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Description  |
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This invention relates generally to improvements in
signal encoding/decoding methods and apparatus and, more particularly, to a new and improved digital encoding and decoding system for lower distortion, higher resolution, and increased dynamic range reproduction of analog signals while remaining
compatible with industry standardized signal playback apparatus and standards not incorporating the decoding features of the present invention. In addition, recordings lacking the encoding process features of the invention are likewise compatible with
playback decoders which do embody the invention, and are provided some enhancement.
Quite often a recording or communications system is standardized and its format cannot be readily altered without affecting a substantial quantity of equipment already in existence. Hence, adding information with supplemental codes may not
always be practical unless provisions have been standardized for such insertions. Unfortunately, modern digital systems are not very expandable since data bandwidth, resolution, error correction, synchronization, ancillary data and other "housekeeping"
information essentially occupy the entire digital capacity of the storage or transmission medium.
However, electronic equipment manufacturers and users of such devices continue to seek enhanced performance and more features from such standardized systems. An important example is the need to make a compatible recording well suited
simultaneously for portable, automotive, television and audiophile markets. Today, many recordings are made for the most profitable market while other users suffer compromised sonics. The obvious conflicting performance requirements of different
listening environments and the need for sonic improvement should desirably be implemented by a new system which is compatible with older systems and recordings.
Automobile and portable equipment are usually low cost and must operate in noisy environments. Hence, in such situations, a slightly restricted dynamic range playback is beneficial. Audiophile systems require utmost accuracy, dynamic range, and
resolution beyond that which is available in the current standards. Thus, in any new compatible system, as provided by the present invention, encoded dynamics and slew rate modifications which achieve lowest distortion and best resolution for the
audiophile when decoded, should also provide improved sonics for portable and automotive playback when not decoded.
Compact Disc pulse code modulation and other digital audio encoding schemes are good examples of highly developed and standardized systems which push signal conditioning and digital information limits. Most such digital systems originally
evolved around then practical 2.5 to 3.5 mHz rotary head video recorder bandwidths. In such standards, the data bits with error correction and housekeeping entirely fill the available bandwidth. Accordingly, the need for a "smart" optimization
technique, which does not rely upon increased bandwidth for its implementation, becomes apparent.
By way of background, let us consider a typical digital audio record-play system, its most frequently encountered components, operation, and difficulties. In its simplest form, the recorder includes a sampling switch and an analog to digital
converter. The switch breaks the continuous analog signal into a series of voltage steps, each of which is converted to number groups or digital words. Digital level meters and simple communication systems often operate with just these functions in a
single IC chip. Practical high performance record and playback systems require many added operations to prevent undesired internal and external analog-digital signal interactions, as well as beats and non-linear feed-through between digital and analog
frequencies. Well-known technologies to deal with these problems include sharp cut-off or "brick wall" low-pass filters, fast sample and hold circuits, and high common mode rejection amplifiers. Unfortunately, although these components and subsystems
solve many problems, they also create others.
Briefly, in typical digital recording systems, low-pass filters ring, and if of analog construction, have pre-echo, are subject to sudden phase shifts near band edge, and have capacitors which often cause troublesome dielectric hysteresis
effects. Sample and hold circuits have unpredictable timing and capture errors for different signal slew rates and also suffer from capacitor problems. Fast digital signals and the high speed amplifiers needed to handle them often create and are
sensitive to ground currents which can cause audible strobe-beat effects. Digital reproducing systems have similar problems, along with spike or glitch generation caused by digital to analog conversion, and digital filter word length round off problems. Usually the recorder is designed to have state of the art performance while that of the reproducer degrades depending on the economies of "consumer" construction. These and other problems continue to plague modern high performance digital audio systems.
Unfortunately, such technical difficulties usually create jarring non-harmonic distortions, typically centered in the most sensitive and perceptive human hearing range. Often these distortions are caused by the highest, almost inaudible
frequencies contained within the program material. Taking the ratio of high and low frequency hearing acuity into account, and the fact that sounds unrelated to the program material stand out, the presence of even an extraordinarily small amount of
these distortions can be quite objectionable to the listener. Fortunately, often only very small corrections are needed to minimize some of these distortions. However, left as is, these distortion errors can combine to yield the equivalent of 13 to 14
bit performance accuracy from systems originally designed for 16 bit resolution. In practice, while some feel the advantages of current digital recordings outweigh the disadvantages of their distortion errors, many sophisticated listeners and
audiophiles are not so tolerant.
Accordingly, those concerned with the development and use of digital signal encoding and decoding systems for analog signals have long recognized the need for a higher quality, lower distortion digital system for reproduction of such analog
signals, which for all practical purposes is also compatible with existing equipment standards. The present invention fulfills all of these needs.
SUMMARY OF THE INVENTION
Briefly, and in general terms, the present invention provides new and improved digital encoding/decoding methods and apparatus for ultra low distortion reproduction of analog signals which are also compatible with industry standardized signal
playback apparatus not incorporating the decoding features of the present invention. In addition, signals lacking the encoding process features of the invention are likewise compatible with playback decoders which do embody the invention, and are
provided some overall enhancement.
Basically, the present invention is directed to various aspects of an improved encode/decode system for providing a predetermined balance or interplay of gain structures, filter characteristics, various slew rate modifications, and wave synthesis
operations to reduce signal distortions and improve apparent resolution. During the encoding process, an analysis of the signal to be encoded is made over time and the results of this analysis are subsequently utilized in the encoding and decoding
process to more accurately reconstruct the original waveform upon playback. This is accomplished while minimizing the deleterious effects normally encountered in sampling and converting analog signals to digital signals and subsequently reconverting the
digital signals back to an accurate simulation of the original analog waveform.
In accordance with the invention, control information developed during the aforedescribed waveform analysis is concealed within a standard digital code and this information is subsequently used to dynamically change and control the reproduction
process for best performance. These concealed control codes trigger appropriate decoding signal reconstruction compensation complementing the encoding process selected as a result of the aforementioned signal analysis. Since the control code is silent
and the overall digital information rate is normally fixed, the process can operate compatibly with existing equipment and industry standards. In addition, and as previously indicated, signals lacking the encoding process features of the invention are
likewise compatible with playback decoders which do embody the invention, to provide some beneficial enhancement.
To achieve higher performance with a fixed information rate, an on-going trade-off is made between dynamic range, to achieve improved small signal resolution, and peak level and/or slew rate, to achieve fast signal response accuracy. These small
change and fast change aspects of a signal, as well as large and small amplitude aspects, each have their own digital distortion or system compromise mechanisms. Since both large and small aspects will not occur at the same time, an optimum encoding
process or mix of processes favoring each signal condition can be chosen dynamically, in accordance with the invention, to achieve an improved signal reproduction within a fixed digital information rate. A silent or hidden control code documents these
changes from time to time in the signal encoding process and is used to create the complementary level, slew rate, filter character, and waveform synthesis necessary to restore the original signal during the decoding process.
In a presently preferred embodiment of the invention, the encoder system has much higher resolution and speed than the industry standard or encoded product, and is set up as an acquisition system with sufficient look forward and look behind
memory to compute the optimum processing of the signal and its corresponding reconstruction control code. As previously noted, the processing of the signal is determined based on a consideration of which trade-offs of resolution, speed, and level are
most appropriate for the signal conditions over time and how the reproducer can best be programmed to allow the most accurate reproduction of the original analog signal.
To be inaudible, the computed reconstruction control signal is encoded or encrypted to a random number sequence which may be inserted continuously or dynamically when needed into the least significant digital bit or bits. The processed audio or
signal becomes encoded to the remaining bits.
Conventional decoding by a simple digital to analog converter of all bits of a recording encoded in accordance with the invention, yields a signal with slightly less dynamic range and only slightly higher background noise. However, the signal
will have lower quantization and slew induced distortions and, hence, the processed encoded product, when reproduced on non-decoding standard equipment, will sound equal to or better than an unencoded product.
A fully decoding player, in accordance with the invention, retrieves the control signal and uses it to set up, operate and dynamically change a complementary process to recover the pre-computed high accuracy information and provide low distortion
reproduction of the original analog signal. Operations to do this include fast peak expansion, averaged low level gain reductions, selecting complementary interpolation filters, waveform synthesis, and others. When these are selected according to
ongoing trade-offs, optimum for a particular set of signal conditions, an apparent increase of bandwidth and resolution occurs.
An improved digital system, in accordance with the invention, uses groups of dynamically changing pre-determined performance trade-offs made when signal conditions of the recorded program would create distortion. Since digital distortions occur
at extremes of high level, slew rate, and high frequencies, on one hand, and with quiet signals and short small transients on the other hand, a best encode/decode strategy is chosen for that extreme without the process compromise hurting the opposite
aspects of the program. To achieve this, the program is delayed long enough so that a most likely distortion mechanism is identified prior to its emergence from the time delay, thereby allowing a best encoding strategy and complementary decoding method
to be determined and encoded. Performance is improved because any distortion compromise made occurs for opposite signal conditions, which are essentially nonexistent at that time.
In the simplest form of the system, an encoded dynamic range compression and complementary reproduce expansion will improve performance. Furthermore, improvements are had by using averaged levels of small signals independent of their lower
frequency and near supersonic frequency spectral components to control processing providing improved complementary restored resolution. In a similar manner, the strongest signals receive processing having DC to maximum bandwidth for instantaneous peak
conditions, which also yields best complementary restoration. Only one correction need operate at a time and, hence, digital information is saved, or conversely, more apparent performance is obtained from an unchanged digital information rate.
In addition, a further reduction of known and predictable digital distortions occurs by selecting a best low pass filter with the least compromise for program conditions during encoding and using a complementary interpolation or low pass filter
during reproduction. Also, other improvements are had from the reduction of known recurrent distortions, such as transient errors, by synthesizing these components from lookup table curves of these distortions or missing information, and then scaling
these to the signal at hand.
All of the aforedescribed improvements can also operate with varying degrees of success in a default or "open loop" mode at the reproducer by detecting information about the encoded signal and then varying these processes from the detected
signal.
Digital systems typically have a very high signal to noise ratio, but have a restricted working dynamic range of levels and restricted frequency response. The improved system of the present invention reduces distortions and, as such, uses signal
character dependent gain changes, filter optimization, slew rate processing, and waveform reconstruction or synthesis to do this. The improved system computes, within memory and process time limits, a continuously changing best compromise strategy of
available processes to give the best signal reconstruction. This obviously complex task yields a restoration control signal silently encrypted or noise disguised in a least significant bit code. By comparison, the reproducer system is simple, since its
decoding and complementary signal restoration can occur with conventional multiplying converters, digital signal processors and other analog and digital devices similar to or already used in consumer electronics.
A conventional recording and reproducing digital system appears relatively simple and potentially accurate for all the data bits encoded. In practice, however, using a very near to theoretical minimum sampling rate and the least acceptable
number of data bits substantially aggravates speed and accuracy limitations from even the best state-of-the-art circuits and components. In this regard, the worst offenders are items such as filters, sample and hold circuits, analog-to-digital
converters, digital-to-analog converters, and system grounding, timing and various process interactions and crosstalk.
The aforedescribed practical technological difficulties and their potential distortions can be greatly minimized by using higher sampling rates and more data bits than current standards allow. In fact, current technological capability permits
the reduction of cross-talk, time jitter and other noise interaction problems which, along with digital bandwidth limitations, prevented the practical implementation of higher data rates when current digital standards were first envisioned and
established. With today's high speed converters operating much faster with more data bits, filters can become less severe and the greater difference between highest audio frequencies and the digital sampling rate then reduces beats, sideband foldovers,
aliasing, as well as loss of small signal information. The present invention uses these capabilities by employing a high speed conversion process. The digital information rate, though now much higher, can be computed, as an ongoing acquisition process,
to an "error free" mathematically filtered lower sampling rate 16 bit code compatible with current standards. Most decimation oversampling encoders work like this. However, in addition, the invention anticipates alias, aperture, interpolation and
amplitude resolution distortions from an "ideal" standard reproducer and computes them during the encoding process for correction during reproduction. When the full process of the invention is used, even certain frequencies above the audio range or
Nyquist limit of industry standard equipment can be sent through the system without creating sub-harmonic or foldover distortions. Hence, a closer to perfect record/playback system is provided with minimal problems from filters, converters, and other
components or subsystems while remaining compatible with industry standards.
For a Compact Disc system, "perfect" reproduction to 16 bit industry standards will have a maximum of 65,536 well defined equally spaced resolution steps, each about 150 microvolts in amplitude when scaled to normal professional audio levels (10
volts peak-to-peak maximum). This number, when stepped consecutively at the industry standard 44.1 kHz sampling rate, provides a slew of less than 7 volts per second. Faster rates will skip numbers until, for a 10 kHz triangle segment, only 2.2 sample
points remain to define that waveshape as it would be filtered to its 20 kHz bandwidth. In this regard, more than a 1 giga Hertz sample rate would be required to include all 65,536 resolution points to create that wave segment. Fortunately, an ideal
interpolation filter will fill in all of these points provided the 2.2 samples have been timed accurately enough. To do this to achieve a one half bit RMS averaged accurate sample of a fast changing signal the sample timing must occur within: ##EQU1##
This sample, accurate in time and amplitude, must be held long enough for conversion to digital code. Usually, a charge on a capacitor represents this information. However, most dielectrics and insulators used to fabricate capacitors have
complex losses as well as past history memory which create a complex delayed voltage change, field re-distribution errors and leakage. When abrupt changes in level from sample to sample occur, as they do with sampled high frequency audio signals, these
errors often become much greater than when signal levels don't change. To have less than a half LSB of RMS averaged error the hold accuracy becomes: ##EQU2## or about 2.3 u Volt per u sec.
Such performance is well beyond simple applications of most modern electrical passive components, much less integrated circuits. Obviously, practical consumer playback equipment will not do better, and the resulting errors can produce slew rate
related transient intermodulation distortion components, which are among the most audibly objectionable. Specifically, these result from acquisition time uncertainty or jitter, slew rate related non-linear switching offsets, various types of dielectric
hysteresis causing previous event related errors, polarity dependent sample discrepancies, and unpredictable hysteresis within converters as well as other factors. Thus, practical systems often have complex signal related errors as high as twenty times
more than theoretical resolution limits of the current 16 bit standard. Hence, a process providing more sample points per second with the least voltage change per sample will yield a signal with lower transient intermodulation distortion.
A second distortion mechanism occurs with very small signal amplitude changes of about 5 to 20 millivolts represented by digital activities of less than about 8 bits in a typical 16 bit system. These levels seldom occur by themselves yet can
still be a small but audible part of a larger low frequency dominated signal. Hence, these small signals can occur averaged at many different voltage levels or digital numbers of a larger slow waveform. A practical example of this would be midband hall
reverberation decay and bass sounds combined. The reverberation signal attenuates and sometimes completely disappears as it becomes chopped or broken segment parts of the bass waveform. As previously indicated, these breaks represent the 150 uV
resolution limits of a "perfect" 16 bit reproducer. In practice, very small signal changes can become stepped outputs, or more often distort to irregular step to step changes with an uncertainty or hysteresis which occurs due to errors within converters
and from external interference and crosstalk. This produces a collapse of the sense of space in a recording and generates impulsive grainy noise effects which are usually made less objectionable by adding a random noise voltage to the signal prior to
encoding so that the step errors become randomized from the uncertain samples created. Thus, the stepped or quantized distortion becomes a less objectionable noise modulation and the least bit signal cut-off levels are now smoothed to a gradual gain
loss with progressively smaller signal changes. A better form of distortion reduction occurs by increasing the sample points per unit voltage change. Unfortunately, like the process to increase slew accuracy, a much higher digital information rate than
that of the current standard is needed to accomplish this.
Low signal level digital errors produce distortions such as quantization noise and resolution loss. Whereas, high signal level high frequency and slew rate related errors produce distortions such as sporadic beats and fast signal change envelope
related subharmonics, referred to as transient intermodulation distortion or TIM. One is easily misled by test signals with a continuous envelope nature, in that they tend to average over time and cancel many of these distortions and therefore
incorrectly indicate only very small resolution and converter inaccuracy distortions. Unfortunately, waveforms like those in music continually change and, as noted, may provide much higher and far more objecti onable non-ha rmonic TIM and resolution
problems.
Digital distortions occur with high slew rate and small amplitude signal change con ditions and, as previously indicated, both are not likely to occur at the same time. Hence, in accordance with the invention, the system identifies either a fast
slew or a small change character of the signal waveform and implements the appropriate corrective process. During encoding, the nature of program signal changes can then determine which corrective process is used as well as a best reproduce conjugate or
process at any time during decoding. One process can borrow information rate from a less needed performance capability when potentially severe distortion conditions in the sign al call for it. In this manner, a decision to provide more points per fast
voltage change yields an equivalent higher sampling rate at the expense of less important low level resolution. Conversely, a smaller voltage change per sample automatically reduces the momentarily unneeded speed capability. Such interplay and
compromise can be managed and/or computed to maintain a substantially constant digital information rate. Under these circumstances, the processed, decoded, analog output may have an apparent increase of bandwidth and resolution and, as noted earlier,
when these improvements occur, one or the other as needed, the fundamental causes of digital distortions as well as their effect on imperfect reproducers can be reduced.
A similar correction strategy is applied to reduce filter trade-off compromise errors between transient response, phase accuracy, settling time, group delay, and other distortions inherent with filtering methods. Such errors may not be
non-linear, and hence, will not appear as harmonic distortion; however human hearing is sensitive to manipulations of waveform shape and to the settling time of complex signals. Typically, the smallest amplitude high frequency signals are likely to have
excessive transient ringing and process noises from aggressive filtering, whereas sub-harmonic beats and other filtering noises may occur with intense high frequency signals. The instantaneous versus non-instantaneous character of complex signals is
reproduced differently from one filter type to another. As before, the same large signal/small signal selection criteria hold, allowing a best encode and decode filter choice, without having to compromise for the opposite, essentially non-coexistent,
program conditions.
Hence, the method and apparatus of the present invention utilize a pre-calculated optimal interplay of gain, slew, filter selection, and waveform synthesis operations done individually or as a composite all inclusive process which becomes encoded
and decoded in a complementary manner to reduce distortions and improve resolution. Included in such a system is a record compress-play expand system with some features similar in ways to those used in noise reduction systems. Most such noise reduction
systems use either peak or RMS detectors to examine the incoming signal and convert its level to either fast or slowly changing internal DC control signals which ultimately drive a transient free switching element or an analog variable gain device. When
set up for gain reduction, with increased input signal level, the output signal is compressed so that tiny signals are amplified and strong distortion prone signals are attenuated. Upon playback or decoding, a similar circuit set up for gain expansion,
detects level changes and restores the signal to an approximation of its original dynamics.
In contrast to traditional noise reduction, the system of the present invention corrects distortion. It does this by altering gain structure, as well as amplitude and slew rate linearity, for extreme low and high level signal conditions. Low
level, small changing parts of the signal are detected and used to control the gain of the whole signal which then includes more encoded bits. This gain control is derived from a broad middle spectrum of the signal and is active at signal levels
representing the lowest levels perceived by human hearing. It is not activated by low frequencies, near supersonic frequencies, or when higher level mid-band signals are present. In this manner, the gain structure increase maintains a minimum LSB
dither-like activity independent of inaudible sounds and maintains ambient and background information as well as masking quantization and monotonistic error distortions previously described.
Infrequent peak levels are instantly compressed with a transfer function having very low distortion for signals near maximum level and producing minimum upper harmonics once the limit threshold is traversed. This type of operation does create an
occasional higher distortion on peaks, however it prevents catastrophic overloading during recording and allows a higher recording level with overall lower distortion.
Infrequent fast slew portions of the waveform can be expanded symmetrically in time, and/or in samples, to encompass more encoded bits, and, as before, other parts of the waveform may be unaltered. This operation may be a dispersion process
where time delay is altered, or it can be a graphical waveform synthesis. It takes an instantaneous event and spreads it in time, and like the peak limiter, it creates distortion in undecoded playback.
Gain change, peak limit, and slew rate compression operations and their complements or restorative operations are practical with analog or digital techniques. Voltage controlled amplifiers, diodes, delay lines, and chirp filters, and multipliers
are typical analog building blocks which can be assembled to create these functions. Equivalent digital sub-routines and dedicated process algorithms and components are also available. Distortion free digital processing is complex; for example,
rounding off errors may have to be dithered and interpolated over time. However, once implemented, digital operations are very stable and precise compared to the variables subject to tolerances and adjustments required for the analog control of gain,
dispersion, bandwidth and time constants.
The aforedescribed level and slew processes of the present invention correct distortions occurring from opposite signal conditions which are not likely to occur at the same time. Hence, these can interplay and at maximum correction capacity can
borrow from an opposite less needed performance capability to maintain constant digital information rates. The wave synthesis process of the present invention operates with known distortion waveshapes which, when encountered during encoding, are
subsequently called out of memory by code for complementary correction during reproduction.
Level and slew correction works for known signal conditions having unpredictable distortions and synthesis works for known distortions occurring from signal conditions unpredictable at the reproducer. Unlike even state-of-the-art noise reduction
processes, this system's processing is under intelligent control and given sufficient computation, trial and error, or successive approximation time, the best correction scheme and its encoding for reproducer process control is readily determined and
optimized.
Wave synthesis, in accordance with the invention, is a keyed operation used to recall from memory a number of predictable and/or recurrent distortions known to occur at the reproducer. Small waveform segments falling outside of the Nyquist
sampling limits, repeated quantization distortions, and interpolation filter parameters can be recalled from a look-up table in memory or synthesized from information sent in the hidden code, and used for improved playback. The synthesis memory can
carry several interpolation waveshapes which best connect points at and between samples. These larger waveforms will maintain their characteristic shape independent of level, just as the reproduced signal would do. Once the connecting waveshape has
been recalled from ROM, it must be scaled to fit the signal. Since only very slowly changing waveforms will have samples without bit resolution levels in between, a form of level detection is necessary to make synthesized segments scaled to the signal.
What would have been level detectors and gain controlled devices in an analog system are replaced by equivalent digital signal processing functions in a digital system. Once this has been accomplished, the reconstructed waveform has more equivalent data
points in time and level and, when pre-computed properly, a lower distortion results from the curve fitting.
In light of the foregoing, a practical system, in accordance with the invention, may have many times better signal resolution and much better fast transient signal accuracy. A much greater digital information rate would normally be necessary to
achieve these results. Data is saved by processing only distortion producing conditions. As noted, resolution is selectively and adaptively traded off for slew accuracy and slew rate or maximum level is borrowed for higher resolution. Information rate
is conserved by toggling back and forth or fading from process to process when needed.
It should also be apparent that implementation of various subsystem designs may be in either analog or digital form, monitoring and analysis of the waveform may be accomplished at varying locations in the system including the reproducer and in
either analog or digital form, other parameters of the waveforms may be selected for compensation, and control codes or other waveform corrective message information may be inserted and extracted in a variety of different ways, without departing from the
basic concepts of the present invention.
Hence, the method and apparatus of the present invention for encoding/decoding signals with minimal distortion satisfies a long lasting need for a compatible system which provides an adaptive interplay of gain, slew rate, filter action and wave
synthesis processes to substantially reduce signal distortions and improve apparent resolution.
The above and other objects and advantages of the invention will become apparent from the following more detailed description, when taken in conjunction with the accompanying drawings of illustrative embodiments.
DESCRIPTION OF THE
DRAWINGS
FIG. 1 is an overall block diagram of an analog to digital encoding system in accordance with the invention;
FIG. 2 is an overall block diagram of a digital to analog decoding and reproducing system in accordance with the invention;
FIG. 3 is an more detailed block diagram of an example of an analog to digital encoding system in accordance with the invention;
FIG. 4 is an more detailed block diagram of an example of a digital to analog decoding and reproducing system in accordance with the invention;
FIGS. 5a through 5e graphically depict waveforms illustrating sampling and encoding errors encountered with low level and rapidly changing waveforms;
FIGS. 6a through 6f graphically depict various signal waveforms during the limiting and reconstruction of a triangle wave, in one embodiment of the invention;
FIGS. 7a through 7d graphically depict waveforms illustrating various types of distortion encountered with different types of filters;
FIG. 8 is a block diagram of a processing system in accordance with the invention, using analog processing technology;
FIG. 9 is a block diagram illustrating filter selection control in one embodiment of the invention;
FIG. 10 is a block diagram of a process switcher utilized in one embodiment of the invention;
FIGS. 11 and 12 graphically depict waveforms and distortion plots illustrating system response before and after process for two filter types, in accordance with the invention;
FIG. 13 is a block diagram of an analog implementation of a slew rate compression and expansion system;
FIGS. 14a through 14e show waveforms illustrating the operation of a slew rate compression and expansion system;
FIG. 15 is a block diagram of a more advanced, presently preferred digital embodiment of the encode system, in accordance with the invention; and
FIG. 16 is a block diagram of a more advanced, presently preferred digital embodiment of the decode system, in accordance with the invention.
DESCRIPTION OF THE PREFERRED EMBODIMENTS
The present invention is directed to a system for an electronic method and apparatus for signal encoding and decoding to provide ultra low distortion reproduction of analog signals, while remaining compatible with industry standardized signal
playback apparatus not necessarily incorporating the decoding features of the invention. The improved system provides a selective interplay of gain, filter selection, slew rate and wave synthesis operations to reduce signal distortions and improve
apparent resolution from a recorded product, under the control of concealed or silent control codes when necessary for triggering appropriate decoding signal reconstruction compensation based upon a previous signal waveform analysis made during the
encoding process for the recorded product. In addition, signals lacking the encoding process features of the invention are likewise compatible with playback decoders which do embody the invention, and are provided the benefits of some overall
enhancement based upon a signal waveform analysis made during playback.
Referring now to the drawings, and more particularly to FIG. 1 thereof, there is shown, in general terms, the analog to digital conversion and encoding subsystem of a typical recording system embodying features of the present invention.
As shown in FIG. 1, an analog signal 99 is directed as input to a processing subsystem 100 which converts the analog signal into digital form, including such tasks as filtering, sample and hold, analog to digital conversion and the like. The
digital output 100a from the subsystem 100 is directed to two subsystems, a memory subsystem 101 and an analysis and computation subsystem 102. In memory subsystem 101, the digital signal is delayed or stored for further use and manipulation. The
digital signal output of memory subsystem 101 is sent to subsystem 102 at input 102E. Using the output of subsystem 100 at input 102A, the waveform analysis and correction computation subsystem 102 continuously monitors and evaluates the digital format
waveform as it is being stored in the memory subsystem 101 in order to determine the physical characteristics of the stored waveform ultimately to be reconstructed and the required corrections necessary for accurate reconstruction and restoration of the
original analog waveform 99. This evaluation relates to reconstructive level, slew, and waveform synthesis requirements ultimately to be provided by complementary compensation in an appropriate decoding and signal reproduction system (FIG. 2). The
evaluation may also predict alias components for subsequent conjugate neutralization. Some aspects of the signal evaluation may be performed on the analog signal by subsystem 100, and the results sent to subsystem 102 at input 102d.
The corrective procedures are applied to the digital signal from the memory subsystem 101 by subsystem 102 under the control of signals resulting from the analysis. The process controller 102 also generates control codes for use by the decoder
which are converted to proper format and appropriately encrypted into the digital signal so that the control codes can silently ride along with the digital representation of the original analog waveform 99 and be provided as an encoded digital output
103. Some of these corrective procedures will relate, not just to distortion characteristics occurring as a result of the basic conversion of the analog waveform itself, but also to procedures deliberately introduced by the encoder for subsequent
complementary decoding, such as peak limit/subsequently expand for high level signals and averaged compress/subsequently expand for low level signals.
As best observed in FIG. 2, there is shown, again in general terms to illustrate some of the basic overall concepts embodied in the present invention, a digital to analog conversion and decoding subsystem of a typical reproducing system embodying
various features of the present invention for reconstructing the original analog waveform.
In FIG. 2, the encoded digital signal 103, recaptured from any appropriate recording medium (not shown) such as tape or disc, is directed as input to a digital signal analysis and processing subsystem 104 and to memory subsystem 107, which delays
the digital signal. Signal analysis subsystem 104 extracts control code information inserted in the signal at the encoder and may also analyze the signal itself to determine its characteristics. These operations include appropriate means for control
code detection, signal filtering, level detection, spectral analysis and the like. The detected control codes and signal analysis in the processing subsystem 104 are used to generate control signals directed to a reconstruction compensation subsystem
105 which interacts with the processing subsystem 104 and operates on the delayed digital input signal 108. Subsystem 105 includes digital to analog conversion, and may include further memory, such as one or more ROM's or look-up tables, for various
types of reconstruction compensation used, in accordance with the invention, to correct the digital signal 103.
The compensation subsystem 105 typically will respond to the various control codes, or the absence thereof, to generate a variety of corrective compensations such as slew rate, level, filter selection, and waveform synthesis which, through
appropriate interaction with the processing subsystem 104, yields a reconstructed analog signal 106 with minimal distortion and enhanced apparent resolution, all without the need for increasing industry standardized digital bandwidth.
It will be appreciated by those of ordinary skill in the art that the systems of FIGS. 1 and 2 are merely illustrative of simplified general approaches for practicing certain basic aspects of the present invention, and implementation of the
systems of FIGS. 1 and 2 may take a wide variety of specific forms without in any way departing from the spirit and scope of the invention.
It should also be apparent that implementation of various subsystem designs may be in either analog or digital form, monitoring and analysis of the waveform may be accomplished at varying locations in the system and in either analog or digital
form, other parameters of the waveforms may be selected for compensation, and control codes or other waveform corrective message information may be inserted and extracted in a variety of different ways, without departing from the basic concepts of the
present invention.
By way of example, one possible implementation of the general structure above is presented in more detail in FIGS. 3 and 4. These drawings correspond to FIGS. 1 and 2, and illustrate more internal detail.
Referring now more specifically to FIG. 3 of the drawings, there is shown an analog to digital encoding system in accordance with the invention. Analog input signal 99 is applied to a buffer amplifier, the first element of the analog to digital
subsystem 100. The output of the buffer amplifier drives an analog low pass anti-alias filter, which removes any high frequency components of the input signal falling above the Nyquist limit of half the sampling frequency. The output of the low pass
filter has an analog dither signal added to it and then it is applied to the input of a sampling analog to digital converter. In the converter, the signal amplitude is sampled at regular intervals and the amplitude of each sample is converted into a
number or digital word. The series of digital words from the converter make up the digital signal, which is sent to the analog to digital conversion process controller. This process controller has generated the dither signal which was added to the
analog signal before conversion, and, typically, the controller subtracts the dither from the digital signal, giving a vernier enhancement to the conversion accuracy as well as spreading any converter nonlinearities into a noise-like signal. The ADC
process controller may also make other corrections or additions to the conversion process, such as noise shaping. The output of this module is a high resolution digital signal 100a which is sent to subsystems 101 and 102. It should be noted that this
digital signal has both higher amplitude resolution and greater sampling rate or time domain resolution than the industry standard digital signal which is the final output of the encoding system.
Memory subsystem 101 is used to delay the high resolution digital signal 100a before sending it to 102e. This time delay gives subsystem 102 time to analyze the signal and choose appropriate corrective procedures to be applied during encoding.
The high resolution digital signal from subsystem 100 is also sent to the signal analysis process controller unit of subsystem 102 at input 102a. This unit analyzes the characteristics of the signal as it is being stored in the delay memory 101
and makes decisions about employing corrective procedures such as instantaneous peak amplitude limiting, low level gain compression, choice of best "brick wall" low pass filter, transient reconstruction and so forth. The unit then sends commands 102b to
the units which process the delayed digital signal to carry out the corrective procedures. The signal analysis process controller also generates a control code 102c which it sends to the code encryption unit for addition to the output signal. This
control code tells the decode system what has been done and how to recover an accurate representation of the original input signal.
The delayed high resolution digital signal from the memory subsystem 101 is sent to the decimation filter unit at 102e. Here, the oversampled input signal is decimated down to the industry standard sampling rate. The choice of optimal filter
characteristics is dependent on the nature of the program signal at the time. Such factors as transient content of the signal, presence of large amounts of alias producing high frequencies, etc. are taken into account by the signal analysis process
controller, and a filter control signal 102b tells the decimation filter which parameters to use. The output of the decimation filter has the industry standard sampling rate and very high amplitude resolution. It is sent to the level control processing
unit.
The level control processing unit uses such operations as instantaneous peak level compression and low level average gain compression to squeeze the high amplitude resolution of the signal into the industry standard resolution (such as 16 bits
for CD). These operations are done under the control of the signal analysis process controller. The level control unit may also include other techniques such as the addition of digital dither to allow resolution below the least significant bit level
and transient time domain or slew rate compression. The output of this unit is sent to the silent code encryption unit.
The silent code encryption unit takes the control codes 102c from the signal analysis process controller, which are commands and information for the decoder system, and adds them to the digital signal. One method of doing this involves
encrypting them into a pseudo-random noise-like signal and inserting it as needed into the least significant bit of the digital signal. Other methods include the use of "user" bits in standard code or unused bit combinations which may appear to be
errors to a normal decoder. The common characteristic of these methods is that they provide a silent side channel for control information which rides along with the program digital signal.
The final task of the code encryption unit is to encode the composite digital signal into an industry standard format for recording, etc. The output of this unit is a standard digital signal 103, which, for instance, could be sent to a recorder.
This completes the description of the encoding system.
Referring now to FIG. 4 of the drawings, there is shown an example of a digital to analog decode/reproduce system in accordance with the invention. The input digital | | |
