A. Field of the Invention
The present invention relates to a method and device for compensating for channel and echo distortion in a communication receiver. Specifically the invention relates to an equalization method and structure for efficiently eliminating near end echoes while also equalizing the receive data to compensate for channel distortion. The invention is particularly useful in DMT modulation as typically used in ADSL transceivers.
B. Description of the Related Art
1. Echo Cancellation
Full duplex data transmission over a single twisted pair of wires is the simultaneous transmission of data in both directions. The equipment at both ends transmits signals to the two wires and receives signals from the same wires through the use of a hybrid device. The hybrid is also commonly referred to as a four-wire to two-wire converter because it converts a four wire circuit consisting of a transmit pair of wires and a receive pair of wires into a single two wire circuit. When the signals transmitted by both end stations occupy the same frequency band, confusion may result when a receiver attempts to distinguish a signal received from the distant end from its own transmitted signal.
The simplest and most well known technique of eliminating such near-end echoes is the analog technique of impedance balancing the hybrid circuit. The transmit signal placed on the two-wire full-duplex circuit also appears as an input to the receive side of the hybrid. The exact nature of the local echo signal that appears on the input of the hybrid will depend upon the impedance of the two-wire full-duplex line. The hybrid therefore attempts to model the impedance on the full-duplex two-wire line in order to create a local replica of the transmitted signal that will appear at its input. The hybrid then subtracts this local replica from the received signal, which includes the echo, leaving only the signal that was transmitted by the distant end data transmitter.
Because the impedance model is not perfect, attenuated and distorted echoes are mixed with the received signal. Data driven adaptive techniques have been developed to address the problem and to provide improved echo cancellation. These include well-known fixed solutions such as least squares method, and the well-known adaptive gradient algorithms. A simple baseband model of one adaptive gradient technique is shown in FIG. 1. Transmit symbols a.sub.n are converted to an analog signal at D/A converter 110. The echo signal u.sub.t is a distorted version of the transmit symbols represented by the signal a.sub.t filtered by filter G(t) 120. Echo canceller 100 also receives the transmit sequence a.sub.n. The received signal x.sub.t is the sum of the echo u.sub.t, the signal r.sub.t, from the distant end, and noise w.sub.t, and is sampled in A/D converter 130. The echo canceller 100 utilizes a stochastic gradient algorithm, also known as an LMS algorithm, based on an N-tap adaptive linear transversal filter to generate an estimate u.sub.n of the echo signal u.sub.n, which is the component of x.sub.n contributed by the sampling of u.sub.t. The filter taps are updated using the following algorithm: ##EQU1##
where c.sub.n+1 is a vector with the updated filter taps, c.sub.n is a vector with the previous set of filter taps, a.sub.n is a vector containing the most recent data symbols, z.sub.n is the signal remaining after the echo estimate has been subtracted, and .alpha. is the adaptation gain.
These and other digital signal processing techniques provide large echo attenuation. Such techniques have only been made practical by the computational capabilities of modern microprocessors. Even so, many sophisticated techniques over-burden today's microprocessors. Generally speaking, the length of the echo cancellation filter is determined by the duration of the echo impulse response G(t), and it is well known that the complexity of the algorithms increases with the increase in length of the filter.
2. Channel Equalization
Similar fixed and adaptive filter techniques are used to remove the effects of distortion imposed by the transmission channel, i.e., the two-wire circuit and the accompanying analog electronics of the transmitter and receiver. Channel equalizers using the LMS algorithm are effective at removing inter-symbol interference (ISI) that results from the symbols being spread out into adjacent symbol periods by the channel impulse response. Initially, adaptive equalizers are trained by the transmission of a training sequence. The training sequence is known to both the transmitter and the receiver. This allows the equalizer in the receiver to adjust its filter coefficients to minimize an error criterion. Once trained, the adaptive equalizer uses data decisions to determine the error, relying on the assumption that data errors will be infrequent. The filter may be allowed to continually adjust itself based on the error, or remain fixed after training.
As with echo cancellation, the complexity of equalization algorithms increase with the length of the equalization filters, which is in turn determined by the duration of the channel impulse response.
3. Asymmetric Digital Subscriber Lines
Asymmetric Digital Subscriber Line (ADSL) is a communication system that operates over existing twisted-pair telephone lines between a central office and a residential or business location. It is generally a point-to-point connection between two dedicated devices, as opposed to multi-point, where numerous devices share the same physical medium.
ADSL supports bit transmission rates of up to approximately 6 Mbps in the downstream direction (to a subscriber device at the home), but only 640 Kbps in the upstream direction (to the service provider/central office). ADSL connections actually have three separate information channels: two data channels and a POTS channel. The first data channel is a high-speed downstream channel used to convey information to the subscriber. Its data rate is adaptable and ranges from 1.5 to 6.1 Mbps. The second data channel is a medium speed duplex channel providing bi-directional communication between the subscriber and the service provider/central office. Its rate is also adaptable and the rates range from 16 to 640 kbps. The third information channel is a POTS (Plain Old Telephone Service) channel. The POTS channel is typically not processed directly by the ADSL modems--the POTS channel operates in the standard POTS frequency range and is processed by standard POTS devices after being split from the ADSL signal.
The American National Standards Institute (ANSI) Standard T1.413, the contents of which are incorporated herein by reference, specifies an ADSL standard that is widely followed in the telecommunications industry. The ADSL standard specifies a modulation technique known as Discrete Multi-Tone modulation.
4. Discrete Multi-Tone Modulation
Discrete Multi-Tone (DMT) uses a large number of subcarriers spaced close together. Each subcarrier is modulated using a type of Quadrature Amplitude Modulation (QAM). Alternative types of modulation include Multiple Phase Shift Keying (MPSK), including BPSK and QPSK, and Differential Phase Shift Keying (DPSK). The data bits are mapped to a series of symbols in the I-Q complex plane, and each symbol is used to modulate the amplitude and phase of one of the multiple tones, or carriers. The symbols are used to specify the magnitude and phase of a subcarrier, where each subcarrier frequency corresponds to the center frequency of the "bin" associated with a Discrete Fourier Transform (DFT). The modulated time-domain signal corresponding to all of the subcarriers can then be generated in parallel by the use of well-known DFT algorithm called Inverse Fast Fourier Transforms (IFFT).
The symbol period is relatively long compared to single carrier systems because the bandwidth available to each carrier is restricted. However, a large number of symbols is transmitted simultaneously, one on each subcarrier. The number of discrete signal points that may be distinguished on a single carrier is a function of the noise level. Thus, the signal set, or constellation, of each subcarrier is determined based on the noise level within the relevant subcarrier frequency band.
Because the symbol time is relatively long and follows a guard band, intersymbol interference is a less severe problem than with single carrier, high symbol rate systems. Furthermore, because each carrier has a narrow bandwidth, the channel impulse response is relatively flat across each subcarrier frequency band. The DMT standard for ADSL, ANSI T1.413, specifies 256 subcaniers, each with a 4 kHz bandwidth. Each sub-carrier can be independently modulated from zero to a maximum of 15 bits/sec/Hz. This allows up to 60 kbps per tone. DMT transmission allows modulation and coding techniques to be employed independently for each of the sub-channels.
The sub-channels overlap spectrally, but as a consequence of the orthogonality of the transform, if the distortion in the channel is mild relative to the bandwidth of a sub-channel, the data in each sub-channel can be demodulated with a small amount of interference from the other sub-channels. For high-speed wide-band applications, it is common to use a cyclic-prefix at the beginning, or a periodic extension at the end of each symbol, in order to maintain orthogonality. Because of the periodic nature of the FFT, no discontinuity in the time-domain channel is generated between the symbol and the extension. It has been shown that if the channel impulse response is shorter than the length of the periodic extension, sub-channel isolation is achieved.