulation technique (i.e., frequency or phase modulation, amplitude modulation, etc.) as opposed to presently available squelch systems which are highly modulation dependent. Furthermore, bit corruption is what determines the voice quality and not the prevailing RF noise. Hence, the DSP 110 is only concerned with the quality of the bit stream being received and its ability to successfully retrieve and correct any corrupted bits, rather than the prevailing RF noise.

Another benefit of the present invention is that it uses the existing components of a communication device to achieve its objectives in preventing undesired signals from reaching the speaker. In a digitized voice communication system a DSP and a controller are essentially required to allow for recovery of digitized voice. These components are used in this invention to provide an additional benefit of determining whether that digitized voice is of high quality or of unrealizable quality, hence preventing unintelligible signals from getting to the speaker. It is well understood that the DSP 110 could be so designed to include the functionality of the controller 112 with a presentation of the components of the communication device 100 is not to imply limitations to the invention and is there merely to provide a preferred embodiment of the present invention.

What is claimed is:

Claims

1. In a radio communication device having an audio output device, a metiiod of preventing low quality voice signals from reaching the audio output device comprising the steps of: receiving a signal to produce a received signal; calculating the Bit Error Rate (BER) of the received signal; comparing the received signal BER with a threshold value; and coupling the received signal to the audio output device when the received signal BER is below the threshold value.

2. The method of claim 1, further comprising the step of decoupling the received signal from the audio output device when the received signal BER is above the threshold value.

3. A communication device, comprising: receiver means for receiving a signal to produce a received signal; means for calculating the BER of the received signal to produce a calculated BER; comparator means for comparing the calculated BER with a threshold value; and means for muting the received signal when the calculated BER is above the threshold value.

4. The communication device of claim 3, further including means for unmuting the received signal when the calculated BER is below the threshold value.

5. The communication device of claim 3, further including means for correcting errors in the received signal.

6. The communication device of claim 3, wherein the means for muting includes means for unmuting the received signal when the calculated BER is below the threshold value.

7. The communication device of claim 3, wherein the means for calculating the BER comprises a Digital Signal Processor (DSP) means.

8. A communication device having an audio output device, comprising: receiver means for receiving a voice signal to produce a recovered signal; means for recovering a digitized voice signal from the received signal; means for converting the digitized voice signal to an analog signal; and squelch means for sensing when to prevent the analog signal from reaching the audio output device, the squelch means comprising: means for calculating the BER of the digitized voice signal; means for comparing the BER of the digitized voice signal with a threshold level; and means for decoupling the analog signal from the audio output device when the BER is above the threshold level.

9. The communication device of claim 8, further includin means for correcting errors in the digitized voice signal.

10. The communication device of claim 8, wherein the means ^~~ for recovering the digitized voice signal includes means for recovering a signal having a plurality of uniquely encoded segments.