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The implementation of an acoustic-echo canceler for a speakerphone using multiple cascadable adaptive FIR filter IC's (Motorola's DSP56200) is described. Ideally, canceling acoustic echoes requires a delay line which provides a delay equal to the impulse response time of the echo. It is shown that it is feasible to use the 24 bit coefficient precision of the DSP56200 to reduce the length of the delay line. Simulation results show that with only 256 taps an Echo Return Loss Enhancement (ERLE) > 10 dB can be achieved on an echo lo00 samples in length. L IntroductionIn recent years then has been a growing interest in acoustic-echo cancellation for speakerphone, acousticconference, and video-conference applications. Of the many phenomena which affect the speech quality in such applications, (the manifestations of) acoustic echoes are dominant. One of the most important uses of adaptive signal processing is canceling echoes. In speakerphone applications there are two echoes which must be canceled -1) acoustic echoes due to multi-path room reverberations and, 2) line echoes due to impedance discontinuities in telephone lines. In the case of line echoes, which are particularly noticeable when making long distance calls, the theory has been well documented [1,2]. This paper discusses some aspects of implementing acoustic echo cancelers.The following section describes the acoustic-echo canceler model which was assumed for the simulations and suggests performance criteria. In section I11 a brief overview of the key points of the echo cancellation algorithm is presented.Section IV deals with the results of computer simulation m s . The hardware set-up which is used to verify the results of the simulation is described in section V. The correspondence between theory and experiment will be reported at the ICASPP-89 conference in May. IL Acoustic-Echo Canceler ModelThe assumed acoustic-echo canceler model is shown in Fig. 1. Typically, the user of a speakerphone adjusts the speaker gain, G I so that the far-end speaker sounds as loud as the near-end speaker hears himself. As a result the round trip gain in the system ends up being on the order of 0 dB. This, of course, is conducive to oscillation (howling) if the acoustic echo is significant. Depending on the characteristics of the acoustic chamber the acoustic echo may be sufficiently strong, such that loss must be i n d u c e d into the echo path. The amount of loss inaoduced is termed the Echo R e m Loss Enhancement (ERLE) defined as [3] : where €@(k)'] and €[e(&)'] are the expected values of microphone input signal power and uncanceled echo signal power, respectively, as shown in Fig. 1. A goal for ERLE is at least 10 dE3 with > 20 dB being preferable.Due to advances in CMOS process technology, inexpensive adaptive digital filters are readily available. The DSP56200 represents such a realization. This part, which is described in more detail in section V, offers 256 taps and 24bit coefficients. As is shown in the following section, ERLE is a function of many parameters inclu...

This paper describes a jointly adapting echo canceller and equalizer for a V.32 Modem. The coupled structure is analyzed and a software solution is resented for implementing the structure on a digifa! signal processor 8 r real time data throughput. The importance of thrs software solution in creating the complete modem function in software for the V.32 modem is discussed. IntroductionV.32 ( C m standard) modems are intended for use on two-Wire general switched telephone networks (0 at data signalling rates of up to 9600 bit per second @ps). The principal characteristics of the modem are defined in [l], the following being the most critical and computationally intensive blocks: channel separation by echo-cancellation techniques, channel equalization to compensate for intersymbol interferences due to h e a r distortion on the channel, encoding and decoding of the signal using quadrature amplitude modulation (QAM) for each channel with synchronous line transmission at 2400 baud.The simple and efficient implementation for the QAM block using a 32-state trellis encoding and Viterbi decoding algorithm has been introduced by the authors [l] and is well documented in [2]. The adap tive real-time algorithms for full-duplex voice transmission with electric and acoustic echo-cancellers have also been studied thoroughly by the authors [3,4]. However, it has been shown that the real performance of the V.32 modem is driven by the optimum channel equalization and exceptional echo-cancellation in excess of 60 dB under worst case conditions [5,6]. This paper will discuss the analysis, implementation and simulation of jointly adapting the echo cancellers and channel equalizers to achieve this performance.The. conventional approach performs echocancellation, transversal equalization and decision feedback equalization separately [7l as shown in Figure 1. However, this decoupled technique provides only a local minimum of the entire receiver, while any misjudgments in the Modulator 'r Transmitted slgnal I Error 2 . r Recelvedslgnals! I Equahr [ Figure 1. Conventional Decoupled Structureecho-canceller can deteriorate the equalizer performance. There are at least two recent ideas for improving the overall performance of the system, the first being to jointly optimize the adaptive echo-canceller, transversal equalizer and decision feedback equalizer [6]. The overall error output from the three different blocks may be used to update all three, which makes them a coupled system, as shown in Figure 2. The second idea is presented in [5]; and it is to separate the echo canceller into two blocks, the near end echo which is not affected by the cannel distortion and the far end echo which is affected by the same channel distortion which affects the transmitted data itself. By combining these two ideas as shown in Figure 2, the performance improvement of both can be realized. Received slgnal 1 Qunuvrl Figure 2. Coupled Structure ,It has also been shown that coupling the echo-canceller and the channel equalizer in the receiver decreases the resid...