Echo Cancellation - Seminar Report

ABSTRACT

  Wireless phones are increasingly being regarded as essential communications tools, dramatically impacting how people approach day-to-day personal and business communications. As new network infrastructures are implemented and competition between wireless carriers increases, digital wireless subscribers are becoming ever more critical of the service and voice quality they receive from network providers. A key technology to provide near-wireline voice quality across a wireless carrier's network is echo cancellation.

Introduction

People have been using phones as a means of distant voice communication for more than a century now.Using phones has become sort of a usual thing. We use the phones almost every day and just about every where  At home,at work,outside,in our cars and so on.

Subscribers use speech quality as the benchmark for assessing the overall quality of a network. Regardless of whether or not this is a subjective judgment, it is the key to maintaining subscriber loyalty. For this reason, the effective removal of hybrid and acoustic echo inherent within the digital cellular infrastructure is the key to maintaining and improving perceived voice quality on a call. This has led to intensive research into the area of echo cancellation, with the aim of providing solutions that can reduce background noise and remove hybrid and acoustic echo before any transcoder processing. By employing this technology, the overall efficiency of the coding can be enhanced, significantly improving the quality of speech. This tutorial discusses the nature of echo and how echo cancellation is helpful in making mobile calls meet acceptable quality standards. 

Causes of Echo 

Acoustic echo apart, background noise is generated through the network when analog and digital phones are operated in hands-free mode. As additional sounds are directly and indirectly picked up by the microphone, multipath audio is created and transmitted back to the talker. The surrounding noise, whether in an automobile or in a crowded, public environment, passes through the digital cellular vocoder, causing distorted speech for the wireline caller. 

Digital processing delays and speech-compression techniques further contribute to echo generation and degraded voice quality in wireless networks. Delays are encountered as signals are processed through various routes within the networks, including copper wire, fiber optic lines, microwave connections, international gateways, and satellite transmission. This is especially true with mixed technology digital networks, where calls are processed across numerous network infrastructures. 

Echo-control systems are required in all networks that produce one-way time delays greater than 16 ms. In today's digital wireless networks, voice paths are processed at two points in the network within the mobile handset and at the radio frequency (RF) interface of the network. As calls are processed through vocoders in the network, speech processing delays ranging from 80 ms to 100 ms are introduced, resulting in an unacceptable total end-to-end delay of 160 ms to 200 ms. As a result, echo cancellation devices are required within the wireless network to eliminate the hybrid and acoustic echoes in a digital wireless call. 

 The Need for Echo Cancellation

People have been using phones as a means of distant voice communication for more than a century now. Using phones has become sort of a usual thing. We use the phones almost every day and just about everywhere: at home, at work, outside, in our cars and so on. A big thanks goes to the cellular phones, which set us free from wires!

Although we all enjoy this remarkable possibility to talk on the phone, there's always something we'd like to be better. This something is the speech quality in the phone conversations. The speech quality has always been an issue for both the phone network service providers and their subscribers. There are several reasons for the undesirable quality degradation and the appearance of audible echoes is one of them. This kind of quality degradation is inherent in the network equipment and the end-user phone devices.

Today it is easy to implement echo cancellation on DSPs and this is what engineers are doing in their devices. However, many of them face certain difficulties with achieving echo cancellation because of incomplete understanding of the echo cancellation principles and not meeting the requirements imposed by the echo cancellers. The purpose of this article is to demystify the topic of the echo cancellation by explaining its basics and providing useful information for those engineers, who need to implement the echo cancellation in their devices. We shall see where the echoes come from, how to fight them and what the known problems with the echo cancellation are. The information provided herein is based on the experience of developing echo cancellers and supporting echo canceller customers at SPIRIT Corp.

There are generally two kinds of the echo, which can appear when talking on the phone. The two differ by the place where they are created and by their characteristics.

Types of Echo’s

Hybrid Echo

The first kind is the line echo (also known as electric or hybrid echo) and it is created by the electrical circuitry connected to the wire lines.

Let us first see a simplified version of a network with two abonents.

On the Figure 1 you can see the network using 2-wire lines to connect the abonents' phones with the switching station. Each of the 2-wire lines between a phone and the switching station carries voice signals in both directions, e.g. from one phone to the other through the station and back. The switching station provides the power supply to feed the microphones and the switching functionality, which is needed if there are more than two abonents.

The 2-wire lines are obviously cheaper than the 4-wire lines and this is why the regular phones and switching stations were designed to operate with each other over 2-wire lines.

The above network is indeed simple and it can operate very well provided the distance between the abonents is short. Now, if we want to make calls between very distant abonents, we need to do something about the signals because of their attenuation in the long lines. So, we need to amplify the signals. But we can't just amplify what is being sent and received over the 2-wire line because there are both signals coming in both directions at the same time. The solution to this is amplification of separated send and receive signals from the 2-wire line. Such a separation is performed by a dedicated electrical device, called a hybrid. The hybrid basically provides a conversion between 2-wire and 4-wire lines (the switching stations are now connected with 4-wire lines). Another reason for signal separation is that the signals can be transmitted over digital networks between the switching stations. Digital transmission improves the quality of the calls and increases the capacity of the phone networks due to digital signal compression. This makes a more efficient use of the network equipment.

Now, the interesting part is the hybrid performance because it is the hybrid where the echo can be and, in fact, is created. Ideally, the hybrid should just have a sum of the send and receive signals on the 2-wire side and these same signals separated on the 4-wire side. But in the reality, there are things like spread of equipment parameters and mismatch of line impedances, which all contribute to imperfect signal separation in the hybrid, which is the cause of the echo creation:

So part of the signal being sent to the hybrid on the 4-wire side is returned back as the echo superimposed on the signal being received from the hybrid on the 4-wire side. If, for example, the left hybrid of the Figure 2 has this kind of impairment, then the right talker will be hearing his own voice in the handset as an echo and the more the distance (and thus the signal delay) between the abonents' phones, the better this echo will be audible.

Since the hybrid echo is inherent in the designs involving 2-to-4-wire conversion, we always need to cancel this echo in the switching stations and any other devices having this kind of conversion.

Regular phones, which are connected by 2-wire lines with the switching stations, may also have this kind of conversion, but there's an excuse for not doing full-blown echo cancellation in the regular phones. The delay in the electrical path between the microphone and earpiece (or speaker) in the phone is essentially zero, so a cheap transformer-based attenuator can be used, and hearing your own undelayed voice of low amplitude does not cause much, if any, of the discomfort. Actually, hearing talker's own voice is desirable as people expect to hear themselves but being not able to do so makes them think that the phone is not working.

However, using echo cancellation is required in the hands-free phones and all phones, which amplify the signal right before the earpiece or loudspeaker. Not doing echo cancellation in such devices leads not only to very well audible echoes for those who make calls to such phones (acoustic echo cancellation will be treated in the following section), but also to self-excitation of the amplifier. The self-excitation results in from non-ideal signal separation in the phone's hybrid, e.g. part of the signal from the microphone reflects at the hybrid to the other signal path and gets amplified by the amplifier, so what the microphone is picking can be heard from the speaker. The acoustic feedback between the amplifier's output and input effectively turns the amplifier to a generator. Therefore, the hands-free and all other amplifying phones (for example, phones for people with hearing impairments, who tend to speak louder) must include line echo cancellers.

Unlike the phones, the dialup modems and faxes always employ built-in echo cancellers to combat the local echo, because these digital devices are much more sensitive to the distortions of the received signals than humans. The same echo cancellation may be desirable in the answering machines, which record the voice from the phone line.

There are a few peculiar properties of the hybrid echoes. One is that the echo path delays are very short and each hybrid has a single echo path. Another is that the echo paths don't change or change very slowly over time because of very slow changes of the electrical circuitry parameters and wire lines parameters in the network.

Acoustic Echo

The second kind of the echo is the acoustic echo. It is easier to understand why and where this echo is created, although as we will see later, this doesn't make it easier to efficiently cancel it.

The acoustic echo is created by the loudspeaker in a phone. The sound comes out of it, bounces the walls, ceiling and other objects in the room, reflects and comes back to the phone's microphone. The same thing is possible to have not only in the buildings, but also in cars, basically, everywhere, where the sound from the loudspeaker can be reflected to the microphone, and this also includes the phone's case as the sound can and, usually does, go from the speaker to the microphone inside the hands-free phone! Similarly, if there's bad acoustic decoupling between the microphone and earpiece in the handset, the acoustic echo will exist in the handset.

  Acoustic echo was first encountered with the early video/audioconferencing studios and—as Figure  shows—now also occurs in typical mobile situations, such as when people are driving their cars. In this situation, sound from a loudspeaker is heard by a listener, as intended. However, this same sound also is picked up by the microphone, both directly and indirectly, after bouncing off the roof, windows, and seats of the car. The result of this reflection is the creation of multipath echo and multiple harmonics of echo, which, unless eliminated, are transmitted back to the distant end and are heard by the talker as echo. Predominant use of hands-free telephones in the office has exacerbated the acoustic echo problem.

Often times, when making conference calls at workplace, we use the hands-free features of our phones, so all of the colleagues participating in the call can hear the other side. The acoustic echoes differ a lot from the hybrid echoes. First of all, the echo path delays aren't short (the echo path delay is the echo path length divided by the wave propagation speed. Electromagnetic waves propagate at about the speed of light in the wires, e.g. 3*108 meters/second, while the sound propagation speed in the air is about 3*102 meters/second. As you can see, the difference is 6 orders of magnitude!). The echo path is determined by the size of the room where the phone is used, and, obviously, the more the room size is, the longer the echo path delay is. And if we don't cancel this acoustic echo, the person who has called to the hands-free phone can hear a very annoying echo, which is delayed by the sum of the acoustic echo path delay in the room plus the round-trip delay in the network between the phones. But longer echo path delays aren't the only interesting feature of the acoustic echo. The other interesting thing, which imposes certain problems on the acoustic echo cancellation, is that there are many echo paths available in the room as the sound now can be reflected by many objects to the microphone and the paths can vary over time as the objects change their locations. Suppose you move around the room or somebody opens or closes the door in it. This makes the effective echo path change

Switching station with hybrid echo canceller

(common wires and microphone power supply not shown)

On the above figure we see the switching station with an echo canceling system integrated on the 4-wire side between points A, B, C and D. The signals (all shown as functions of the sample number, i) are as follows: x(i) is the signal from the abonent connected by a 2-wire line to the switching station (near-end talker signal), y(i) and u(i) are the signals from and to the other abonent (far-end talker), which come through the 4-wire line.

The idea of an echo canceller is simple. The signal from the far-end talker, y(i), when passing through the hybrid's echo path (between points B and A) is affected by the echo path's impulse response and is transformed to the signal r(i), which is the undesired echo. The signal from the near-end talker, x(i), is added to r(i) at point A. The adaptive filter (normally, FIR) used in the system mimics the impulse response of the hybrid's echo path and produces a replica, r'(i), of the echo signal r(i). If r(i) and r'(i) are the same, then they will cancel each other in the summer connected between points A and C and the filter's output. If r(i) and r'(i) aren't the same, the far-end talker will be hearing not only the near-end talker's x(i) signal, but also the difference of r(i) and r'(i), which is called the residual echo error signal.

The residual echo error, e(i) = r(i) - r'(i), is used to adapt the filter's coefficients. That is, the echo canceller is a tracking system with the residual echo error signal used as the feedback and the purpose if this system is to minimize this error.

Obviously, since the echo path's impulse response is unknown, some time is needed for the echo canceller to minimize the residual echo error signal below a required level. This time is called the convergence time. Note that while the far-end talker's signal y(i) is equal to zero, the echo canceller is not able to converge, because both r(i) and r'(i) are zero, thus the feedback is also zero and there's no adaptation possible. This is why the reference signal y(i) should be present in the beginning of the conversation; grabbing the handset and just saying "hello" should be more than enough for the adaptation to proceed.

Note that the adaptation is possible to do when the near-end talker's signal x(i) is close to or is zero, otherwise this signal x(i) will effectively be an additive noise in the feedback, causing the system to become unstable, diverge and stop working. This is why no filter coefficient adaptation is done at all or the adaptation is very slow during the double-talk periods, e.g. when both the near-end and far-end talkers talk simultaneously

It is worth mentioning that such an echo cancellation scheme is naturally linear and is very sensitive to the nonlinearities in the echo path. This linear system will not be able to match the impulse response of a nonlinear echo path and therefore effectively remove the echo.

Good echo cancellation performance can be achieved by using the NLMS (Normalized Least Mean Squares) algorithm, which is also known as the normalized stochastic gradient algorithm, or its many variations. The NLMS algorithm is the most widely used one and it provides a low cost way to determine the optimum filter coefficients. The algorithm minimizes the mean square of the residual echo error signal at each adaptation step (e.g. at each sample), hence the name of the algorithm. Normalization by the signal power is used because speech is a highly non-stationary process.

Without derivations, which you can find elsewhere in the literature on adaptive signal processing, we give the general formula for the coefficient adaptation for the NLMS algorithm:

Where:

i is the sample number

ak is the k-th coefficient of the filter

N is the number of filter coefficients

b is the adaptation step, which controls the convergence time and adaptation quality

e is the residual echo error signal

y is the far-end talker signal

s2 is the reference signal power

The number of coefficients in the filter should be large enough to cover the echo path delay and all additional delays due to the lines and circuitry between the echo canceller and the place where the echo is created (e.g. the total delay between points A and B of the echo canceller as per Figure ). This should also include the dispersion time due to the network elements.

The hybrid echo path delay is known to be short. The time span over which its impulse response is significant, is typically 2 to 4 milliseconds, but usually when canceling the hybrid echo, the number of filter coefficients is chosen to cover hybrid echo path delays up to 16 milliseconds, which is usually the upper bound of the hybrid echo path delay. The 16 ms path needs 128 coefficients at the sampling rate of 8 KHz.

The length of the acoustic echo path, as has already been pointed out, depends on the size of the room, where it exists. So, without going into room measurements, for acoustic echo path delays of up to 256 ms we will have to have 2048 coefficients at the sampling rate of 8 KHz.

In modern telephone networks, echo cancellers are typically positioned in the digital circuit, as shown in Figure 4. The process of canceling echo involves two 

steps. First, as the call is set up, the echo canceller employs a digital adaptive filter to set up a model or characterization of the voice signal and echo passing through the echo canceller. As a voice path passes back through the cancellation system, the echo canceller compares the signal and the model to cancel existing echo dynamically. This process removes more than 80 to 90 percent of the echo across the network. The second process utilizes a non-linear processor (NLP) to eliminate the remaining residual echo by attenuating the signal below the noise floor. 

Today's digital cellular network technologies, namely TDMA, CDMA, and GSM, require significantly more processing power to transmit signal paths through the channels. As these technologies become even more sophisticated, echo control will be more complex. Echo cancellers designed with standard digital signal processors (DSPs), which share processing time in a circuit within a channel or across channels, provide a maximum of only 128 ms of cancellation and are unable to cancel acoustic echo. With network delays occurring in excess of 160 ms in today's mixed-signal network infrastructures, a more powerful, application-specific echo-cancellation technology is required to control echo across wireless networks effectively. 

Controlling Acoustic Echo 

In echo cancellation, complex algorithmic procedures are used to compute speech models. This involves generating the sum from reflected echoes of the original speech, then subtracting this from any signal the microphone picks up. The result is the purified speech of the person talking. The format of this echo prediction must be learned by the echo canceller in a process known as adaptation. It might be said that the parameters learned from the adaptation process generate the prediction of the echo signal, which then forms an audio picture of the room in which the microphone is located. Figure 5 shows the basic operation of an echo canceller in a conference room type of situation. 

During the conversation period, this audio picture constantly alters, and, in turn, the canceller must adapt continually. The time required for the echo canceller to fully learn the acoustic picture of the room is called the convergence time. The best convergence time recorded is 50 ms, and any increase in this number results in syllables of echo being detected. 

Other important performance criteria involve the acoustic echo canceller's ability to handle acoustic tail circuit delay. This is the time span of the acoustic picture and roughly represents the delay in time for the last significant echo to arrive at the microphone. The optimum requirement for this is currently set at 270 ms—any time below this could result in echoes being received by the microphone outside the ability of the echo canceller to remove them, and hence in participants hearing the echoes. 

Another important factor is acoustic echo return loss enhancement (AERLE). This is the amount of attenuation which is applied to the echo signal in the process of echo cancellation—i.e., if no attenuation is applied, full echo will be heard. A value of 65 dB is the minimum requirement with the non-linear processor enabled, based on an input level of -10 dBm white noise electrical and 6 dB of echo return loss (ERL). 

The canceller's performance also relies heavily on the efficiency of a device called the center clipper, or non-linear processor. This needs to be adaptive and has a direct bearing on the level of AERLE that can be achieved. 

 Controlling Complex Echo in a Wireless Digital Network 

Although acoustic echo is present in every hands-free mobile call, the amount of echo depends on the particular handset design and model that the mobile user has. On the market are a few excellent handsets that limit the echo present, but, due to strong price pressures, most handsets do not control the echo very well at all—in fact, some phones on the market have been determined to have a terminal compiling loss of 24 dB. Echo becomes a problem when the processing inherent to the digital wireless network adds an additional delay (typically in excess of 180 ms round-trip). This combination makes for totally unacceptable call quality for the fixed network customer.

This back-to-back configuration ensures a high audio quality for both PSTN and mobile customers. In addition, the echo canceller's software configuration is designed to provide a detailed analysis of background noises, including acoustic echo from the mobile user's end. Some echo cancellers incorporate a user-settable network delay, which enables network operators to fine-tune the echo control to suit their parameters via a menu option on the canceller's hand-held terminal or on the network management system (NMS).

 Room for Improvement in the Handset 

Applying effective echo control via the echo-cancellation platform is one way of improving the overall call clarity on digital cellular networks. Another derives from improvements that must be made within the handset or terminal itself. There also is considerable room to enhance the network itself, focusing principally on vocoder development. 

Recent headlines have charted the ongoing commercial battles regarding which digital technologies will eventually emerge as the winners, as equipment manufacturers fight it out. However, this public battle will soon be overshadowed by another battle concerning handsets. At present, there are four major players in the digital cordless market. Europe has cordless telephony (CT2) and digital European cordless telephony (DECT), while Japan has the personal handyphone system (PHS) and the United States has personal communications services (PCS). 

Connecting directly into the plain old telephone system, CT2 was one of the first digital technologies to provide low-cost mobile phones. Although the technology worked well, it had a fundamental problem: it could not handle cell handovers. DECT and GSM have overcome this problem and will eventually dominate European cellular services. 

During the development of early cordless telephony, attention was paid to basic and enhanced functions and interworking with different network architectures. While the early generation of handsets looked very elegant and aesthetically pleasing, very little attention was paid to designing the handset with echo suppression/cancellation in mind. The result was that they looked good but were extremely poor at reducing acoustic echo. 

In the setting of standards for GSM and PCS, handset design and the impact of different design approaches on call quality was researched. As a result, recommendations stated a range of parameters, including sidetone tolerance and echo return loss performance. With the resultant advent of new recommendations with much tighter requirements for handsets, there is a call for greatly improved designs to be implemented. This, complemented by ongoing improvements in network technology and echo cancellation techniques, will bring digital wireless telephony much closer to matching wireline quality.

Echo Canceller Performance

The NLMS-based echo cancellers for both hybrid and acoustic echo canceling do exist and perform well, however, acoustic echo cancellation is more complex due to the specifics of the acoustic echo paths and the need for the acoustic echo cancellers to operate in the presence of noise in the echo path (examples: noise in the car, noise in a crowded room). For these reasons a number of enhancements has been proposed and implemented in the acoustic echo cancellers (AECs) by researchers.

There are certain improvements possible when employing a frequency-domain AEC. As you should have already realized, the NLMS algorithm presented earlier entirely operates in the time domain, no work is done there on the spectrum.

The first problem with time-domain AECs is their resource requirements, MIPs. Normally, the AECs need to have many filter coefficients to efficiently cancel the acoustic echo. But doing long convolutions to generate the replica of the echo signal is expensive when doing them directly in the time domain. It is possible to modify the initial NLMS algorithm so that the filter coefficients are updated once per a block of samples y(i)…y(i+N) instead of doing that each new sample y(i). The NLMS algorithm such modified is called the block NLMS (or BNLMS) algorithm. The advantage of keeping the coefficients fixed during the block of N samples y(i) is that it is possible to replace the time-domain convolution by multiplication in the frequency domain. The direct convolution computation in the time domain has a cost proportional to N2 (e.g. the number of multiplications, if we compute it for N samples and there are N coefficients in the filter). Frequency domain processing requires computation of several Discrete Fourier Transforms (DFTs) of the signals. The Fast Fourier Transform (FFT) is a very efficient DFT implementation and it is known to have a computational cost proportional to N*log2 (N). So it is more efficient to use convolution by FFT for big Ns. The disadvantage of using BNLMS is that it has slower convergence, which is due to the effective scaling of the adaptation step b down N times. Also, such and similar simple frequency-domain echo cancellers introduce a processing delay. Hence, we have a tradeoff between the convergence time, delay and cost. It should be noted, however, that FFTs require additional memory, so, we also trade memory for MIPs.

Another problem with AECs is that if their implementation is a time-domain NLMS, then they will perform poorly in the presence of noise (BNLMS will suffer from noise too). This problem is especially important for the AECs that need to work in the cars, crowded rooms or otherwise noisy locations. When implementing the AECs with frequency-domain analysis and synthesis blocks (or sub-band processing), it can be possible not only to reduce the computational cost, but also reduce the processing delay, have better noise immunity and even suppress the noise by frequency-domain noise suppressors directly integrated in such AECs. Doing so will greatly improve the quality of speech in the end. Also, with frequency-domain processing, it's possible to have better immunity to the nonlinearities in the echo paths because the AEC will adapt to the strong fundamental frequencies, while their weak harmonics can be suppressed as part of noise. This all is impossible to achieve with time-domain AECs.

Finally, it is important how the echo canceller behaves in the double-talk situations, when both talkers talk simultaneously. Obviously, the parties prefer to hear each other throughout the entire conversation and hear little or no echo during double-talks. Therefore, the echo canceller's double-talk performance should also be addressed when designing and testing echo cancellers or simply choosing which one to integrate to the phone.

Frequency-domain echo cancellers are very effective in canceling acoustic echoes. Unlike time-domain AECs, they need fewer DSP MIPs, perform better in double-talk situations, work well in presence of noise, can have embedded noise suppression almost for free and perform better with nonlinearities in the echo path. This is why frequency-domain echo cancellers should be preferred over time-domain ones.

Testing Echo Cancellers

It is a good practice for the customer to ask the echo canceller algorithm supplier how well their echo canceller conforms to the appropriate ITU-T recommendations (which are de-facto standards) and provide these figures alone with the resource requirements so a right decision can be made when choosing an echo canceller. The related ITU-T recommendations are: G.168 for LECs and G.167 for AECs.

It is beneficial for the customer to understand the basics of the echo cancellation and maybe even be familiar with the listed recommendations, however, it always makes sense to make a few tests of the echo canceller of interest. If a live test is possible, which is very desirable for AECs, it is good to make it. The echo canceller suppliers should provide a test or demo suit and a few test waveforms (the reference signal y(i) and the signal with the echo, x(i)+r(i) as per Figure 5), on which the echo canceller can be tested. Such a test can be carried out on either a PC or the customer's target hardware, whichever is arranged. This ensures the echo canceller operation and the suit can also be used to test the echo canceller performance on specific waveforms if the customer has any concerns about particular cases. It's also a good thing to test double-talk performance of the echo canceller to make sure the quality is delivered to the end users.

By the time the echo canceller integration is about to start, the hardware of the target device must have a sufficiently low level of nonlinear distortions. Only after having fixed all of the hardware problems, the echo canceller integration should begin. As soon as the echo canceller integration is finished, the echo canceller test can be repeated in full real-time with true I/O instead of file processing. Should there be any quality problems, the hardware and software must be checked against possible violations of the requirements imposed by the echo canceller, which have been stated earlier.

Applications

1) Echo Cancellation on a PC

Echo cancellation on a PC, equipped with a digital sound card, microphone and active speakers should be possible to achieve in principle, but it's not always feasible. It's an interesting topic of its own; so let's see what problems can arise when trying to get the echo cancellation work on a PC and how to cope with them.

The vast majority of users of PC multimedia hardware such as sound cards, microphones, speakers and amplifiers are PC gamers. Often, the digital sound cards are used only for output in games and music players. This unfortunate practice allows making low-quality hardware, which is perfectly suitable for the outlined applications, and it is cheap. To find out whether or not the hardware is low quality, it's possible to carry out a nonlinear distortions measurement as we suggested earlier. Replacing of the microphone or active speakers with better ones can improve the overall quality of the system.

One problem with sound I/O on the PC is that the input and output can be independent and may even have different clock sources so each one can have its own sample rate. This is a very undesirable feature, which leads to changes in the echo path delay. It's somewhat similar to the problem of the codec synchronization and solutions analogous to adaptive interpolators can be used.

The hardware isn't the only place, where the problems (like nonlinear distortions and others) can appear. Another place is the software, which controls this hardware, namely, the device drivers and operating systems. The drivers for digital sound cards can be incorrectly implemented in that they may lose the samples. The end-user desktop operating system running on the PC can make a considerable contribution to this problem as well. Often, the desktop operating systems and their software, which can't work as part of a real time system and meet certain deadlines, lose responsiveness to I/O and processing requests on heavily and even moderately loaded PCs. While this can be tolerated in many applications such as games or mp3 players (a few lost samples will go unnoticed!) and the like, the echo cancellers will simply fail to do what they're supposed to. Possible solutions to this problem include closing all CPU-intensive applications and services in the operating system, using real-time operating systems (if applicable) and upgrading to a faster PC.

All of the above issues make echo cancellation on PCs problematic because of not meeting the basic requirements imposed by the echo cancellers.

The Future of Echo-Cancellation Technology 

New digital cellular networks and network scenarios reflect a significant change taking place in the operation of echo cancellers. Instead of being a means of simple echo control, echo cancellers have now become highly sophisticated transmission equipment at the center of highly complex networks. Network operators and telcos implementing echo cancellation across their networks will hold the key to improved call quality, directly impacting their ability to provide enhanced network performance, maintain customer loyalty, increase talk time revenues, and reduce subscriber churn. 

Conclusion

As we have seen in the preceding sections of this article, there are many possible problems, which can arise when designing and implementing a system with an echo canceller. But there is no black art or any other magic behind the failures. The reasons for them are well known and perfectly consistent with the echo canceller internal organization and requirements. To prevent delays in the development and reduce the costs, consider designing the system to meet the requirements at the very beginning. Redesigning the whole system at the middle or last stage because of not meeting the requirements will be expensive.

Solid understanding of the basics of the echo cancellation and meeting the general requirements imposed by echo cancellers will avoid all of the echo canceller problems and therefore shorten the development time and product costs, which is always desirable.

The engineers at SPIRIT Corp hope that this little investment in the form of an article will make a good service to all parties interested in canceling echoes in their products.