Process and device to acknowledge the disabling tone of the echo suppressor in telecommunication systems

Abstract: The device, object of this invention, foresees the sampling and digitization of an input signal x(n) and its filtering through a passband filter (BP) and an Hilbert transformer (HT). This last unit has the same mask of the passband filter (BP) but introduces a 90° phase shifting versus the output of this filter. Signals coming out from these units reach the input of identical decimator devices (M1, M2) reducing the sampling frequency. The signals so treated are applied at the input of a processing block calculating the power of the filtered complex signal. Parallely, the input signal x(n) is treated by a block (.) 2 suitable to produce the square and afterwards, by a FIR filter used to perform the arithmetic mean of samples raised to the square on a number of samples equal to the decimation factor of a decimator (M3) at the output of which a signal, representing the power of the input signal, corresponds.

Description: Field of the Invention The invention relates to the process for the acknowledgement of the disabling tone of the echo suppressor in telecommunication systems and to a device implementing this procedure. Elimination of the known echo phenomenon during voice telephone communications is implemented in telephone exchanges through two different typologies of devices. The first typology is called echo canceller and essentially consists of adaptive digital filters which, operating on voice signals in digitized form, allow to obtain the original wave shape of the voice signal without the echo (effect). The second typology of devices is called echo suppressor and essentially consists of a high impedence device which is inserted in the telephone line to attenuate the echo signal as much as possible. As it is known, the presence of both the device typologies can originate drawbacks during communications of non voice type, as data transmission, telefax connections, etc.. In these cases, it shall be possible to automatically disable the devices. The present invention relates to a procedure and a device to acknowledge the disabling tone of the above mentioned second device typology (echo suppressor). In fact, the line terminal equipments of transmission systems, when a non voice call is established send a predeterminated signal or tone to control the disabling of the echo suppressor. Conventionally this predeterminated tone consists of a sinusoidal signal having 2100 Hz frequency. The device shall therefore be able to acknowledge the presence of this tone and, following the acknowledgement, disable the echo suppressor. However, the check of the presence or not of this tone is not easy to perform, mainly due to the noise which can be superimposed to the useful signal, and to possible variations of the tone characteristics versus the above mentioned nominal values, within some tolerances. In fact, CCITT G.164 and G.165 specifications require that the signal in presence of which disabling must be possible, can be a tone within the frequency band 2079 to 2121 Hz (that is 2100 Hz ± 21 Hz). Furthermore, in a widened band: 1900 Hz to 2350 Hz, disabling can be possible but out of this widened band, disabling can not be possible. The tone presence acknowledgement system shall remain in operation also in presence of a white noise in the band 300 to 3400 Hz together with the 2100 Hz disabling signal. A white noise having level ≥ -5 dB versus the disabling tone of 2100 Hz must be tolerated by the acknowledgement system. Vice versa, should the level of the white noise be higher than the 2100 Hz disabling tone level, the echo suppressor disabling shall be inhibited. A 2100 Hz level tone higher than - 30 dB must be accepted while a tone having level lower than - 35 dB shall be rejected. The enabling operational time of the echo suppressor disabling system shall be lower than 400 ms. Background Art Some solutions have been proposed for the acknowledgement of the disabling tone of the echo canceller, that is solutions complying with G165 CCITT specifications have been proposed. On the contrary, the applicant is not aware of proposals for the acknowledgement of the echo suppressor disabling tone, that is, he is not aware of solutions complying with G164 CCITT specifications. However, it must be considered that the procedure and device, scope of this invention, though having preferential application for the acknowledgement of the echo suppressor disabling tone, can find advantageous application also for the acknowledgement of the echo canceller disabling tone. For sake of clearness, we shall refer to a device for the acknowledgement of the echo suppressor disabling tone, but this statement has not to be considered as restrictive, being the subject device advantaegeously applicable also for the acknowledgement of the echo canceller disabling tone in compliance with Reccomandation CCITT G164. In the paper "A simple phase-reversal tone disabler" by J.L. Melsa and R.J. Raglan, IEEE 1986, ICASSP 86, TOKYO, 48.22.1-48.22.4, page 2619 through 2622, an algorithm has been proposed for the acknowledgement of the echo canceller disabling tone capable to operate also in presence of noise. This method foresees the use of one filter only, and, to consider the possible shifting in frequency of the tone (within allowed limits) it foresees the presence of a compensation circuit. However, the algorithm at the basis of this method requires for its implementation a delay unit, a phase corrector, two low-pass filters and a phase discriminator. The implementation of all these functions on a DSP (Digital Signal Processor) results complex and requires the accomplishment of floating point arithmetic calculations which poorly suit to the characteristics of a DSP having normal performances and limited cost, and make difficult the implementation of real time acknowledgement systems. A tone disabling acknowledgement procedure considerably less expensive and intrinsecally more reliable also in presence of heavy noises or when the tone has a signal level particularly low is disclosed in WO-A-9418759 in the name of the Applicant. According to this procedure, a digitized sample sequence x(n), is filtered through a bandpass digital filter (BP) and at the same time through a Hilbert transformer (HT) with similar characteristics and the two signals 90-degree phase shifted are both sub-sampled through identical decimators. Two components of a signal (complex) are thus considered, whose module or power, records the possible phase-reversal event of the disabling tone as sharp temporary reduction of amplitude. The event can be identified employing values assumed of the complex signal magnitude in three . subsequent sampling moments at least. While achieving a reliability degree complying with CCITT Recommendations G.164 and G165 , and not requiring the calculation of inverse trigonometric functions (typically tan -1 ) to calculate the phase skip value, the system maintains a rather favourable computation complexity, in consideration of processing times which can be obtained employing a common DSP (Digital Signal Processor). Object of the Invention It is one object of the present invention to find out a method for the acknowledgement of the echo suppressor or echo canceller disabling tone which while assuring a high reliability, can be implemented employing a common DSP and a maximum processing time, compatible with a real time implementation of the acknowledgement function of the disabling tone presence. A further object of the invention is to enable the implementation of the acknowledgement process of the echo suppressor or echo canceller disabling tone with simplified hardware, at low cost. Summary of the Invention The invention fully reaches these objects through an acknowledgement method of the echo the echo suppressor or echo canceller disabling tone as set forth in claim 1. According to an aspect of the invention, the compliance of the acknowledgement system with some criteria of CCITT Recommendations can be better assured performing preventive adequacy tests of the filtered and sub-sampled complex signal level and of the same signal averaged on a sufficiently long observation period (for instance 8-10 times the decimation factor) before enabling the relevant comparisons with similar sub-sampled, non filtered signals. This to the purpose to reject disabling tones under the minimum level (-35 dB) determined by CCITT Recommendations. Preferably the length of filters employed is equal to the decimation factor (M) so that the input sequence can be directly processed per blocks having length M without being stored. As it shall be pointed out in the description hereafter, the algorithm of the invention is such to require the execution of three Droducts and three additions for each sample PCM at input, in practice required for the updating of three accumulators: a first accumulator relating to the measurement of the digitized signal power, a second accumulator relevant to the bandpass digital filter and the third accumulator relevant to the corresponding Hilbert transformer. Accumulators can have for instance 32 bit capacity. The performance of these simple operations requires a very reduced processing time. The above mentioned objects are also achieved through a device for the acknowledgement of an echo suppressor or an echo canceller disabling tone in a telephone exchange, as set forth in claim 11. Brief Description of the Drawings The features of the present invention which are believed to be novel are set forth with particularity in the appended claims. The invention, together with further objects and advantages thereof, may be understood with reference to the following description, taken in conjunction with the accompanying drawings and in which: Figure 1 is a block diagram of the invention device; Figure 2 shows the mask of the frequency response of the low pass filter (BP) and of the Hilbert transformer (HT) of the diagram in Fig. 1; Figure 3 is a flow chart of the invention procedure according to a preferred realization form; Figure 4 is a flow chart of the program implementing the invention device; Figure 5 is a flow chart of operations performed in the Identification of the general diagram of Fig. 4; Figure 6 is a flow chart showing the operations performed by the Release block of the general diagram of Fig. 4. Detailed Description of a Preferred Embodiment Referring to the diagram in Fig. 1, x(n) represents a digital signal obtained sampling the telephone signal received at the telephone exchange, at a given sampling frequency which can be typically of 8KHz and digitizing said samples in the known way. The input digitized signal x(n) is filtered at the same time through a bandpass filter (BP) and through a Hilbert transformer (HT). The filter (BP) has a mask (response in frequency) shown in Fig. 2. Preferably the non attenuated band is included between 2000 and 2200 Hz, and therefore it is narrower than the band in which the disabling signal could be accepted, according to CCITT Recommendations. The Hilbert transformer (HT), essentially has the same mask of the bandpass filter (BP) as far as the response in frequency module is concerned, but shows a 90° phase shifting versus the filter (BP) output. Output signals of blocks (BP) and (HT), y1(n) and y2(n), respectively are processed through identical decimator devices M1 and M2 reducing in the same way, for instance 256 times, the sampling frequency, that is they employ a sample (a(m) and b(m)) every 256 samples virtually received at input, that is m = n/256. Signals coming out from blocks M1 and M2 (a(m) and b(m)) are 90° phase shifted between them and represent the components of a complex filtered signal derived from the input digital signal x(n). The two flows a(m) and b(m) reach the input of a processing block calculating the power of the filtered complex digital signal, producing at output the digital signal indicated with P u (m). Likewise, the input digitized telephone signal x(n) is treated in a block (.) 2 suitable to produce the relevant square and then, in a filter FIR suitable to make the arithmetic mean of samples raised to square on a number M of samples equal to the decimation factor of the decimator device M3 following in cascade the filter FIR. The decimation factor (Mi) of the device M3 is identical to that of devices M1 and M2 and produces at output a sub-sampled digital signal (Px(m)) representing the value of the power of the non filtered input digitized signal x(n). Of course, the architecture shown in the block diagram of Fig. 1, from the conceptual point of view, can be simplified realizing BP and HT components with FIR filters and calculating digital flows a(m) and b(m), according to techniques which are known by a skilled in the art. Using filters having length equal to the decimation factor Mi, the input sample x(n) can be directly processed without the need to store it, making in practice a processing of blocks having Mi length of the input signal. The choice of a decimation factor Mi of 256, and therefore of a filter length equal to 256, facilitates the implementation of filters with very narrow non attenuated band and a strong attenuation (>46 dB) in stop band. As shown in Fig. 2, the non attenuated band can extend from 2000 to 2200 Hz, even through the limits required by CCITT Recommendations are 1900 and 2350 Hz, respectively, thus obtaining an accentuated noise rejection. It is clear that neglecting the noise presence, a simple comparison between the two signals P u (m) and P x (m) representing the filtered signal power and the non filtered one respectively, would enable to determine the presence or lack of the echo suppressor disabling signal. A particularly simple algorithm has been found, perfectly able to assure the reliability of the function of disabling tone presence acknowledgement even in presence of noise, within the limits of the above mentioned CCITT Recommendations. Therefore, according to an important and preferred embodiment of the invention, the acknowledgement procedure is that shown as flow chart in Fig. 3. Considering that the digital signal P u (m), representing the power of the input digital signal filtered by blocks BP and HT shows oscillations due to the noise allowed by specifications even when the sinusoid forcing to 2100 Hz has comparatively constant amplitude and that extreme values of these oscillations are of about -3 dB and 1.76 dB, the P u (m) samples are pre-processed to see if they exceed or not a first threshold value, purposely widened to >-34 dB, for instance. Only if the sample is included in the allowed widened range, a first comparison is enabled between the P u (m) signal and the P x (m) signal, representing the power of the digitized, non filtered, input signal, fixing a first increased discrimination threshold R 1 which can be set around 2.5 approx. This first comparison processing among the sub-sampled power signals of the digitized, filtered and non filtered signal, made on single pairs of samples of said sub-sampled signals, can be iteratively made for a given number of times. It has been found that a maximum number of "turns" of this internal processing loop can be fixed 6 to 12, and preferably at 8 (jmax = 8), even if a lower or higher number of cycles versus these limits can be fixed. The flow diagram of Fig. 3 begins therefore with a phase 1 where j=0. In phase 2, a time index which is a function of the decimation factor of said M units is indicated with m. In phase 3 said pre-processing operation aiming at inspecting the exceeding or not of a first threshold value is made. In phase 4 the comparison is made between the P u (m) signal and the P x (m). In phase 5 the j value is increased by one unit and in phase 6 the comparison is made between the present j value and the pre-set jmax value. For this observation period, defined by the maximum number of tours of the internal loop, it is also calculated the average value (P^ u ) of the power P u (m) of the filtered signal and this mean value (P^ u ) is previously tested in phase 8 to define if it is included in a narrow allowed range, defining for instance a threshold >-33 dB (narrower than the >-34 dB widened threshold of the similar test conducted on the single sample). Also in this case, the non exceeding of the threshold initializes the process shown in Fig. 3. Processing contemplates also a phase 9 in which the calculation of the average value (P^ x ) is made, for the same observation interval, of signal P x (m), representing the power of the digitized, non filtered, input signal. The final comparison between mean values of the filtered signal power (P^ u ) and of the non filtered signal power (P^ x ) is made in phase 10, determining a lower discrimination threshold R 2 , for instance around 1.5. If the comparison made in phase 10 gives positive results, the flow chart terminates in a phase 11 where it is stated the presence of the 2100 Hz disabling tone. It was found that the above mentioned algorithm can assure the acknowledgement of the presence of an echo suppressor disabling signal in a completely reliable way, in any condition included in the above mentioned CCITT G.164 Recommendations. EXAMPLE The device of the invention has been implemented in the frame of a development plan aiming at realizing the following functions for eight PCM junctions: "soft processor" for the electrical echo generated by the fix telephone user, in compliance with the Recommendation DECT ETD 300 175-8 clause 7.10, requirement 2; generation of an Artificial Echo, relating to the mobile telephone user, in compliance with the Recommendation DECT ETD 300 175-8 clause 7.4.1.2, requirement A; disabling of the previous functions with 2100 Hz tone, in compliance with the Recommendation CCITT G.164. The component used to implement the processing program is microprocessor 1610 DSP of AT&T, particularly suitable to the digital treatment of a voice signal, having the following main characteristics: Instruction cycle 25 ns; Parallel multiplier with 16 bit inputs and 36 bit accumulator; 16 bit RAM internal memory for programs and data; Input and output registers for the acquisition of PCM data, synchronization and performance control. The program and data memory of the microprocessor employed for the device implementation has been: 16 bit, 180 words program memory; 16 bit, 520 words constant data memory; 16 bit, 14 words variable data memory; The flow diagram followed in the program implementing the device of the invention is shown in Fig. 4. With n we indicate the time index (absolute) or the module 256 time index, according to the context. BP(.) and HT(.) indicate coefficients relevant to the band pass filter (BP) and to the Hilbert transformer (HT). Flow diagrams of operations performed by the "Identification" block and by the "Release" block of the general diagram of Fig. 4, are given as flow diagrams in Fig. 5 and 6, respectively, not described in detail since the same can be easily understood considering the previous descriptions referred to flow diagram of Fig. 3. As shown in Fig. 4, typical operations made for each sample PCM at input, that is every n<256, are: updating of a 32 bit accumulator P x (m), relevant to the power measurement of the digitized voice signal; updating of a 32 bit accumulator a(m), relevant to the coefficients of the bandpass filter (BP); updating of a 32 bit accumulator b(m), relevant to the coefficients of the Hilbert transformer (HT). In total, arithmetic operations made are three products and three additions, In the implementation through the 1610 DSP processor, 55 instruction cycles are employed, including also updating instructions of variable data pointers. Operations made every 256 PCM samples, that is each n = 256, are those relevant to the acknowledgement of the 2100 Hz disabling tone or at disabling release . Only in case of execution of the acknowledgement block, operations made assume some complexity, as per Fig. 5. Always in the implementation with DSP1610 and following the flow diagram shown in Fig. 5, 160 instruction cycle max. result employed in the worst case. Computer simulations and tests made on the device so implemented with DSP1610, have checked in all the cases foreseen by Recommendations CCITT G.164, the correct operation of the invention algorithm. According to the example described above, the device of the invention requires for each PCM sample at input, the execution of a minimum number of arithmetic operations and therefore a typical processing time resulting of about one third of the time available in a tone acknowledgement phase of 2100 Hz. Furthermore, the tone acknowledgement is made only every 256 PCM samples. In the case several independent devices are realized with a unique microprocessor, the total processing time required between a PCM sample and the next one is equal to the sum of typical processing times of each device plus the acknowledgement time of only one of them. In fact, it is sufficient to adequately initialize pointing indexes of coefficients of the bandpass filter (BP) and of the Hilbert transformer (HT) of each device to avoid to execute the 2100 Hz disabling tone identification routine in coinciding PCM frames.

Claims: Method for the acknowledgement of the presence of a disabling tone of an echo suppressor or an echo canceller on a telephone line, including the steps to sample and digitize the received telephone signal, obtaining an input digital signal (x(n)) consisting of a sequence of digital signals, obtaining from said input digital signal (x(n)) a complex signal (a(m),b(m)) by filtering the same through a bandpass filter and a Hilbert transformer having an essentially identical response in frequency and obtaining two signals (y1(n),y2(n)) respectively with the same phase and 90-degree phase shifted with respect to the phase of said digitized signal and subsampling (M1, M2) the two signals (y1(n),y2(n)), characterized by the fact that the presence of said disabling tone is acknowledged from the continuous repeated exceeding of two predetermined discrimination thresholds, a first threshold comparing a ratio of a first pair of values (P x (m) and P u (m), respectively of a first sub-sampled signal (p x (m)), representing the power of said input digital signal, and of a second similarly subsampled signal (p u (m)), representing the power of the complex signal and a second threshold comparing a ratio of a second pair of values p^ x and p^ u of said first and second sub-sampled signals P x (m), P u (m), averaged in a predetermined observation period. Method according to claim 1, characterized by the fact to include the steps of Filtering said input digital signal (x(n)) through a bandpass filter and a Hilbert transformer having identical mask, obtaining two quadrature components (y1(n), y2(n)) constituing a filtered complex signal; sub-sampling with a decimation factor M, of said two components (y1(n), y2(n)) obtaining two sub-sampled components (a(m), (b(m)) of said filtered complex signal; summing the squares of said two sub-sampled components (a(m), (b(m)), obtaining a signal representing the power of said complex filtered and sub-sampled signal; comparing a multiple (R1) of the power of said filtered complex and sub-sampled signal (P u (m)) evaluated every M samples of said input sequence with the power of a non filtered and sub-sampled signal with a decimation factor M(P x (m)) obtained from said input sequence and evaluated every M samples, fixing a first discrimination threshold; comparing values derived from the power of said two signals, similarly sub-sampled, evaluated every M samples and averaged on a rather long observation period, fixing a second discrimination threshold, than the first discrimination one; acknowledging the presence of said disabling tone from the continuous repeated exceeding of both discrimination thresholds mentioned above. Method according to claims 1 and 2, characterized by the fact that the power of said complex filtered signal is submitted to a first processing to define if it exceeds a third threshold level before enabling said first comparison with said first discrimination threshold. Method according to claims 1 and 2, characterized by the fact that the averaged value of the power of said complex filtered signal is submitted to a second processing to define if it exceeds a fourth threshold value higher than said first threshold value, before enabling said second comparison with said second discrimination threshold. Method according to claim 3, characterized by the fact that said first processing and said first comparison are made for a given number of cycles (j) before starting said second processing. Method according to claim 5, characterized by the fact that the number of cycles of said first processing and said first comparison is around 8. Method according to claim 2, characterized by the fact that said first discrimination threshold is increased and defined by a first coefficient (R1) around 2.5. Method according to claims 2 and 6, characterized by the fact that said second discrimination threshold is increased by a coefficient lower than that of augmentation of said first threshold and showing a value around 1.5. Method according to claim 2, characterized by the fact that Filtering of said digitized signal is made with filters of M length without storing the input digitized signal. Method according to claim 1, characterized by the fact that the non attenuated band of said bandpass filter and of said Hilbert transformer, extends from 2000 to 2200 Hz. Device for the acknowledgement of the disabling tone of an echo suppressor or an echo canceller connected to a telephone line, including: a first digital bandpass filter (BP) with non attenuated band centered on the frequency of said disabling tone suitable to receive at input a digitized telephone signal (x(n)); a Hilbert transformer (HT), having filtering characteristics similar to those of said bandpass filter (BP) suitable to receive at input said digitized signal; two decimators (M1, M2) each one connected to the output of one of said bandpass filters (BP) and Hilbert transformer (HT); means suitable to generate a (P u (m)) signal representing the power of a digital complex filtered signal, and obtained from two components (a(m), b(m)) available at the output of said decimators (M1, M2); means suitable to receive at input said digitized telephone signal (x(n)) and to produce at output a (P x (m)) sub-sampled signal with a decimation factor identical to the decimation factor of said decimators (M1, M2) representing the power of said input digitized signal, not filtered; processing means suitable to receive said sub-sampled power signals (P u (m), P x (m)), respectively of the input digitized signal (x(n)), filtered and not filtered to calculate an average (P^ u , P^ x ) of the sub-sampled power signals in a predetermined observation period and to determine the presence of said disabling tone through comparisons among a ratio of said signals (P u (m), P x (m)). representing a first threshold value, and among a ratio of the averaged sub-sampled power signals representing a second threshold value. Device according to claim 11, characterized by the fact that the length of the bandpass filter (BP) and of the Hilbert transformer (HT) is identical to the decimation factor of said decimators (M1, M2). Device according to claim 11, characterized by the fact that said processing means include: a first accumulator for the power value of the input digitized signal; a second accumulator for coefficients of said bandpass filters (BP); a third accumulator for coefficients of said Hilbert transformer (HT).

Inventors: COMPAGNONI DARIO MONZIO, DALMAZIO RAVEGLIA, ORAZIO BIGLIETTI, COMPAGNONI DARIO, MONZIO, DALMAZIO, RAVEGLIA, ORAZIO, BIGLIETTI
Applicants: ITALTEL SPA, ITALTEL S.P.A.
Application year: 1996
Main classification: H04B3
IPC codes: H04B3