Abstract:
An interactive apparatus allows the user to interrupt an outgoing prompt and remove component which is normally found in the users&#39; responses (e.g., a frequency band) from the outgoing output prompt. An input signal analysis unit in the apparatus is able to detect the response of the user (and distinguish it from an echo of the outgoing prompt) by noting the presence of the component which is lacking from the outgoing prompt. As an alternative, the apparatus may force spaced timeslots in the outgoing signal to silence. In that case, the input signal analysis unit can detect the presence of the user&#39;s signal over a predetermined time interval. As well as being applicable to apparatuses which involve the user in prompt/response dialogues, the invention is also useful in relation to the interruption of messages being replayed by voice-controllable answerphones of the like.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to an interactive apparatus. 
     2. Related Art 
     In recent years, an increasing number of everyday telephone interactions have been automated, thereby removing the need for a human operator to progress the interaction. 
     One of the first interactions to be automated was simply the leaving of a message for an intended recipient who was not present to take the call. Recently, more complex services such as telephone banking, directory enquiries and dial-up rail timetable enquiries have also been automated. Many answerphones now additionally offer a facility enabling their owner to telephone them and hear messages which have been left. Another service which has now been automated is the reading of stored e-mail messages over the telephone. 
     In each of the above cases, a user, in effect carries out a spoken dialogue with an apparatus which includes an interactive apparatus, the telephone he or she is using and elements of the Public Switched Telephone Network. 
     In the spoken dialogue it is often useful if the user is able to interrupt. For example, a user might wish to interrupt if he or she is able to anticipate what information is being requested part way through a prompt. The facility enabling interruption (known as a “barge-in” facility to those skilled in the art) is even more desirable in relation to message playback apparatuses (such as answerphones) where a user may wish to move onto another message without listening to intervening messages. 
     Providing a barge-in facility is made more difficult if some of the output from the interactive apparatus is fed-back to the input which receives the user&#39;s commands. This feedback arises owing to, for example, junctions in the network where voice-representing signals transmitted from the interactive apparatus are reflected back to its input. It is also caused by the acoustic echo of the speech output from the speaker of the user&#39;s telephone back to the microphone (this is especially problematic in relation to handsfree operation). There is therefore a need to distinguish fed-back output signals from the user&#39;s input in order to provide a more reliable barge-in facility than has hitherto been possible. 
     According to the present invention there is provided an interactive apparatus comprising: 
     signal output means arranged in operation to output a signal representative of conditioned speech; 
     signal input means arranged in operation to receive a signal representative of a user&#39;s spoken command; 
     wherein the conditioned speech lacks a component normally present in speech; 
     command detection means operable to detect a user&#39;s command spoken during issuance of the conditioned speech by detecting the input of a signal which represents speech including the component lacking from the conditioned speech 
     SUMMARY OF THE INVENTION 
     The advantage of providing such an apparatus is that it is better able to detect the presence of a user&#39;s commands. This is particularly useful in relation to an apparatus which uses a conventional speech recogniser, as the performance of such recognisers falls off sharply if the voice signal they are analysing is in any way corrupted. In an interactive apparatus distortion caused by an echo of the interactive apparatus&#39;s output can cause the user&#39;s command to be corrupted. The present invention alleviates this problem by enabling the apparatus to stop outputting voice-representing signals or speech as soon as the user&#39;s response is detected. 
     In some embodiments, the apparatus further comprises a means for conditioning signals representative of speech output by the interactive apparatus. Because the quality of recorded speech is better than the quality of speech synthesised by conventional synthesisers, many conventional interactive apparatuses use recorded speech for those parts of the dialogue which are frequently used. However, for apparatuses such as those which are required to output signals representing a spoken version of various telephone numbers or amounts of money it is currently impractical to record a spoken version of every possible output. Hence, such outputs are synthesised when required. A recorded speech signal can be pre-conditioned to lack the said component at the time that the speech signal is recorded. Hence, apparatuses whose entire output is recorded speech do not require a means for conditioning the signals representative of speech to be output by the interactive apparatus. Such apparatuses have the clear advantage of being less complex in their construction and are hence cheaper to manufacture. 
     Preferably, the said lacking component comprises one or more portions of the frequency spectrum. This has the advantage that the apparatus is easy to implement. 
     The apparatus is found to be most effective when the portion of the frequency spectrum lies in the range 1000 Hz to 1500 Hz. 
     Preferably, the width of the frequency band is in the range 80 Hz to 120 Hz. It is found that if the width of the frequency band is greater than 120 Hz then the output which the user hears is significantly corrupted, whereas if the width is less than 80 Hz the conditioning of the output of the interactive apparatus is made more difficult and it also becomes harder to discriminate between situations where the user is speaking and situations where he or she is not. 
     According to a second aspect of the present invention there is provided a method of detecting a user&#39;s spoken command to an interactive apparatus, said method comprising the steps of: 
     outputting a signal representative of conditioned speech, wherein the conditioned speech lacks a component normally comprised in users&#39; spoken commands; 
     monitoring signals input to the interactive apparatus for the presence of signals representative of speech including said component; and 
     determining that the input signal represents the user&#39;s spoken command on detecting the presence of signals representative of speech including said component. 
     According to a third aspect of the present invention there is provided a voice-controllable apparatus comprising: 
     an interactive apparatus according to the first aspect of the present invention; 
     means for converting said signal representative of conditioned speech to conditioned speech; and 
     means for converting a user&#39;s spoken command to a signal representative thereof. 
     The problems addressed by the present invention also occur in relation to apparatuses which are directly voice-controlled (i.e. where there is no intermediate communications network). Embodiments of the third aspect of the present invention therefore include, amongst other things, domestic and work-related apparatuses such as personal computers, televisions, and video-recorders offering interactive voice control. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     There now follows a detailed description of a specific embodiment of the present invention. This description is given by way of example only, with reference to the accompanying drawings, in which: 
     FIG. 1 is a functional block diagram of part of an automated telephone banking apparatus installed in a communications network; 
     FIG. 2 is a flow diagram representing the progress of a dialogue with a first time user of the apparatus; 
     FIG. 3 is a diagram illustrating the progress of the same dialogue with a more experienced user; 
     FIG. 4A illustrates the spectrum of the user&#39;s voice; 
     FIG. 4B illustrates the spectrum of the signal output by the apparatus; and 
     FIG. 4C illustrates the spectrum of the user&#39;s voice corrupted by an echo of the apparatus&#39;s output. 
    
    
     DETAILED DESCRIPTION OF EXEMPLARY EMBODIMENTS 
     FIG. 1 illustrates a signal processing unit used in providing an automated telephone banking service. In practice, the speech processing unit will be connected by an FDDI (Fibre Distributed Data Interface) local area network to a number of other units such as a telephone signalling unit, a file server unit for providing a large database facility, an assistant back up and data collection unit and an element management unit. A suitable apparatus for providing such a service is the interactive speech applications platform manufactured by Ericsson Ltd. 
     The speech processing unit (FIG. 1) is interfaced to the telecommunications network via a digital line interface  10 . The digital line interface inputs the digital signals which represents the user&#39;s voice from the telecommunications network and outputs this digital signal to the signal processing unit  20 . The digital line interface  10  also inputs signals representing the spoken messages output by the apparatus from the signal processing unit  20  and modifies them to a form suitable for transmission over the telecommunications network before outputting those signals to the network. The digital line interface  10  is capable of handling a large number of incoming and outgoing signals simultaneously. 
     A signal processing unit  20  inputs the modified signals representing the user&#39;s voice from the digital line interface  10  and carries out a series of operations on those signals under the control of a dialogue controller  30  before outputting a signal representing the spoken response to the user via the digital line interface  10 . The signal processing unit  20  includes four output processors  25 ,  26 ,  27 ,  28  and two input processors  21 ,  22 . 
     The recorded speech output processor  25  is arranged to output a digital signal representing one of a number of messages stored therein which are frequently output by the apparatus. The particular message to be output is determined in accordance with a parameter supplied from the dialogue controller  30 . The speech synthesiser processor  26  is used to output digital signals representing synthesised speech. The content of the spoken message is determined by the dialogue controller  30  which sends alphanumeric data representing the content of the message to the speech synthesiser processor  26 . 
     The signal output by the speech synthesiser  26  is input to a digital notch filter  27 . For reasons which will be explained below, this filter  27  is arranged to remove components of the synthesised signal lying in a frequency band from 1200 Hz to 1300 Hz. It will be realised that by those skilled in the art that although the speech synthesiser  26  and digital notch filter  27  are illustrated as separate processors, the two functions may be provided on a single processor. 
     The messages stored in the recorded speech processor  25  are recorded using a filter with a similar transfer function to the digital notch filter  27 . Thus, the output of the speech synthesiser processor  26  might have a spectrum similar to that illustrated in FIG. 4A, whereas the output of the digital notch filter  27  or the recorded speech processor  25  might have a spectrum similar to that shown by the solid line in FIG.  4 B. 
     The outputs of the filter  27  and the recorded speech processor  25  are passed to a message generator  28  which, for messages which have both a synthesised portion and a recorded speech portion, concatenates the two parts of the message before outputting the concatenated message via the digital line interface  10  to the user. 
     The two input signal processors are an input signal analyser  21  and a speech recogniser  22 . 
     The input speech analyser  21  receives the signal representing the user&#39;s voice from the digital line interface  10  and passes it through a bandpass filter whose passband extends from 1200 Hz to 1300 Hz. Thereafter, the input signal analyser compares the output of the bandpass filter with a threshold T (see FIG.  4 ). If the signal strength in the passband lies above the threshold then the input signal analyser outputs a “user present” signal  23  indicative of the fact that the signal being input to it comprises the user&#39;s voice. On the other hand, if the signal strength within the passband falls below the threshold, then the analyser outputs an alternative version of the signal  23  to indicate that the signal input to the signal analyser  21  does not comprise the user&#39;s voice. 
     The incoming speech representing signal is also input to the speech recogniser  22  which is supplied with possible acceptable responses by the dialogue controller  30 . On the user present signal  23  indicating that the user&#39;s voice is comprised in the input signal, the speech recogniser attempts to recognise the current word being spoken by the user and outputs the result to the dialogue controller  30 . 
     The dialogue controller  30  then responds to the word or word spoken by the user in accordance with the software controlling it and controls the output processors in order to provide the user with a suitable response. 
     A dialogue (FIG. 2) between the automated banking apparatus and an inexperienced user is initiated by the user dialling the telephone number of the apparatus. Once the user is connected to the apparatus the dialogue controller  30  instructs the recorded speech processor  25  to output a welcome message R 1 , immediately followed by an account number requesting prompt R 2 . As mentioned above, all recorded messages and prompts stored within the recorded speech processor  25  are recorded so as to have a spectrum similar to the one illustrated by the solid line in FIG.  4 B. FIG. 4B shows that the spectrum of the recorded messages lacks any components having a frequency between 1200 Hz and 1300 Hz, but is otherwise normal. On outputting the message, it may be that an echo in the message is received back at the input signal processors  21 ,  22 . Although it is likely that the spectrum will be altered slightly by the reflection process, the reflection process will not introduce frequencies which were not present in the outgoing signal and hence will not introduce frequencies in the frequency band 1200 Hz to 1300 Hz. Nevertheless, it is likely that some noise will be added to the output signal whilst it is being transmitted from the output signal processes  25 ,  26 ,  27 ,  28  to the input signal processes  21 ,  22 . Hence, the spectrum of the echo may be similar to that shown as a dashed line in FIG.  4 B. 
     Returning to FIG. 1, the echo of the prompt R 2  is received at the input signal analyser  21  where it is bandpass filtered (the passband extending between 1200 Hz and 1300 Hz), and the resulting signal is compare echo of the outgoing prompt does not contain a significant component in the frequency band 1200 Hz to 1300 Hz, the signal falls below the t hreshold and the input signal analyser  21  outputs the signal  23  indicating, throughout the duration of the prompt R 2 , that the user is not speaking. 
     The user then proceeds to enter his account number using the DTMF (Dual Tone Multiple Frequency) keys on his phone. These tones are received by the speech recogniser  22  which converts the tones into numeric data and passes them to the dialogue controller  30 . The dialogue controller  30  then forwards the account number to a customer database file server provided on the FDDI local area network. The file server then returns data indicating what services are to be made available in r elation to the is account and other data relating to the customer such as a personal identification number (PIN). Although not shown in FIGS. 2 and 3, the system will ask for the customer to enter his PIN immediately after having requested his account number. 
     The dialogue controller  30  then instructs the recorded speech processor  25  to output a type-of-service-required prompt R 3  which the user is tens to before replying by saying the word “transfer”. The user&#39;s voice might have a spectrum similar to that shown in FIG.  4 A. When a signal representing his voice is passed to the input signal analyser  21 , it is found that the signal contains a significant component from the frequency band 1200 Hz to 1300 Hz and hence the input to analyser  21  outputs a signal  23  indicative of the fact the user is speaking to the speech recogniser  22 . T he speech recogniser  22  recognises the word currently being input to the apparatus to be “transfer” and passes a signal indicating that that is the word received to the dialogue controller  30 . 
     As a result of having received this response, the dialogue controller  30  then instructs the recorded speech processor  25  to output a prompt asking the user how much money he wishes to transfer. The user then replies saying the amount of money he wishes to transfer, spoken entry of this information being potentially more reliable than information from the telephone keypad because a mistake in entering the DTMF tones may result in the user requesting the transfer of an amount of money which is an order of magnitude more or less than he would wish to transfer. 
     The user&#39;s response is then processed by the speech recogniser  22  and data indicating how much money the user has requested to transfer (£316.17 in this example) is passed the dialogue controller  30 . The dialogue controller  30  then instructs the recorded speech processor  25  to send the recorded speech messages “I heard” and “is that correct?” to the message generator  28 . The dialogue controller  30  then instructs the speech synthesiser  26  to synthesise a spoken version of £316.17. A synthesised version of these words is output by the speech synthesiser  26  and has a spectrum similar to that shown in FIG.  4 A. The signal is then passed through the digital notch filter  27  and is output having a spectrum similar to the solid line spectrum of FIG.  4 B. The modified synthesised message is then loaded into the message generator  28 . 
     The message generator  28  then concatenates the two recorded speech messages and the synthesised speech message to provide the prompt R 5  which is output via the digital line interface  10  to the user. The dialogue then continues. 
     A user who is more familiar with the system may carry out a dialogue like that shown in FIG.  3 . The initial part of the dialogue is identical to that described in relation to FIG. 2 until the user interrupts the account number requesting prompt R 2 , using his telephone keypad to enter his account number. The DTMF tones output by his telephone are input to the speech recogniser  22  which converts the tones to the account number representing the data and passes that data to the dialogue controller  30 . As soon as the dialogue controller  30  receives this data it sends a signal to the recorded speech processor  25  to halt the output of the account number requesting prompt R 2 . Clearly, once the apparatus has stopped issuing the prompt R 2 , no echo of that prompt will be received back at the apparatus. Hence, the speech recogniser can recognise the other DTMF tones input by the user without the presence of the interfering echo. 
     The dialogue then continues as before until the user interrupts the service required prompt R 3  by saying the word “transfer”. During the first two words of the message R 3 , it will be realised that the input signal analyser  21  will be outputting a signal  23  which indicates that the user&#39;s voice is not present. However, as the user interrupts the output message, the signal received at the apparatus will be a combination of the user&#39;s voice and an echo of the outgoing prompt. The spectrum of this combination signal will be similar to that of the user&#39;s voice alone (FIG.  4 A), but because the spectrum of the echo signal lacks any components between 1200 Hz and 1300 Hz, will feature a small notch between 1200 Hz and 1300 Hz. (FIG.  4 C). 
     The combination signal is passed to the input signal analyser  21  where it is passed through a bandpass filter and found to have a significant component in the frequency range 1200 Hz to 1300 Hz. The input signal analyser  21  therefore outputs a signal  23  (indicating that the user&#39;s voice is present) to both the speech recogniser  22  and the dialogue controller  23 . On receiving the signal  23 , the dialogue controller  30  instructs the recorded speech processor  25  to halt its output of the prompt R 3 . Soon after, the echo of the prompt ceases to be a component for signals received at the speech recogniser  22 , and the recogniser is better able to recognise the word currently being spoken by the user. Once the response of the user has been recognised, it is passed to the dialogue controller  30 . 
     Thereafter, the user interrupts the next two prompts of the dialogue in a similar way to the way in which he interrupted the type-of-service-required prompt R 3 . 
     It will be realised that in the above embodiment, the component lacking from the pre-conditioned spoken prompt comprises a portion of the frequency spectrum. However, it is also envisaged that other components might be lacking. For example, timeslots of short duration (say 1 to 5 ms) could be removed from the spoken prompt at a regular interval (say every 20 ms to 100 ms). If, for example, the speech is digitally sampled at 8 kHz, this might be achieved by setting 8 to 40 samples to a zero value at an 160-800 sample interval. To take a particular value, if 20 samples were to be removed from the signal at a 400 sample interval, then the input signal analyser might be set up such that if it did not detect a corresponding silence or near silence (i.e. where the volume is below a given threshold) during a received signal duration of 800 samples, then it might output a signal indicative that the user is speaking. 
     It will be seen how the “barge-in” facility allows the user to carry out his transaction more quickly. More importantly, by being able to interrupt the prompt issued by the apparatus in this way, the user feels more in control of the dialogue.