DTMF tone passer in a voice communication system

The present invention relates to a system and method for sending and receiving digital signals in a voice communication system. The system includes a data buffer, a tone detector, a speech encoder, and a transmitting unit. The data buffer receives and stores a digitized signal including voice band tone data and speech data. The tone detector detects the voice band tone data and generates at least one tone packet containing data indicative of the voice band tone data. The speech encoder encodes the speech data into a plurality of compressed speech packets. The transmitting unit in communication with the tone detector and the speech encoder transmits a packet message including tone packets and speech packets over an RF channel. The present invention also relates to a receiver including a speech decoder and a tone generator. The speech decoder decompresses speech packets to generate a digitized speech signal. The tone generator detects tone packets and generates a digitized DTMF tone signal from data contained in the tone packet. The present invention also relates to a method of encoding a digital signal. The method includes the steps of receiving the digital signal having a speech component and a digit component, detecting digit information from the digit component, generating a tone packet signal and a speech packet signal, and generating a digital output signal including the tone packet signal and the speech packet signal. The tone packet signal includes the digit information, and the speech packet signal is generated by compressing the speech component.

BACKGROUND OF THE INVENTION 
Conventional wireless communication system use encoders to reduce the 
amount of bandwidth required for the transmission of speech signals over 
the air. Typically, the encoder compresses the digitized speech signal 
into a packet signal having reduced bandwidth. Generally, the encoder 
analyzes speech signals and designates particular codes to represent 
particular characteristics in the digital signal. For example, fast 
changing sounds requiring more information and therefore receive a higher 
percentage of bits than slow changing sounds. 
After compression by the encoder, compressed speech packets are 
subsequently transmitted through the air over an RF chapel to a base 
station unit, where the speech packets may be transmitted to another 
remote base station unit or decompressed and transmitted to a local 
telephone subscriber. 
Although encoders may be used to reduce transmission bandwidth requirements 
by compressing digitized speech, the encoders may distort voice band tone 
data, such as dialed digit information, that may be present in the speech 
signal. Typically, dialed digit information is represented by dual tone 
multifrequency (DTMF) tones that are generated when a subscriber presses a 
key on a telephone keypad. Since conventional encoders are designed to 
compress speech data instead of tone data, compressing a speech signal 
containing DTMF tones may result in losing at least some of the dialed 
digit information. 
One approach to address this problem has been used by conventional mobile 
telephone systems using an IS-54 mobile telephone. The IS-54 mobile 
telephone has a keypass that senses which key is pressed and then sends 
digit information for the pressed key to the base station unit over a 
communication channel that is separate from the voice chapel. Typically, 
the keypass information is transmitted over the control channel and the 
voice information is transmitted over a traffic channel. The IS-54 
telephone also receives digit information over the separate communication 
channel. 
However, there are many applications where a subscriber would rather use a 
conventional analog telephone instead of a mobile telephone for 
communication in a wireless system. One such application involves a 
wireless telephone system allowing a plurality of subscribers in a remote 
location to receive telephone service. In this application, a radio 
transmitting unit including an encoder is used to send speech and DTMF 
digit data over an RF channel to a base station unit that is coupled to a 
local office within the public telephone network. In such an application, 
many potential customers would rather use inexpensive traditional analog 
telephones than the more expensive IS-54 type mobile telephones. In 
addition, where the radio transmitting unit is physically separated from 
the subscriber touch tone keypad, a keypress detector similar to those 
used in IS-54 telephones will not operate correctly. 
However, in such an application, a subscriber using a traditional analog 
telephone will probably not be able to send voice band tones, such as DTMF 
tones, after establishing a telephone call. The need to send DTMF tones 
generally arises when the subscriber is communicating with an automated 
touch tone based system, such as a bank account information system. Since 
DTMF tones sent by the subscriber's phone are encoded before being 
transmitted over the air to the base station unit, a DTMF detector at the 
automated touch tone system will have difficulty detecting the previously 
encoded DTMF digits. 
Thus, it would be desirable to allow a subscriber to use an analog 
telephone to send voice band tone data instead of using an IS-54 type 
phone that sends a keypress signal in a wireless voice communication 
system. Such wireless voice communication systems include but are not 
limited to fixed wireless systems and airplane air to ground telephone 
system. Accordingly, there is a need for a wireless voice communication 
system supporting the use of analog telephones. 
SUMMARY OF THE INVENTION 
The present invention relates to an apparatus for sending and receiving 
digital signals in a voice communication system. The system includes a 
data buffer, a tone detector, a speech encoder, and a transmitting unit. 
The data buffer receives and stores a digitized signal including voice 
band tone data and speech data. The tone detector detects the voice band 
tone data and generates at least one tone packet containing data 
indicative of the voice band tone data. The speech encoder compresses the 
speech data into a plurality of compressed speech packets. The 
transmitting unit in communication with the tone detector and the speech 
encoder transmits a packet message including tone packets and compressed 
speech packets over an RF channel. 
Preferably, the tone packet includes a header field, a duration field, and 
a key field. The header field may contain a distinguishable binary 
sequence. In the preferred embodiment, the voice band tone data is 
directed to DTMF digits. The duration field may include a plurality of 
duration combination subfields, and the key field may include a plurality 
of key subfields where at least one of the key subfields contains data 
indicative of the voice band tone data, such as a DTMF digit. 
The present invention also relates to a receiver adapted to receive the 
packet message from the transmitting unit. The receiver includes a speech 
decoder and a tone generator. The speech decoder decompresses speech 
packets to reconstruct a digitized speech signal. The tone generator 
detects tone packets and generates a digitized DTMF tone signal from data 
contained in the tone packet. 
The present invention also relates to a method of encoding a digital signal 
having a speech component and a digit component. The method includes the 
steps of receiving the digital signal, detecting digit information from 
the digit component, and generating a digital output signal including a 
tone packet signal and a speech packet signal. The tone packet signal 
includes digit information from the digit component, and the speech packet 
signal is a compressed signal encoded from the speech component. 
Preferably, the digital output signal is transmitted over a common 
communication channel such as an RF channel. 
The present invention may also include the steps of receiving the digital 
output signal, recovering digit information from the tone packet signal, 
and decoding the speech packet signal to construct a digitized speech 
signal. Preferably, the method also generates a digitized tone signal, 
such as a DTMF signal, representative of the digit information. Finally, 
the method preferably generates a decompressed output signal from the 
digitized speech signal and the digitized tone signal. 
The invention, together with further objects and attendant advantages, will 
best be understood by reference to the following detailed description, 
taken in conjunction with the accompanying drawings.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
As shown in FIG. 1, a voice communication system 10 according to a 
preferred embodiment of the present invention includes a first subscriber 
analog telephone 12, a transmitter 13, a receiver 19 including a tone 
generator 20, and a second subscriber analog telephone 23. The transmitter 
13 includes a tone detector 14, a first antenna 16, and a second antenna 
18. In the preferred embodiment, the first subscriber analog telephone 12 
is connected to the transmitter 13, and the second subscriber analog 
telephone 23 is connected to the receiver 19. The transmitter 13 receives 
an analog signal 11 carrying speech and DTMF signals from the first 
subscriber telephone 12. The transmitting unit 13 converts the analog 
speech signal into a digital speech signal. The tone detector 14 
determines whether the digital speech signal contains any voice band tone 
data, preferably dialed digits such as dual tone multifrequency (DTMF) 
tones, and encodes any detected digits into special tone packets. The 
voice band tone data may also be facsimile tones or modem tones. The 
transmitting unit 13 compresses the digital speech signal, replaces 
compressed speech packets with any special tone packets from the tone 
detector 14 if necessary, and transmits a digitally modulated and 
compressed output signal using the antenna 16 via an RF transmission 
channel. The digitally compressed signal is received by the receiver 19 
using the second antenna 18. The tone generator 20 recovers any special 
tone packets in the received signal and generates a corresponding tone 
signal to be sent to the second telephone subscriber 23. The receiver 19 
decompresses the digital speech signal and converts the signal to an 
analog speech signal 22 that is sent to the second subscriber telephone 
23. 
As shown in FIG. 2, a preferred embodiment of the tone detector 14 includes 
an A/D converter 24, a speech encoder 25, a DTMF analyzer 26, a switch 34, 
and an RF modulator 36. In the preferred embodiment, the A/D converter 24 
is connected to the speech encoder 25 and to the DTMF analyzer 26. The 
switch 34 is connected to the speech encoder 25, the DTMF analyzer 26, and 
the RF modulator 36 which is connected to the antenna 16. 
The A/D converter 24 receives the analog speech signal 11 from the 
telephone subscriber phone 12 and generates a digital pulse code modulated 
(PCM) signal 28. The PCM signal 28 is received by the speech encoder 25 
and the DTMF analyzer 26. The speech encoder compresses the PCM signal 28 
and generates a compressed speech packet signal 30. The DTMF analyzer 26 
detects DTMF digits present in the PCM signal 28 and generates a tone 
packet signal 32. The tone packet signal preferably includes a plurality 
of special tone packets with each tone packet containing information 
indicative of the DTMF digit detected. 
The DTMF analyzer 26 controls the switch 34 with a control signal 33. When 
the DTMF analyzer 26 detects a DTMF digit, the control signal 33 instructs 
the switch 34 to replace the compressed packet received from the speech 
encoder 25 with the special tone packet received from the DTMF analyzer 
26. Thus, the output signal from the switch 34 is a packet message that 
contains both compressed speech packets interleaved with any special tone 
packets. The RF modulator 36 modulates the output signal from the switch 
34 to an RF frequency suitable for transmission. The resulting modulated 
output signal is transmitted over the air using the antenna 16. 
As shown in FIG. 3, a preferred embodiment of the receiver 19 includes an 
RF demodulator 40, a packet analyzer 44, a silence packet generator 45, a 
speech decoder 46, a first switch 49, a DTMF tone generator 50, a second 
switch 52, and a D/A converter 56. In the preferred embodiment, the RF 
demodulator 40 is connected to the packet analyzer 44, and the packet 
analyzer 44 is connected to the first switch 49 and to the second switch 
52. The first switch 49 is also connected to the silence packet generator 
45 and the speech decoder 46. The second switch 52 is preferably connected 
to the speech decoder 46, the DTMF tone generator 50, and the D/A 
converter 56. 
The RF demodulator 40 receives an input signal, such as the modulated 
output signal from the transmitter 13, from the antenna 18 and generates a 
demodulated compressed packet signal 42. The packet analyzer 44 receives 
the packet signal 42 and analyzes the packet signal 42 for special tone 
packets. The packet analyzer 44 sends the compressed digital signal 42 to 
the speech decoder 46. When the packet analyzer 44 detects speech packets, 
the decoder 46 decompresses the compressed digital signal 42 and generates 
a digital PCM speech signal. When the packet analyzer 44 detects a DTMF 
tone packet, the speech decoder 46 preferably receives a silence packet, 
such as an all logical zero entry, from the silence packet generator 45. 
The speech decoder 46 generates silence instead of the digital PCM speech 
signal when receiving the silence packet. 
When the packet analyzer 44 detects a special DTMF tone packet, the 
analyzer 44 recovers DTMF tone information including a key value and a 
tone duration value. The packet analyzer 44 sends the tone information to 
the tone generator 50, and the generator 50 transmits a PCM digital 
representation of the DTMF tone corresponding to the tone information. In 
the preferred embodiment, the packet analyzer 44 also sends a signal over 
a switch control line 48 to the first switch 49 to send a silence packet 
from the silence packet generator 45 to the speech decoder 46 and to the 
second switch 52 to send the tone generated by the DTMF tone generator 50 
to the D/A converter 56. As a result, the output signal 54 includes both 
PCM speech from the speech decoder 46 and tone data from the DTMF tone 
generator 50. The digital output signal 54 is converted by the D/A 
converter 56 into an analog speech signal 22 sent to the second subscriber 
telephone 23. 
FIG. 4 is a block diagram of a transcoder module 108 embodying the present 
invention. The transcoder 108 performs the functions of the transmitter 13 
and the receiver 19 described above. The transcoder module 108 includes a 
telephone subscriber interface 110, a plurality of digital signal 
processors (DSPs) 116-130, a dual port RAM 134, a common switching module 
(CSM) 136, and an RF modulator 138. The plurality of DSPs may be 
programmed to perform a variety of functions. In a preferred embodiment, 
DSP1 116 functions as an input buffer 116, DSP2 118 functions as a DTMF 
tone detector and packet analyzer, and DSPs 3-8 120-130 perform digital 
compression and decompression. The telephone subscriber interface 110 
communicates with the input buffer 116 over a PCM bus 112 and a PCM 
highway controller 114. The DSPs communicate with each other over a time 
division multiplexed (TDM) bus 115 and are synchronized by a clock 117. 
DSP2 118 communicates with the dual port RAM 134 which in turn 
communicates with the CSM 136. The CSM 136 communicates with the RF 
modulator 138, and an RF demodulator 142. The RF modulator 138 
communicates with an RF upconverter 140, and the RF demodulator 142 
communicates with an RF downconverter 144. 
Each of the DSPs 116-130 are preferably ATY1610 type DSPs available from 
AT&T Microelectronics. The DSPs 116-130 may be programmed in the ATT1610 
assembler programming language. The telephone subscriber interface 110 is 
an ALCN type interface available from Alcatel. The PCM bus 112, the PCM 
highway controller 114, and the TDM bus 115 is provided by the ATY1610 
from AT&T Microelectronics. The clock 117 is preferably a 65.536 Mhz clock 
from Connor Winfield Corp, 2111 Comprehensive Dr., Aurora, Ill. The dual 
port RAM device 134 is preferably a 2K.times.8 IDT71421 RAM available from 
Integrated Device Technology, 3236 Scott Boulevard, Santa Clara, Calif. 
95054. The RF section includes an RF modulator, an RF upconverter, an RF 
downconverter, and an RF demodulator as described in North American 
Digital Cellular Standard IS-54. 
In a transmitting mode, the transcoder module 108 modulates and transmits 
digital data over the air using RF channels to a base station unit 146. 
The base station communicates with the transcoder module via an RF 
upconverter 150 and an RF downconverter 148. The digital data transmitted 
by the transcoder module 108 may include speech data, tone data, PCM data, 
compressed data, or any other type of data represented in digital form. 
The telephone subscriber interface 110 preferably represents a plurality of 
subscriber telephones but may interface with a single analog telephone. As 
is well known in the art, when an individual speaks into the transmitting 
end of a telephone, an analog speech signal representing the individual's 
voice is created. PCM circuitry within the telephone subscriber interface 
110 receives the analog speech signal and samples the analog speech signal 
at 8 Khz with 8 bits per sample creating a digital PCM speech signal 
having a 64 Kb/sec data transmission rate. The resulting digital PCM 
speech signal is preferably transmitted over the PCM bus 112. In a 
preferred embodiment where the telephone subscriber interface supports 
multiple subscribers, the PCM highway controller 114 multiplexes digital 
speech signals from multiple subscribers into a single digital signal that 
is stored in an input buffer 116. The input buffer area 116 preferably is 
implemented with a DSP such as DSP1116 which stores samples of the digital 
speech data and acts as a selective buffering device. The DSP1 116 also 
converts between the A-law PCM data used by the PCM highway controller 114 
and a linear PCM signal used by the TDM bus 15. 
The digital speech data is transmitted from DSP1 116 to DSP2 118 and then 
to a corresponding compression digital signal processor designated for the 
particular subscriber channel, e.g. DSPs 3-8 120-130. The DSP associated 
with the channel compresses the digital speech data into compressed speech 
packets and sends the packets to DSP2 118. Subsequently, DSP2 118 performs 
voice activity detection as well DTMF detection on the digital data 
received from DSP1 116. 
If DSP2 118 detects the presence of voice data, the corresponding 
compressed speech packets received from the associated compression digital 
signal processor are transmitted to a dual port RAM device 134. The dual 
port RAM 134 acts as a data exchange buffer between the transcoder module 
108 and the CSM 136. The CSM 136 is a high level controller receiving 
digital data from the dual port RAM 134 and transmitting data to the radio 
modulator 138 for transmission over the air via the RF upconverter 140. 
The transmitted data may be received by the RF downconverter 148 of the 
base station unit 146. The CSM 136 may allocate appropriate frequencies 
for communicating with the base station unit 146. 
If DSP2 118 detects the presence of a DTMF signal, DSP2 preferably replaces 
the compressed speech packets with specially encoded DTMF packets to be 
sent over the air, such as to the base station unit 146. DTMF signals are 
present whenever a number or symbol is pressed on a subscriber telephone 
key pad on the subscriber telephone 12. Each number or symbol on the key 
pad is represented by a particular pair of frequencies, hence the name 
"dual tone." DTMF frequencies are specified by CCITT Q.24. Special coding 
by the transmitter, such as by sending a special tone packet, is necessary 
to allow the receiver to properly distinguish DTMF signals from other 
voice data. 
FIG. 5 illustrates a preferred method of performing DTMF detection that may 
be executed on DSP2 118. As shown in FIG. 5, the tone detection process 
reads a 40 ms segment including 320 samples of PCM speech 200 and performs 
DTMF tone detection on the 40 ms segment 202. At 204, the 40 ms segment 
enters an encoding process. If a DTMF tone is present, as determined by a 
comparison 206, then a special tone packet is created and transmitted at 
208. Otherwise, the 40 ms segment is encoded into a compressed speech 
packet to be transmitted 210. 
Referring to FIG. 6, each 40 ms segment of speech is preferably further 
divided into four 10 ms sub-frames. Each sub-frame is preferably analyzed 
using digital Fourier transform (DFT) techniques for the presence of a 
valid DTMF tone. As shown in FIG. 6, there are eight possible DTMF tone 
duration combinations within the 40 ms segment 212-226. These eight DTMF 
tone duration combinations 212-226 are encoded using a 3 bit field in a 
DTMF tone packet. Also, as is known in the art, there are 16 valid DTMF 
digit combinations described herein as a key value. The DTMF key value may 
be encoded in a 4 bit field in the special DTMF tone packet. 
In order to improve robustness in the DTMF digit detection process, several 
techniques have been developed. First, adherence to the CCITT 
Recommendation Q.24 requires that the received signal should be above 25 
dbm, that the twist, the difference between the amplitudes of the pair of 
tones, should be within -8 to +4 dbm, and that the frequency deviation 
should be within 1.5%. 
In addition, the preferred embodiment of the present invention uses a 
comparison of in-band energy to total energy. Since most of the signal's 
energy should be concentrated around the DTMF frequencies for a received 
DTMF tone, a ratio of in-band to total energy provides an excellent 
indicator of a DTMF tone. 
In FIG. 7, an "ND" indicates no DTMF tone detected, and a "D" indicates a 
DTMF tone is detected in a received digital signal 228. For a detected 
DTMF digit, the energy ratio should exceed an energy threshold value for a 
contiguous string of 10 ms sub-frames 230. Each of the contiguous 10 ms 
DTMF sub-frames 230 should have the same DTMF key value. Also, the energy 
detected should not vary more than about 2 dB. 
The DTMF detection process also determines whether the DTMF digit signals 
are preceded by a 50 ms silence interval. For example, if a 10 ms subframe 
indicates no digit detection followed by a 10 ms sub-frame in which a DTMF 
digit is detected, then the 40 ms segment before the first 10 ms subframe 
should be below a silence threshold, preferably 48 dbm. As shown in FIG. 
7, a DTMF digit interval 232 is preceded by a silence interval 234 and 
each of these intervals include contiguous 10 ms subframes 230. 
Although the preferred embodiment requires that each of the above 
conditions be satisfied before a DTMF digit is detected, the present 
invention is not limited to using all of these techniques. In applications 
allowing a lower degree of DTMF digit reception robustness, any of the 
above techniques or combination of techniques may be used but are not 
required. 
In a receiving mode, the base station unit 146 may modulate compressed 
speech packets over the air via an RF up converter 150 to the RF down 
converter 144 of the multisubscriber transcoder unit 108. The CSM 136 
receives compressed speech packets and transmits the packets to the dual 
port RAM 134 acting as a data exchange buffer between the base station 
unit 146 and the CSM 136. 
DSP2 118 searches for the presence of compressed speech packets in the dual 
port RAM 134. When DSP2 118 receives compressed packets, they are sent to 
one of the decompression digital signal processor, such as one of DSPs 3-8 
120-130. Preferably, each of the decompression DSPs 120-130 are assigned 
to decompress speech signals routed to particular telephone subscribers so 
that load balancing occurs among the decompression DSPs 120-130. The 
assigned decompression DSP decompresses the packets into PCM speech data 
and sends the PCM speech data to DSP1 116. DSP1 116 transmits the PCM data 
to the telephone subscriber interface 110 where a D/A converter using an 8 
Khz clock reconstructs an analog speech signal. 
The DSP2 118 performs at least some of the functions of the packet analyzer 
44 described above. Specifically, when the DSP2 118 detects a special DTMF 
tone packet from the received signal, the DSP2 118 commands DSP1 116 to 
generate a specific tone. DSP1 116 then replaces the PCM data from the 
decompression DSP 120-130 with PCM tone data corresponding to the received 
tone information. An output signal including PCM speech and tone data is 
then transmitted to the telephone subscriber interface 110 where the D/A 
converter reconstructs the analog signal. 
A preferred DTMF tone generation process that may be executed in the DSP2 
118 and DSP1 116 is shown in FIG. 8. First, the tone generation process 
reads a single packet 300 and determines whether the packet is a special 
tone packet 302, 304. If the packet is a tone packet, the decoder state is 
cleared by feeding a compressed value of silence to the decoder 306, and 
320 samples of PCM data for the DTMF digit corresponding to the tone 
packet is generated at 308. Otherwise, the decoder decompresses the speech 
packet and creates 320 samples of PCM speech signal to be sent to a 
subscriber. The decoder preferably corresponds to one of the decompression 
DSPs 120-130 described above. 
FIG. 9 shows a preferred embodiment of a tone packet 312. The tone packet 
312 includes a plurality of fields including a header field 314, a 
duration field 316, and a key field 318. Each duration field 316 
preferably has seven 3 bit subfields 320. Each of the key fields 318 
preferably has seven 4 bit subfields 322. Although the tone packet 312 as 
shown is 200 bits long, the tone packet 312 may be any other length as 
long as the key 318 and duration 316 information may be reliably detected 
by the receiver. Preferably, the tone packet 312 is the same length as a 
compressed speech packet, allowing easy replacement of a compressed speech 
packet with a tone packet 312. 
The fields in the tone packet 312 are preferably formatted for improved 
DTMF digit recovery. During transmission, the header field 314 is 
preferably coded with 64 bits each having a logic "1" value. Preferably, 
the tone generator examines the header field 314 for the occurrence of a 
predetermined number, preferably at least 60, logic "1" bits when 
detecting the tone packet 312. The following bits of the tone packet 312 
are then examined. If at least a majority, and preferably 5, of the 7 
3-bit duration combinations match, the duration combination value from the 
matching duration subfields 320 is selected. If at least a majority, 
preferably 5, of the 7 key field 4 bit combinations match, then the key 
represented by the 4 bit field 322 is selected. A DTMF tone generator may 
generate a DTMF tone based on the detected key value and tone duration 
value. 
Additional advantages and modifications will readily occur to those skilled 
in the art. The invention, in its broader aspects, is therefore not 
limited to the specific details, representative apparatus, and 
illustrative examples shown and described. Various modifications and 
variation can be made to the present invention without varying from the 
scope or spirit of the invention, and it is intended that the present 
invention cover the modifications and variations provided they come within 
the scope of the appended claims and their equivalents.