Patent Publication Number: US-2020295849-A1

Title: High performance telemetry system with a combination of soft and hard decision decodnig

Description:
REFERENCE CITED 
     US PATENT DOCUMENTS 
     U.S. Pat. No. 4,642,800 February 1987 Umeda 
     U.S. Pat. No. 4,908,804 March 1990 Rorden 
     U.S. Pat. No. 7,577,528 August 2009 Li et al. 
     OTHER PUBLICAITONS 
     
         
         Jiang Li et al., “Efficient compression of borehole resistivity and acoustic imaging data using wavelets”, Geophysics, vol. 70, no. 6, November-December 2005, pp. F53-F60. 
         A. Oppenheim et al., “Discrete-time signal processing”, Prentice Hall, 1998. 
         J. Proakis, “Digital communications”, McGraw Hill, 2000. 
         G. Ungerboeck, “Trellis-coded modulation with redundant signal sets, Part II: State of the art”, IEEE Communication Magazine, vol. 25, no. 2, February 1987, pp. 12-21. 
         A. Viterbi, “Error bounds for convolutional codes and an asymptotically optimum decoding algorithm”, IEEE Transactions of Information Theory, vol. 13, no. 2, April 1967, pp. 260-269. 
       
    
    
    
     BACKGROUND OF THE INVENTION 
     The telemetry system used in the measurement while drilling (MWD) or logging while drilling (LWD) is essentially a digital communication system. The commonly used communication media include the drilling mud, subsurface formation, the drill pipe, and the cable. Correspondingly, a communication system is called a mud pulse telemetry (MPT) system when a pulser is designed and used to regularly change the pressure or the flow rate of the drilling mud to generate pulse trains or continuous sinusoid waves, into which the digital information acquired at the downhole can be mapped, and thus conveyed to the surface. A communicate system is called an electromagnetic telemetry (EMT) system when an electromagnetic wave is generated at the downhole, travels through the subsurface formation, and received on the surface. A communicate system is called a wired pipe telemetry (WPT) system when wires are wrapped around the drill pipe and the digital information is conveyed through the wires, which may require more complicated design and equipment. Other forms of telemetry system, such as one using the acoustic waves traveling along the drill pipes to convey the digital information, exist but may not be as popular as the MPT and EMT systems in terms of a relatively good performance and cost ratio. On the other hand, however, the low data rate is a major concern for the MPT and EMT systems. It is always a challenge to find a balance point between increasing the data rate and maintaining the system reliability and the decoding quality. This invention presents digital signal processing (DSP) solutions to such a challenge. 
     In general, an MWD/LWD telemetry system works as shown in  FIG. 1 . The system consists of two parts, the downhole system and the surface system. The downhole system implements the data acquisition, encoding, modulation and transmission, and the surface system is to receive the transmitted signals, from which to recover the downhole data by the inverse steps, namely demodulation and decoding. Specifically, the data acquisition module in the downhole system may include a variety of sensors to obtain the downhole information, including but not limited to (1) the bottom hole assembly (BHA) orientation, i.e., tool face; (2) the wellbore survey, such as measured depth, inclination and azimuth; (3) the drilling related information, such as downhole temperature, mud pressure, rotations per minute (RPM), weight on bit, torque on bit, shocks and vibrations; and (4) the formation related information, such as natural gamma ray, resistivity, density and porosity. The data acquisition module may also include an analog to digital (A/D) converter, to generate the digital information by sampling and digitizing the analog measurements from the sensors. Once the digital information is available, a series of DSP operations, such as data compression, data frame, channel coding and modulation, will play important roles to get the downhole information ready to be transmitted to the surface efficiently. The data compression is to remove the redundant information inherent in the measurements, and thus reduce the total amount of information that needs to be transmitted, essentially increasing the data rate in an indirect way. The data frame is an operation to pack the digital information into multiple frames with the additions of the frame identifications and the synchronization information for the purpose of convenience and robustness of the digital communications. The channel coding is to introduce some redundant information with known patterns, as opposed to the data compression, for the purpose of detecting and correcting errors during the digital communications. The modulation is to map the digital information into some high frequency carrier signals for the purpose of improving overall performance of the telemetry system. Once the modulated digital signals are ready, a digital to analog (D/A) converter in the transmitter module may be used to convert the digital information to analog signals that will then be used to drive a transmitter, such as a pulser, to send the downhole information to the surface. 
     The receiver module in the surface system may use certain types of receivers, such as sensors to detect changes of pressure or flow rate of the drilling mud, to receive the downhole information sent by the downhole system. Then, an analog to digital (A/D) converter may be used to generate the digital information by sampling and digitizing the received analog signals. Once the digital information is available, a series of DSP operations, such as demodulation, channel decoding, data de-frame and data de-compression, may be conducted to recover the original downhole information. These DSP operations are basically the inverse operations of those conducted in the downhole systems. Because of the inversible property of these DSP operations, the original downhole information can be recovered easily if the received signals are the same as the transmitted signals, which requires an ideal communication channel where neither noise nor distortion occurs. Unfortunately, however, the reality is that the hostile drilling environment introduces all types of noise and distortion during the communications, which makes the recovery of the original downhole information from the received signals become extremely challenging. This invention presents DSP solutions to such a challenge. 
     SUMMARY OF THE INVENTION 
     One embodiment of the invention is the DSP solutions to a high performance telemetry system of transmitting the downhole measurements and information to the surface with the high data rate, high system reliability and high decoding quality. The system consists of a downhole system and a surface system. The DSP solutions to the downhole system include the compact representation of the digitized downhole measurements and information; the packing of the digital information into data frames, with additions of information such as frame identification and synchronization; the introduction of redundant information with known patterns to the data frame; the modulation of the digital information onto high frequency carrier signals, with choices of various carrier frequencies and modulation techniques. The DSP solutions to the surface system include the demodulation of the digital information from the received carrier signals; the detection and correction of errors in the received digital information by making use of the transmitted redundant information with known patterns; the unpacking of the digital information from the data frame by making use of the information of synchronization and frame identification; the recovery of the original information from the compact representation of the downhole measurements and information. 
     Another embodiment of the invention is the DSP solutions to combat the noise and distortion in the received signals, introduced by the hostile drilling environment, in order to recover the downhole information with high decoding quality and to make the system work with high reliability. The DSP solutions include the noise suppression of the received carrier signals; the compensation of the signal distortion caused by non-ideal communication channel; the synchronization and timing between the transmitter and the receiver; the coherent and non-coherent demodulation; and the optimum estimation and decision making to recover the original downhole information with high confidence. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  shows a block diagram of a MWD/LWD telemetry system. 
         FIG. 2  shows a process of the modulation and demodulation with an ideal communication channel. 
         FIG. 3  shows a process of the modulation and demodulation with a non-ideal communication channel. 
         FIG. 4  shows a process of the channel equalization with the soft decision decoding. 
         FIG. 5  shows a process of the channel equalization with the hard decision decoding. 
         FIG. 6  shows a process of the channel equalization with a combination of the soft and hard decision decoding. 
     
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
     The MWD/LWD telemetry system, as shown in  FIG. 1 , is essentially a digital communication system. The system  100  consists of two parts, the downhole system  110  and the surface system  120 . The communication channel  130  connects the two parts, where the noise and distortions such as attenuation, delay, and multipath interference may be introduced to the signals transmitted from the downhole system  110  so that the signals received at the surface system  120  will be different from the transmitted signals. How to recover the transmitted signals and further the original downhole information from the received signals reliably and accurately, particularly at high data rate, becomes a primary challenge. The DSP methods provide solutions to such a challenge. In general, the DSP methods for the downhole system  110  includes the data compression  112 , the data frame  113 , the channel coding  114 , and the modulation  115 . The DSP methods for the surface system  120  includes the data de-compression  122 , the data de-frame  123 , the channel decoding  124 , and the demodulation  125 . 
     The data compression  112  in the downhole system  110  is to implement the compact representation of the downhole measurements from the data acquisition  111 . In general, the data compression is implemented in consecutive three steps. The first step is to remove the redundant information, which is essentially the correlations among measurements. For the same reason, this step is also referred to as the de-correlation. The methods for the de-correlation fall into two categories, the differential and the transformation. The differential method is to represent the original measurements in two parts, a predefined reference and the differences from the reference. If the correlations in the measurements are high, then the differences will be low, which means the original information can be represented in a compact way. The transformation follows the similar concept, which is to represent the original information in a different domain, where the information is compacted into a small group of transformation coefficients. The commonly used transformations include the cosine transform, a variant of the Fourier transform, and the wavelet transform (Li et al., 2005). The second step is called the quantization, which is a process to remove insignificant information while preserving the significant information. The consequences from the quantization lie in two folds. On one hand, it saves the bandwidth of the communication since only a fraction of information is transmitted. In many cases, it is acceptable because the information that is not transmitted is insignificant. On the other hand, there is a loss of information, although insignificant, for which case the data compression is called the lossy compression. In order to preserve all the information, the quantization will not be applied, for which case the data compression is called the lossless compression. The third step is called coding, which is essentially an efficient way to represent the de-correlated information. Two commonly used skills for the coding are the entropy coding and the bit-plane coding. The entropy coding, such as the Huffman coding and the arithmetic coding, makes use of the probability distribution information of symbol alphabets to encode different symbols with different-length codewords. Basically, short codewords are used for the symbols with high occurring frequency and long codewords for the symbols with low occurring frequency. As a result, on average with the probability as a weight, for the same amount of information smaller number of binary bits is needed to represent the same information. Since the length of the codewords are different for the different symbol, the entropy coding is often referred to as the variable-length coding. The bit-plane coding essentially incorporates the quantization into the coding, and basically preserves or transmits the most significant bit in a codeword first and the least significant bit last. The data compression represents the original information in a compact way and essentially increases the data rate in an indirect way. The data compression is an inversible operation, and thus the data de-compression  122  in the surface system  120  is simply an inverse step to recover the original information from the compact representations. 
     The data frame  113  in the downhole system  110  is to pack the original digital information from the data acquisition  111 , or the compact digital information from the data compression  112 , into multiple frames. The contents of the information to be packed and the way how the information to be packed is defined in the frame identifications. In addition, in order to make the surface system and the downhole system work on the same data frame, certain synchronization information is added to the data frame. The data de-frame  123  in the surface system  120  is simply to extract the contents in the data frame by using the synchronization and frame identification information. 
     The channel coding  114  in the downhole system  110  is to implement the error protection of the data frame  113 . The idea is to introduce certain redundant information with known patterns to the data frame. If there is any errors occurred during the communication, the channel decoding  124  in the surface system  120  is able to use the known patterns to detect and even to correct the errors during the decoding process. In general, there are two types of channel coding, the linear block coding and the convolutional coding. The commonly used parity check belongs to the linear block coding. Other examples of the linear block coding include but not limit to the Hamming code, the cyclic redundancy code, and the reed solomon code. The convolutional coding becomes popular because of an optimum and efficient decoding algorithm, namely the Viterbi algorithm (Viterbi, 1967). 
     The modulation  115  in the downhole system  110  is to map the digital information in the data frame  113  protected by the channel coding  114  into some high frequency carrier signals, which will be sent through the communication channel to the surface in an efficient way. The demodulation  125  in the surface system  120  is basically to recover the digital information from the received carrier signals. The demodulation  125  becomes complicated and challenging when the received carrier signals are contaminated by the channel noise and distortions. It is worth to mention that when the channel coding  114  and the modulation  115  are implemented jointly, the overall performance of the communication system may be improved. This process is known as the trellis coded modulation (Ungerboeck, 1982). 
       FIG. 2  shows how the modulation and demodulation works when the communication channel is an ideal channel with neither noise nor distortions. Mathematically, the process of modulation  202  is basically a product of a signal pulse p(t) bearing the digital information and a sinusoid signal with a given carrier frequency f c , i.e., 
         s ( t )= p ( t )·cos(2π f   c   t )
 
     where s(t) is called the modulated signal, which is also the signal that will be transmitted through the communication channel. It is known that the product operation in time domain is equivalent to the convolutional operation in the frequency domain. Therefore, the process of modulation  202  may be expressed in the frequency domain mathematically as, 
         S ( f )= P ( f )*½[δ( f−f   c )+δ( f+f   c )]
 
     where the notation * represents a convolution operation, S(f) and P(f) are the spectrum of s(t) and p(t), respectively, and δ(f) is a delta function. Note that this convolution operation is basically a translation process  201 , wherein the spectrum, P(f), of a signal pulse p(t), located at the zero frequency or in the base band, is translated to the high-frequency pass band, located at the carrier frequency f c  and its symmetry −f c . If ignoring the negative frequency, with a proper normalization, the spectrum, S(f), of the modulated signal s(t) is simply a translated and duplicated version of the spectrum, P(f), of the original signal, p(t), i.e., 
         S ( f )= P ( f−f   c ) 
     The advantages of using the modulation may include but not limited to avoiding large-size antenna, allowing bandwidth sharing, avoiding low-frequency noise, and achieving high data rate. 
     In the process of modulation  202 , there are basically three parameters that can be modulated, and they are the amplitude A, carrier frequency f c , and phase ϕ of the sinusoid carrier signal, in its typical form, A·cos(2πf c t+ϕ). When the amplitude is modulated, its corresponding digital modulation process is called an amplitude shift keying (ASK). When the carrier frequency is modulated, its corresponding digital modulation process is called a frequency shift keying (FSK). When the phase is modulated, its corresponding digital modulation process is called a phase shift keying (PSK). There are also many variants from these three basic modulation processes. 
     Mathematically, the process of demodulation  203  is basically a product of the received signal r(t) and a sinusoid signal with the same carrier frequency f c , followed by a lowpass filter. When the communication channel is ideal, the received signal r(t) is the same as the modulated and transmitted signal s(t), i.e., r(t)=s(t). Therefore, the demodulated signal before the lowpass filter is, 
         d ( t )= r ( t )·cos(2π f   c   t )= p ( t )·cos 2 (2 πf   c   t )= p ( t )·½[1+cos(4 πf   c   t )]
 
     The lowpass filter removes the frequency that is two times of the carrier frequency, and consequently the original signal pulse p(t) may be recovered from the signal d(t). 
     When the communication channel is not ideal, as shown in  FIG. 3 , the received signal v(t) is a contaminated version of the modulated signal s(t). If the communication channel is represented by its channel impulse response h(t) and the additive noise to the communication channel is represented by n(t), then the received signal v(t) can be represented as, 
         v ( t )= s ( t )* h ( t )+ n ( t ) 
     In order to recover the original signal pulse p(t), the modulated signal s(t) needs to be recovered from the contaminated receiving signal v(t). Obviously, the additive noise n(t) needs to be removed, which is known as a process of noise suppression  304 . Ideally, after the noise suppression, the contaminated receiving signal v(t) becomes, 
         w ( t )= v ( t )− n ( t )= s ( t )* h ( t )
 
     Then, the channel impulse response h(t) needs to be matched so that the modulated signal s(t) can be extracted from w(t). This process is known as the channel equalization  305 , which is expressed mathematically as, 
         r ( t )= s ( t )* h ( t )* c ( t ) 
     where c(t) is defined as the impulse response of a matched filter. It is known that the convolutional operation in time domain is equivalent to the product operation in the frequency domain, and thus the process of channel equalization  305  is expressed mathematically in the frequency domain as, 
         R ( f )= S ( f )· H ( f )· C ( f )
 
     Note that if H(f)·C(f)=1, then R(f)=S(f) , or equivalently r(t)=s(t) . That is, the modulated signal s(t) and thus the original signal pulse p(t) is recovered from the contaminated receiving signal v(t) after the noise suppression and the channel equalization. The condition H(f)·C(f)=1 for the channel equalization yields that the frequency response of the matched filter c(t) is the inverse of the frequency response of the communication channel h(t), i.e., 
         C ( f )= H   −1 ( f ) 
     In addition, when the communication channel is not ideal, the carrier frequency and phase used in the demodulation  303  may deviate from the carrier frequency and phase used in the modulation  302 , and thus the process of demodulation  303  becomes, 
         d ( t )= r ( t )·cos[2π( f   c   +Δf ) t+θ]=p ( t )·cos(2 πf   c   t )·cos[2π( f   c   +Δf ) t+θ] 
 
     In this case, the original signal pulse p(t) may not be recovered from d(t) by simply using a lowpass filter, as for the case of the ideal communication channel. Therefore, it is preferred that there is a process that can synchronize the carrier frequency and phase at the modulation and demodulation. 
     In short, the demodulation process becomes complicated and challenging when the communication channel is not ideal, as encountered in the MWD/LWD telemetry system. Three processes, namely the noise suppression, the channel equalization, and the synchronization, are in need to recover the original signals from the contaminated receiving signals. The performance of these three processes will eventually determine the overall performance of the telemetry system. 
     The noise in the telemetry system is generally treated as the additive noise, which may fall into three categories: wideband, narrowband and harmonic noise. The wide-band noise may be suppressed by classical lowpass, highpass and bandpass filters, with either linear-phase finite impulse response (FIR) implemented by the windowing methods, or infinite impulse response (IIR) implemented by the butterworth, Chebyshev or elliptic approximation methods (Oppenheim et al., 1998). The narrowband may be suppressed by bandstop or notch filters. The pump noise in the MPT telemetry system may be treated as a type of narrow-band harmonic noise. The methods to suppress such noise include but not limited to the method of making use of the pump stroke information, such as U.S. Pat. No. 4,642,800 and the method of using adaptive notch filtering, such as U.S. Pat. No. 7,577,528. 
     The synchronization for the telemetry system includes the carrier synchronization, which is usually implemented by the adaptive methods such as phase locked loop (PLL), and the symbol synchronization and the frame synchronization, which are usually implemented by the correlation or matched filtering of specially designed synchronization symbols (Proakis, 2000). 
     The channel equalization is to combat two types of channel distortions, namely the frequency selective distortion and the intersymbol inference (ISI). The frequency selective distortion may be compensated by finding the frequency response of a channel equalizer matched to the frequency response of the telemetry channel, as shown in  FIG. 3 . The frequency response of the telemetry channel may be obtained by sending a set of training signals, such as chirps, through the telemetry system, and the calculating the ratio of the frequency responses of the input to and output from the system. 
       FIG. 4  shows the adaptive channel equalization method to compensate for the ISI. The adaptive method is essentially an optimization problem, which is generally implemented in three steps: defining an objective function, optimizing the objective function, and testing the objective function. In the case of the channel equalization, the objective function  407  is defined as a function of the error between the output from the channel equalizer  405  and a desired signal  402 . The adaptive algorithm  408  is used to optimize the objective function. The channel equalizer  405  is essentially a FIR filter, represented by a finite number of coefficients g [k], and thus the channel equalization is essentially a filtering process, defined as a convolution of the input to the channel equalizer the FIR filter g [k]. Therefore, the adaptive algorithm  408  is to iteratively update the FIR filter coefficients g [k], based upon the input to the channel equalizer and the desired signal  402 , in order to optimize the objective function  407 . 
     One of the commonly used adaptive algorithm is the least mean square (LMS) algorithm, in which the objective function is defined as the mean squared error (MSE) between the output from the channel equalizer  405  and a desired signal  402 . The minimization of the objective function is implemented through a stochastic steepest-descent algorithm, where a gradient function is calculated by using the second-order statistics or moments of the input information, namely the covariance of the input to the channel equalizer  405  and the cross-correlation between the input and the desired signal  402 . The steepest-descent algorithm uses a single adaptive parameter, which makes the algorithm simple but also results in a slow convergence of the optimization process. Furthermore, the LMS algorithm uses an estimate of the gradient function for the simplicity. 
     Another commonly used adaptive algorithm is the recursive least squares (RLS) or Kalman algorithm, in which the objective function is defined as the weighed sum of the squared error between the output from the channel equalizer  405  and a desired signal  402 . Different from the LMS algorithm where an expected value of the squared error, i.e. MSE, or an ensemble average is minimized, the RLS algorithm is essentially to minimize a time average, i.e., the weighed sum of the squared error. As a result, the RLS algorithm uses multiple adaptive parameters, which leads to a complex algorithm but a fast convergence of the optimization process. 
     The channel equalization shown in  FIG. 4  assumes that the desired signal  402  is known, which is simply a delayed version  420  of the original signal  410 . This known desired signal  402  is typically referred to as a training sequence. The error sequence that is used to form the objective function  407  and further used in the adaptive algorithm  408  is simply the difference between the delayed version  420  of the original signal  410  and the demodulated value after the channel equalization  430 , i.e., 
     
       
      
       e [k]=m[k−τ]−z [k] 
      
     
     where the demodulated value after the channel equalization  430 , z[k], is defined as the soft decision, and thus the decision making  406  is called the soft decision making. The advantage of the using the training sequences for the channel equalization is obvious, but it comes with the cost of the communication bandwidth. 
     When the training sequences are not available, the channel equalization is typically referred to as the self-recovering or blind equalization, as shown in  FIG. 5 . The decision making  506  is to find an optimal approximation of the demodulated value after the channel equalization  530 . When the decision making  506  is defined as a quantization operation, the optimal approximation may be defined as the nearest element in the message alphabet, i.e., 
     
       
      
       n[k]=Q {z[k]} 
      
     
     As compared to the soft decision, this type of decision making is called the hard decision making and the optimal approximation, n[k], is called the hard decision. The decision making  506  may also be a sign operation. The error sequence using the hard decision for the channel equalization without the training sequences is defined as the difference between the hard decision  540  and the soft decision  530 , i.e., 
     
       
      
       e[k]=n[k]−z[k]=Q{z[k]}−z[k] 
      
     
     This type of channel equalization is also referred to as the decision directed or decision feedback equalization. 
       FIG. 6  shows the use of a combination of the soft decision and the hard decision for the channel equalization, where a set of training sequences, e.g., the synchronization sequence itself, is first used for the channel equalization with soft decision and then the blind channel equalization for the message sequences is implemented by using the hard decision. 
     The method of using the pulse shaping may also be used to overcome the ISI. In the previous discussion of the channel equalization, it is desired to have a channel equalization filter to match the channel filter, i.e., C (f)=H −1 (f), in order to recover the original signal pulse p (t), from which the message bearing in the binary bit stream may be recovered. In fact, as long as the signal pulse p(t) is sampled at the correct time so that the sampled value p[k]=p(kT) has no ISI, the message represented the signal pulse p(t) will be recovered. That is, it is not really the entire shape of the signal pulse p (t) but rather its value at the sampling instant that matters for the message recovery. In general, if the sampled value p [k] is zero at all integer multiples of sampling interval T but one, then this sampled value will not interfere with other sampled values at the sample instants kT, and the signal pulse can have any shape in between without causing any ISI. This condition is known as the Nyquist criterion, i.e., 
     
       
         
           
             
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     where c is a nonzero constant, k is an integer and τ is a time delay. Note that the commonly used pulse shapes such as rectangle, sinc, and raised cosine satisfy the Nyquist criterion. 
     In the context of the communication system, it is said that it is not really the signal pulse itself, but rather the combination of the signal pulse and the receiver, that satisfies the Nyquist criterion. In other words, it is desired to have a receiver filter, g [k], that matches the signal pulse so as to simultaneously overcome the ISI and maximize the signal-to-noise ratio at the receiver. Interestingly note that this receiver filter g [k] is the adaptive channel equalization filter  405 ,  505 , and  605 , with soft and hard decision decoding, as shown in  FIGS. 4, 5, and 6 , respectively. The combination of the signal pulse and the receiver filter is a convolution in time domain, or a product in frequency domain. The convolution of rectangle signal pulse with itself is a triangle signal, which satisfies the Nyquist criterion with the good timing, i.e., sampling at the middle peak of the triangle. The convolution of sinc signal pulse with itself is a sinc signal, and thus satisfies the Nyquist criterion. However, the convolution of the raised cosine signal pulse with itself does not satisfy the Nyquist criterion. Since the raised cosine signal pulse has some good properties, such as the bandwidth efficiency and the rapid die-away of the impulse response, it is good to have a signal pulse that preserves these properties and satisfies the Nyquist criterion as well. The square root raised cosine (SRRC) is such a signal pulse, which may be found by taking the inverse Fourier transform of the square root of the frequency response of the raised cosine signal pulse.