Patent Abstract:
A speech encoding system for use with a digital cellular communication device and a receiving station, includes a mechanism for determining whether a voice communications packet needs to be treated as a data communications packet; a voice recognition mechanism for receiving instructions by voice command; and a control mechanism for responding to said voice command and controlling a controlled entity. A method for encoding a voice command generated on a digital cellular communication device and transmitted over a wireless communication network to a receiving station for controlling a controllable entity includes recognizing a voice command; determining whether the voice command needs to be treated as a data communications packet; encoding the voice command; connecting the voice command to a voice recognition mechanism; and controlling a controlled entity with the voice command.

Full Description:
FIELD OF THE INVENTION 
     This invention relates to mobile communications, and specifically to the elimination of speech drop-outs for certain voice transmissions. 
     BACKGROUND OF THE INVENTION 
     Effective voice recognition technology can reduce the need for keypads and large displays. This is important when considering portable devices which are intended to connect to the world-wide communications network known as the internet. The problem is that current voice recognition technology, which is suitable for use on portable, battery-powered devices, fails to achieve needed speed or accuracy. The solution, because such products are intended to connect wirelessly to a network, is to install voice recognition hardware and software on network-based servers which a user can dial into. 
     Server-based recognition systems are in widespread use in wired telephone networks for such tasks as directory assistance and simple data look-up, and work well as long as the caller is using a wired telephone. Problems develop, however, when a digital wireless, e.g., cellular or PCS, telephone is used. This is because speech processing algorithms in use by all major wireless standards, such as GSM, IS-136, IS-95 and PDC, do not provide for error-free transmission. This results in signal corruption, which appear as muted “blocks” of speech, on the order of 20 ms each. To improve the perceived voice quality at the receiving end, these same systems often perform some form of extrapolation or smoothing operation to make the corruption less noticeable to the human auditory system. Unfortunately, tests have established that the underlying corruption and the follow-on extrapolation or smoothing renders the received speech nearly imperceptible to high-performance server-based speech recognition systems. Prior art systems and methods do not offer a meaningful solution to the aforementioned problem, however, a number of attempts have been made to provide speech recognition systems and GSM communications, although very little work has been done to combine the two fields of art. 
     U.S. Pat. No. 4,058,838, granted Nov. 15, 1987 to Crager et al., for Packet-switched facsimile communications system, describes full duplex communications between a number of communications devices, using a store-and-forward protocol. 
     U.S. Pat. No. 4,624,008, granted Nov. 18, 1986 to Vensko et al., for Apparatus for automatic speech recognition, describes a technique for recognizing sentence end based on pause length. 
     U.S. Pat. No. 4,649,567, granted Mar. 10, 1987 to Childress, for Dispatch overdialing for inter-group and other added calling/called access to communications channels in a trunked radio communications system, describes as system enabling radio transceivers, already operating with a trunked system, to communicate with additional transceivers. 
     U.S. Pat. No. 4,975,957, granted Dec. 4, 1990, to Ichikawa et al., for Character voice communication system, describes the extraction of parameters at the handset and the transmission of codewords as data to a base station which reconstructs the speech, and focuses on transmission of parameters as a bandwidth-saving strategy, and the algorithm presented, assuming error-free codeword transmission, will likely result in significant voice quality degradation. 
     U.S. Pat. No. 5,406,617, granted Apr. 11, 1995, to Bauer, for Cordless telephone/entry intercom system, describes a radio-based intercom system wherein the base station acts as a repeater for the wireless system components. 
     U.S. Pat. No. 5,432,883, granted Jul. 11, 1995, to Yoshihara, for Voice coding apparatus with synthesized speech LPC [linear prediction coefficients] code book, describes a system for coding speech based on LPC and error minimization. 
     U.S. Pat. No. 5,515,375, granted May 7, 1996, to DeClerck, for Method and apparatus for multiplexing fixed length message data and variably coded speech, describes a voice coding techniques wherein a variable rate vocal encoder receives and encodes speech. 
     U.S. Pat. No. 5,570,389, granted Oct. 29, 1996, to Rossi, for Method for reliable exchange of modem handshaking information over a cellular radio carrier, describes a technique for sending an initial FSK-encoded modem handshake. 
     U.S. Pat. No. 5,600,649, granted Feb. 4, 1997, to Sharma et al., for Digital simultaneous voice and data modem, describes a system incorporating a PC for system control, and which allows voice communication, voice mail, EMail, facsimile management, and other communications functions. 
     U.S. Pat. No. 5,684,791, granted Nov. 4, 1997, to Raychaudhuri et al., for Data link control protocols for wireless A TM access channels, describes on-demand available bit-rate data burst transmission in a time division multiple access channel to confirm data accuracy. 
     U.S. Pat. No. 5,737,716, granted Apr. 7, 1998, to Bergstrom et al., for Method and apparatus for encoding speech using neural network technology for speech classification, describes a neural network VRS which operates in single or multi stages. 
     U.S. Pat. No. 5,754,734, granted May 19, 1998 to Emeott et al., for Method of transmitting voice coding information using cyclic redundancy check bits, describes a techniques for prioritizing encoded speech packets prior to error checking. After error checking, the packets are interleaved for transmission. 
     SUMMARY OF THE INVENTION 
     A speech encoding system for use with a digital cellular communication device and a receiving station, includes a mechanism for determining whether a voice communications packet needs to be treated as a data communications packet; a voice recognition mechanism for receiving instructions by voice command; and a control mechanism for responding to said voice command and controlling a controlled entity. 
     A method for encoding a voice command generated on a digital cellular communication device and transmitted over a wireless communication network to a receiving station for controlling a controllable entity includes recognizing a voice command; determining whether the voice command needs to be treated as a data communications packet; encoding the voice command; connecting the voice command to a voice recognition mechanism; and controlling a controlled entity with the voice command. 
     An object of the invention is to provide error-free voice transmission for providing voice control of a controlled entity. 
     Another object of the invention is to provide a voice recognition system for use with a digital cellular phone system. 
    
    
     These and other objects and advantages of the invention will become more fully apparent as the description which follows is read in conjunction with the drawings. 
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a block diagram of the system of the invention. 
     FIG. 2 is a block diagram of the method of the invention. 
     FIG. 3 is a block diagram of conventional wireless signal blocks. 
     FIG. 4 is a block diagram of a signal block used by the invention. 
     FIG. 5 is a block diagram of a non-voice-over-IP protocol of the invention. 
     FIG. 6 is a block diagram of a voice-over-IP protocol of the invention. 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
     The invention disclosed herein provides a method of transferring error-free speech to the server-based voice recognition system. In conventional cellular, analog or digital, and PCS networks, voice and data are handled in a fundamentally different manner. Voice may or may not be coded in such a way as to allow some degree of error detection and correction at the receiving end. However, in no event does the receiving end ever request re-transmission of voice transmissions. The reason is that retry attempts would result in unpredictable delays which are probably less tolerable than occasional speech drop-outs. 
     On the other hand, data transmissions, which may include control messages tochange frequency or power level, are generally supervised and as such, protocols exist which allow retransmission in the event that a data message is not received, or is so corrupted as to not be intelligible. 
     The invention applies to any wireless digital voice communication system and provides for the intermittent special handling of voice information such that, at times when the caller is providing inputs to a speech recognition system, voice transmissions are handled like data transmissions, and thus arrive error-free at the receiving end, ready for submission to the voice recognition system (VRS), also referred to herein as a voice recognition mechanism. 
     Referring now to FIG. 1, the system of the invention is depicted generally at  10 . System  10  includes a mobile handset  12 , which is a digital cellular telephone or PCS. Handset  12  includes a display  14 , a keypad  16 , a set of left-side buttons  18 , and a push-to-talk button  20 , which feature is unique to a handset of an invention, and is used in one embodiment of the invention. Handset  12  is in wireless communication with a mobile telephone switching office  22 , or receiving station, which includes a VRS  24 . System  10  further includes a control mechanism  25  connected to VRS  24  for controlling a controlled entity  26 . The communications link between VRS  24  and control mechanism  25  may be any form of communications system. Handset  12  includes a HQ generation mechanism therein for generating voice command HQ data which is ultimately used to control controlled entity  26 . System  10  is generally part of a telecommunications network which provides wireless and wired communications. 
     One embodiment of the invention includes “push-to-talk” button  18  on the handset. In this embodiment, a user is required to push button  18 , also referred to herein as a high-quality button, when issuing speech commands that are submitted to VRS  24 . While button  18  is pressed, digitized speech packets are treated like data messages, i.e., high quality, supervised transmissions, and office  22  can request re-transmission of any lost HQ data packets. Upon successful arrival of all packets, the network reconstructs the speech command and applies it to the terminating equipment, which includes control mechanism  25 , having some type of intelligent voice response (IVR) system connected to controlled entity  26 . 
     In another embodiment, for use in a more tightly integrated network/IVR system, the PTT button is not required. Both embodiments are depicted in FIG. 2, generally at  30 . A user initiates a call, block  32 , that may include voice commands. In a system constructed according to the first embodiment, the users depresses button  18 , block  34 , to instruct handset  12  to convert speech packets to data packets, block  36 . The data packets, once fully received by office  22 , are forward to VRS  24 , block  38 , and then transmitted, block  40 , to a controlled entity. Once the transmission is completed, the user releases button  18 , and handset  12  returns to normal voice mode, block  42 . 
     In the second embodiment, the network is informed by the IVR that high quality speech inputs are needed, block  44 . At this point, the network places handset  12  in automated response/query (ARQ) voice mode, block  46 , for the duration of the command entry, resulting in high quality transmissions. The system then works as described in conjunction with the first embodiment, returning to normal quality voice mode at the end of the command sequence. 
     The invention may be applied to the scenario wherein speech communication takes place over an IP (Internet Protocol) network, where the IP voice packets, normally transferred by unreliable UDP (User Datagram Protocol), are now transferred by a reliable transmission protocol, such as TCP (Transmission Control Protocol), while the PTT button is pressed, or when the handset is placed in ARQ mode by office  22 . This mechanism allows the reliable transmission of speech commands using the TCP retransmissions. This scenario requires no special support from network infrastructure because the IP network is transparent to any data transferred on IP packets. 
     The speech encoder/decoder used with an IP network, with its higher bit rate, may be used during the retransmission period in order to improve the speech quality. Because the communication is not real-time for that period, speech in any data rate may be transmitted regardless of the available physical bandwidth. 
     Referring now to FIG. 3, a typical encoding sequence for GSM speech communications is depicted generally at  50 . The sequence includes thirteen blocks, wherein speech is broken into four-block normal speech units, “S”,  52 ,  54 , and  56 , each lasting approximately 20 ms, followed by a data block, “X”,  58 . Data block  58  is a slow associated control channel (SACCH). The blocks shown in FIG. 3 represent a total of 60 ms of transmission. Each individual block last approximately 4.615 ms. 
     Turning now to FIG. 4, an encoding sequence according to the invention is shown generally at  60 . The sequence begins with a four-block normal speech unit  62 . At some time during speech unit  62 , the high-quality sequence is triggered,  64 , either by the user pressing the high-quality button, or by automatic detection by IVR. (A conventional data block  66  is still transmitted as every thirteenth block throughout the transmission, although only one data block  66  is depicted in the figure.) Handset  12  indicates to the user that it is ready to begin high quality transmission following the transmission of data block  66 , and signals the user by some form of starting indicator  68 , such as a beep, or other starting confirmatory tone, generated by a notification mechanism in handset  12 , to notify a user that handset  12  is in an HQ data acquisition mode. A start negotiation sequence “N”  70  commences while handset  12  negotiates with the network to begin error-free, high-quality transmission, in the form of a link access protocol, known as an L 2  protocol. The mechanism of sending supervised messages in time slots normally allocated for unsupervised speech is similar to the manner in which Fast Associated Control Channels (FACCH) operate in handovers typical in analog and digital cellular networks. Afterwards, the user speaks instructions, which are sent by HQ transmission. Because of the high likelihood of interface-induced errors, periodic retransmission of HQ speech may be required. The HQ frames, “Q”  74 , will typically encounter some queuing delay, and are thus termed “queued speech.” The high-quality sequence is indicated as being over by the user releasing the high-quality button, or by the IVR providing an appropriate signal or command, as indicated by arrow  72 . If office  22  does not receive all of the HQ frames error-free, it requests a re-transmission of missed frames  76 , and does so until all HQ frames are received error-free. An ‘end negotiation’  78  occurs at the end of the error-free transmission, and after all information has been successfully exchanged. 
     The L 2  connection is released, and an ending indicator  82 , such as an ending confirmatory tone, is generated and transmitted by handset  12 , after a period of time Δt,  80 , which is determined by an internal timer in handset  12  and on the basis of the number of HQ frames that handset  12  must transmit. Ending confirmatory tone  82  is generated by the notification mechanism to notify a user that handset  12  is no longer in the HQ data acquisition mode. Only after all of the HQ blocks for the HQ sequence are acquired will the speech decoder output the audio to the IVR system. Normal speech blocks  84  then resume. Blocks N and Q may be of any length needed to transmit the high-quality information, which include queuing delays, and any time required for re-transmission of data that includes errors. At some point, the voice recognition system ‘decodes’ the HQ speech into instructions for a controlled entity. 
     Two specific embodiments of the system of the invention will now be described. The first embodiment provides non-voice-over-IP protocol, while the second embodiment is a voice-over IP protocol. Turning initially to FIG. 5, a system utilizing a non-voice-over-IP protocol is depicted generally at  90 . A voice input  92  is picked up by a voice coder  93  in handset  12 . Assuming the high-quality function has been initiated, a HQ switch  94  (a.k.a. PTT button  20 ) is in its HQ position, and routes a signal to a queue  96 . Were switch  94  in its normal position, the signal would be sent directly to a media access controller (MAC)  98 . With switch  94  in its HQ position, the signal transits queue  96  and is processed by layer-two (L 2 )  100  prior to being sent to MAC  98 . A digital signal  101  is sent to a data coder  102 , then to a second L 2   104 . A slow associated control channel (SACCH)  106  transmits the data signal to MAC  98 . 
     The signal(s) is transmitted wirelessly to a second MAC  107 . A switch  108  is set to route the signal to an inbound L 2   109 , a speech decoder  110 , or to an extrapolator  112 . If the signal is routed to L 2   109 , it enters a queue  114  until the entire HQ signal is received. The HQ signal is then sent to speech decoder  110 . The signal is output to a receiver by a transducer  116 . At the start of the HQ mode, L 2   109  sends the entire signal as time-contiguous speech to queue, or buffer,  114 . At the end of the HQ mode, the entire captured buffer contents are send to speech decoder  110 . 
     A system using voice-over-IP protocol is shown generally at  120  in FIG.  6 . In this embodiment, a signal is generated by handset  12 , and the signal is sent to a voice coder  122 , which send the signal using transmission real-time protocol (RTP)  124 , which manages the relative timing of the voice packets and the information regarding those packets. If HQ switch  126  is set to normal, the signal is sent by user datagram protocol (UDP)  128  and then by, in the preferred embodiment, internet protocol (IP)  130 , wirelessly, over the world-wide communications system known as the Internet  132 . If HQ switch  126  is in the HQ position, a TCP connection, in the preferred embodiment, is established, and handset  12  generates a confirmatory signal to the user. The signal is assembled, and then sent by TCP  136  over internet  132 . When HQ switch  126  returns to its normal position, the HQ mode terminates and TCP  136  breaks the TCP connection. In this system, there is no distinction between voice and data transmissions. It should be appreciated that any reliable transmission protocol, TCP or otherwise, may be used. 
     On the receiving end, the signal, is sent to a TCP  138  decoder or a UDP decoder  140 . Each packet contains a tag identifying the packet as requiring routing to the TCP decoder or routing to the UDP decoder. Alternately, the packets may be directed to both the TCP and UDP decoder, and the ‘wrong’ decoder simply will ignore the packet. If the signal is appropriate for the TCP, i.e., contains HQ data, the signal and its data are stored in a queue  142 . A receive HQ switch  144  will be set to be in contact with queue  142 , or in contact with UDP  140 . Queue  142  passes the HQ data to RTP  146  only after all HQ data is received and the TCP connection is broken. The signal reaches RTP  146 , is sent to voice decoder  148 , and becomes an output  150 , either in the form of data instructions or voice. The UDP is less reliable than the TCP, however, it has less delay time than a TCP transfer, less overhead, and is operable to provide real-time communications. 
     Although a two embodiments of the invention have been disclosed, it will be appreciated that further variations and modifications may be made thereto without departing from the scope of the invention as defined in the appended claims.

Technology Classification (CPC): 6