The Yamaha YMF278B , also known as the OPL4 (OPL is an acronym for FM Operator Type-L ), is a sound chip that incorporates both FM synthesis and sample-based synthesis (often incorrectly called " wavetable synthesis ") by Yamaha .
79-560: The sample synthesis part is based on pulse-code modulation (PCM). It features: The PCM synthesizer part accepts: The FM part is essentially a YMF262 (OPL3) block; thus, it is also backward-compatible with the YM3526 (OPL) and the YM3812 (OPL2). Like the OPL3, it can operate in one of four ways: Four-operator FM allows more complex sounds but reduces polyphony. Eight waveforms are available for
158-486: A capacitor to store the analog voltage at the input, and using an electronic switch or gate to disconnect the capacitor from the input. Many ADC integrated circuits include the sample and hold subsystem internally. An ADC works by sampling the value of the input at discrete intervals in time. Provided that the input is sampled above the Nyquist rate , defined as twice the highest frequency of interest, then all frequencies in
237-437: A digital encoder logic circuit that generates a binary number on the output lines for each voltage range. ADCs of this type have a large die size and high power dissipation. They are often used for video , wideband communications , or other fast signals in optical and magnetic storage . The circuit consists of a resistive divider network, a set of op-amp comparators and a priority encoder. A small amount of hysteresis
316-438: A saw-tooth signal that ramps up or down then quickly returns to zero. When the ramp starts, a timer starts counting. When the ramp voltage matches the input, a comparator fires, and the timer's value is recorded. Timed ramp converters can be implemented economically, however, the ramp time may be sensitive to temperature because the circuit generating the ramp is often a simple analog integrator . A more accurate converter uses
395-416: A voltage or current (depending on type) that represents the value presented on their digital inputs. This output would then generally be filtered and amplified for use. To recover the original signal from the sampled data, a demodulator can apply the procedure of modulation in reverse. After each sampling period, the demodulator reads the next value and transitions the output signal to the new value. As
474-470: A 500 Hz sine wave. To avoid aliasing, the input to an ADC must be low-pass filtered to remove frequencies above half the sampling rate. This filter is called an anti-aliasing filter , and is essential for a practical ADC system that is applied to analog signals with higher frequency content. In applications where protection against aliasing is essential, oversampling may be used to greatly reduce or even eliminate it. Although aliasing in most systems
553-410: A DS0 is either μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12- or 13-bit linear PCM sample number is mapped into an 8-bit value. This system is described by international standard G.711 . Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress
632-482: A NRZ system to be synchronized using in-band information, there must not be long sequences of identical symbols, such as ones or zeroes. For binary PCM systems, the density of 1-symbols is called ones-density . Ones-density is often controlled using precoding techniques such as run-length limited encoding, where the PCM code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into
711-491: A clocked counter driving a DAC. A special advantage of the ramp-compare system is that converting a second signal just requires another comparator and another register to store the timer value. To reduce sensitivity to input changes during conversion, a sample and hold can charge a capacitor with the instantaneous input voltage and the converter can time the time required to discharge with a constant current . An integrating ADC (also dual-slope or multi-slope ADC) applies
790-525: A constant current source . The time required to discharge the capacitor is proportional to the amplitude of the input voltage. While the capacitor is discharging, pulses from a high-frequency oscillator clock are counted by a register. The number of clock pulses recorded in the register is also proportional to the input voltage. If the analog value to measure is represented by a resistance or capacitance, then by including that element in an RC circuit (with other resistances or capacitances fixed) and measuring
869-432: A faithful reproduction of the original signal is only possible if the sampling rate is higher than twice the highest frequency of the signal. Since a practical ADC cannot make an instantaneous conversion, the input value must necessarily be held constant during the time that the converter performs a conversion (called the conversion time ). An input circuit called a sample and hold performs this task—in most cases by using
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#1732869763285948-446: A flow of digital values. It is therefore required to define the rate at which new digital values are sampled from the analog signal. The rate of new values is called the sampling rate or sampling frequency of the converter. A continuously varying bandlimited signal can be sampled and then the original signal can be reproduced from the discrete-time values by a reconstruction filter . The Nyquist–Shannon sampling theorem implies that
1027-422: A known voltage charging and discharging curve that can be used to solve for an unknown analog value. The Wilkinson ADC was designed by Denys Wilkinson in 1950. The Wilkinson ADC is based on the comparison of an input voltage with that produced by a charging capacitor. The capacitor is allowed to charge until a comparator determines it matches the input voltage. Then, the capacitor is discharged linearly by using
1106-455: A larger aggregate data stream , generally for transmission of multiple streams over a single physical link. One technique is called time-division multiplexing (TDM) and is widely used, notably in the modern public telephone system. The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal. These devices are digital-to-analog converters (DACs). They produce
1185-436: A longer time to measure than smaller one. And the accuracy is limited by the accuracy of the microcontroller clock and the amount of time available to measure the value, which potentially might even change during measurement or be affected by external parasitics . A direct-conversion or flash ADC has a bank of comparators sampling the input signal in parallel, each firing for a specific voltage range. The comparator bank feeds
1264-409: A pulse of a particular amplitude is always converted to the same digital value. The problem lies in that the ranges of analog values for the digitized values are not all of the same widths, and the differential linearity decreases proportionally with the divergence from the average width. The sliding scale principle uses an averaging effect to overcome this phenomenon. A random, but known analog voltage
1343-584: A rate above 3500–4300 Hz; lower rates proved unsatisfactory. In 1920, the Bartlane cable picture transmission system used telegraph signaling of characters punched in paper tape to send samples of images quantized to 5 levels. In 1926, Paul M. Rainey of Western Electric patented a facsimile machine that transmitted its signal using 5-bit PCM, encoded by an opto-mechanical analog-to-digital converter . The machine did not go into production. British engineer Alec Reeves , unaware of previous work, conceived
1422-498: A result of these transitions, the signal retains a significant amount of high-frequency energy due to imaging effects. To remove these undesirable frequencies, the demodulator passes the signal through a reconstruction filter that suppresses energy outside the expected frequency range (greater than the Nyquist frequency f s / 2 {\displaystyle f_{s}/2} ). Common sample depths for LPCM are 8, 16, 20 or 24 bits per sample . LPCM encodes
1501-451: A sampler. It cannot improve the linearity, and thus accuracy does not necessarily improve. Quantization distortion in an audio signal of very low level with respect to the bit depth of the ADC is correlated with the signal and sounds distorted and unpleasant. With dithering, the distortion is transformed into noise. The undistorted signal may be recovered accurately by averaging over time. Dithering
1580-515: A sampling rate greater than twice the bandwidth of the signal, then per the Nyquist–Shannon sampling theorem , near-perfect reconstruction is possible. The presence of quantization error limits the SNDR of even an ideal ADC. However, if the SNDR of the ADC exceeds that of the input signal, then the effects of quantization error may be neglected, resulting in an essentially perfect digital representation of
1659-502: A single sound channel. Support for multichannel audio depends on file format and relies on synchronization of multiple LPCM streams. While two channels (stereo) is the most common format, systems can support up to 8 audio channels (7.1 surround) or more. Common sampling frequencies are 48 kHz as used with DVD format videos, or 44.1 kHz as used in CDs. Sampling frequencies of 96 kHz or 192 kHz can be used on some equipment, but
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#17328697632851738-629: A time. Rather than natural binary, the grid of Goodall's later tube was perforated to produce a glitch-free Gray code and produced all bits simultaneously by using a fan beam instead of a scanning beam. In the United States, the National Inventors Hall of Fame has honored Bernard M. Oliver and Claude Shannon as the inventors of PCM, as described in "Communication System Employing Pulse Code Modulation", U.S. patent 2,801,281 filed in 1946 and 1952, granted in 1956. Another patent by
1817-515: Is a method used to digitally represent analog signals . It is the standard form of digital audio in computers, compact discs , digital telephony and other digital audio applications. In a PCM stream , the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves , Claude Shannon , Barney Oliver and John R. Pierce are credited with its invention. Linear pulse-code modulation ( LPCM )
1896-452: Is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm ). Though PCM is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to
1975-430: Is a system that converts an analog signal , such as a sound picked up by a microphone or light entering a digital camera , into a digital signal . An ADC may also provide an isolated measurement such as an electronic device that converts an analog input voltage or current to a digital number representing the magnitude of the voltage or current. Typically the digital output is a two's complement binary number that
2054-451: Is a very small amount of random noise (e.g. white noise ), which is added to the input before conversion. Its effect is to randomize the state of the LSB based on the signal. Rather than the signal simply getting cut off altogether at low levels, it extends the effective range of signals that the ADC can convert, at the expense of a slight increase in noise. Dither can only increase the resolution of
2133-535: Is added to the sampled input voltage. It is then converted to digital form, and the equivalent digital amount is subtracted, thus restoring it to its original value. The advantage is that the conversion has taken place at a random point. The statistical distribution of the final levels is decided by a weighted average over a region of the range of the ADC. This in turn desensitizes it to the width of any specific level. These are several common ways of implementing an electronic ADC. Resistor-capacitor (RC) circuits have
2212-471: Is also used in integrating systems such as electricity meters . Since the values are added together, the dithering produces results that are more exact than the LSB of the analog-to-digital converter. Dither is often applied when quantizing photographic images to a fewer number of bits per pixel—the image becomes noisier but to the eye looks far more realistic than the quantized image, which otherwise becomes banded . This analogous process may help to visualize
2291-428: Is built into the comparator to resolve any problems at voltage boundaries. At each node of the resistive divider, a comparison voltage is available. The purpose of the circuit is to compare the analog input voltage with each of the node voltages. The circuit has the advantage of high speed as the conversion takes place simultaneously rather than sequentially. Typical conversion time is 100 ns or less. Conversion time
2370-427: Is limited only by the speed of the comparator and of the priority encoder. This type of ADC has the disadvantage that the number of comparators required almost doubles for each added bit. Also, the larger the value of n, the more complex is the priority encoder. A successive-approximation ADC uses a comparator and a binary search to successively narrow a range that contains the input voltage. At each successive step,
2449-467: Is proportional to the input, but there are other possibilities. There are several ADC architectures . Due to the complexity and the need for precisely matched components , all but the most specialized ADCs are implemented as integrated circuits (ICs). These typically take the form of metal–oxide–semiconductor (MOS) mixed-signal integrated circuit chips that integrate both analog and digital circuits . A digital-to-analog converter (DAC) performs
Yamaha YMF278 - Misplaced Pages Continue
2528-459: Is the ADC's resolution in bits and E FSR is the full-scale voltage range (also called 'span'). E FSR is given by where V RefHi and V RefLow are the upper and lower extremes, respectively, of the voltages that can be coded. Normally, the number of voltage intervals is given by where M is the ADC's resolution in bits. That is, one voltage interval is assigned in between two consecutive code levels. Example: In many cases,
2607-414: Is the number of ADC bits. Clock jitter is caused by phase noise . The resolution of ADCs with a digitization bandwidth between 1 MHz and 1 GHz is limited by jitter. For lower bandwidth conversions such as when sampling audio signals at 44.1 kHz, clock jitter has a less significant impact on performance. An analog signal is continuous in time and it is necessary to convert this to
2686-407: Is uniformly distributed between − 1 ⁄ 2 LSB and + 1 ⁄ 2 LSB, and the signal has a uniform distribution covering all quantization levels, the signal-to-quantization-noise ratio (SQNR) is given by where Q is the number of quantization bits. For example, for a 16-bit ADC, the quantization error is 96.3 dB below the maximum level. Quantization error is distributed from DC to
2765-404: Is unwanted, it can be exploited to provide simultaneous down-mixing of a band-limited high-frequency signal (see undersampling and frequency mixer ). The alias is effectively the lower heterodyne of the signal frequency and sampling frequency. For economy, signals are often sampled at the minimum rate required with the result that the quantization error introduced is white noise spread over
2844-475: The Nyquist frequency . Consequently, if part of the ADC's bandwidth is not used, as is the case with oversampling , some of the quantization error will occur out-of-band , effectively improving the SQNR for the bandwidth in use. In an oversampled system, noise shaping can be used to further increase SQNR by forcing more quantization error out of band. In ADCs, performance can usually be improved using dither . This
2923-630: The SIGSALY encryption equipment, conveyed high-level Allied communications during World War II . In 1943 the Bell Labs researchers who designed the SIGSALY system became aware of the use of PCM binary coding as already proposed by Reeves. In 1949, for the Canadian Navy's DATAR system, Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances. PCM in
3002-470: The bandlimited analog input signal. The resolution of the converter indicates the number of different, i.e. discrete, values it can produce over the allowed range of analog input values. Thus a particular resolution determines the magnitude of the quantization error and therefore determines the maximum possible signal-to-noise ratio for an ideal ADC without the use of oversampling . The input samples are usually stored electronically in binary form within
3081-463: The effective number of bits (ENOB) below that predicted by quantization error alone. The error is zero for DC, small at low frequencies, but significant with signals of high amplitude and high frequency. The effect of jitter on performance can be compared to quantization error: Δ t < 1 2 q π f 0 {\displaystyle \Delta t<{\frac {1}{2^{q}\pi f_{0}}}} , where q
3160-474: The public switched telephone network (PSTN) had been largely digitized with very-large-scale integration (VLSI) CMOS PCM codec-filters, widely used in electronic switching systems for telephone exchanges , user-end modems and a wide range of digital transmission applications such as the integrated services digital network (ISDN), cordless telephones and cell phones . PCM is the method of encoding typically used for uncompressed digital audio. In
3239-408: The signal-to-noise ratio performance of the ADC and thus reduce its effective resolution. When digitizing a sine wave x ( t ) = A sin ( 2 π f 0 t ) {\displaystyle x(t)=A\sin {(2\pi f_{0}t)}} , the use of a non-ideal sampling clock will result in some uncertainty in when samples are recorded. Provided that
Yamaha YMF278 - Misplaced Pages Continue
3318-421: The ADC, so the resolution is usually expressed as the audio bit depth . In consequence, the number of discrete values available is usually a power of two. For example, an ADC with a resolution of 8 bits can encode an analog input to one in 256 different levels (2 = 256). The values can represent the ranges from 0 to 255 (i.e. as unsigned integers) or from −128 to 127 (i.e. as signed integer), depending on
3397-617: The FM synthesis: Unlike the OPL3, which has four channels for sound output, the OPL4 features six channels. For ROM wave data access, the Yamaha YRW801 2MB ROM chip can be connected to the OPL4. It holds approximately 330 samples, mostly 22.05-kHz 12-bit samples with some drums at 44.1 kHz. It is compatible with the General MIDI standard (128 melody sounds, 47 percussion sounds). For sound effects,
3476-697: The OPL4 can be connected to the Yamaha YSS225 effects processor (EP), which adds various sound effects. Like all its predecessors, the OPL4 outputs audio in digital-I/O form, thus requiring an external DAC chip. For this purpose, the Yamaha YAC513 DAC chip was used here. The YMF278B was used in the Moonsound MSX sound card and in Yamaha's SoundEdge sound card for IBM PC and compatibles. Pulse-code modulation Pulse-code modulation ( PCM )
3555-519: The actual sampling time uncertainty due to clock jitter is Δ t {\displaystyle \Delta t} , the error caused by this phenomenon can be estimated as E a p ≤ | x ′ ( t ) Δ t | ≤ 2 A π f 0 Δ t {\displaystyle E_{ap}\leq |x'(t)\Delta t|\leq 2A\pi f_{0}\Delta t} . This will result in additional recorded noise that will reduce
3634-409: The application. Resolution can also be defined electrically, and expressed in volts . The change in voltage required to guarantee a change in the output code level is called the least significant bit (LSB) voltage. The resolution Q of the ADC is equal to the LSB voltage. The voltage resolution of an ADC is equal to its overall voltage measurement range divided by the number of intervals: where M
3713-439: The benefits have been debated. The Nyquist–Shannon sampling theorem shows PCM devices can operate without introducing distortions within their designed frequency bands if they provide a sampling frequency at least twice that of the highest frequency contained in the input signal. For example, in telephony , the usable voice frequency band ranges from approximately 300 Hz to 3400 Hz. For effective reconstruction of
3792-445: The channel. In other cases, extra framing bits are added into the stream, which guarantees at least occasional symbol transitions. Another technique used to control ones-density is the use of a scrambler on the data, which will tend to turn the data stream into a stream that looks pseudo-random , but where the data can be recovered exactly by a complementary descrambler. In this case, long runs of zeroes or ones are still possible on
3871-419: The conversion periodically, sampling the input, and limiting the allowable bandwidth of the input signal. The performance of an ADC is primarily characterized by its bandwidth and signal-to-noise and distortion ratio (SNDR). The bandwidth of an ADC is characterized primarily by its sampling rate . The SNDR of an ADC is influenced by many factors, including the resolution , linearity and accuracy (how well
3950-415: The converter compares the input voltage to the output of an internal digital-to-analog converter (DAC) which initially represents the midpoint of the allowed input voltage range. At each step in this process, the approximation is stored in a successive approximation register (SAR) and the output of the digital-to-analog converter is updated for a comparison over a narrower range. A ramp-compare ADC produces
4029-539: The diagram, a sine wave (red curve) is sampled and quantized for PCM. The sine wave is sampled at regular intervals, shown as vertical lines. For each sample, one of the available values (on the y-axis) is chosen. The PCM process is commonly implemented on a single integrated circuit called an analog-to-digital converter (ADC). This produces a fully discrete representation of the input signal (blue points) that can be easily encoded as digital data for storage or manipulation. Several PCM streams could also be multiplexed into
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#17328697632854108-419: The digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques, such as modified discrete cosine transform (MDCT) coding. In telephony, a standard audio signal for a single phone call is encoded as 8,000 samples per second , of 8 bits each, giving a 64 kbit/s digital signal known as DS0 . The default signal compression encoding on
4187-417: The effect of dither on an analog audio signal that is converted to digital. An ADC has several sources of errors. Quantization error and (assuming the ADC is intended to be linear) non- linearity are intrinsic to any analog-to-digital conversion. These errors are measured in a unit called the least significant bit (LSB). In the above example of an eight-bit ADC, an error of one LSB is 1 ⁄ 256 of
4266-546: The first commercial digital recordings. In 1972, Denon unveiled the first 8-channel digital recorder, the DN-023R, which used a 4-head open reel broadcast video tape recorder to record in 47.25 kHz, 13-bit PCM audio. In 1977, Denon developed the portable PCM recording system, the DN-034R. Like the DN-023R, it recorded 8 channels at 47.25 kHz, but it used 14-bits "with emphasis , making it equivalent to 15.5 bits." In 1979,
4345-437: The first digital pop album, Bop till You Drop , was recorded. It was recorded in 50 kHz, 16-bit linear PCM using a 3M digital tape recorder. The compact disc (CD) brought PCM to consumer audio applications with its introduction in 1982. The CD uses a 44,100 Hz sampling frequency and 16-bit resolution and stores up to 80 minutes of stereo audio per disc. The rapid development and wide adoption of PCM digital telephony
4424-499: The full signal range, or about 0.4%. All ADCs suffer from nonlinearity errors caused by their physical imperfections, causing their output to deviate from a linear function (or some other function, in the case of a deliberately nonlinear ADC) of their input. These errors can sometimes be mitigated by calibration , or prevented by testing. Important parameters for linearity are integral nonlinearity and differential nonlinearity . These nonlinearities introduce distortion that can reduce
4503-453: The information to be encoded is represented by discrete signal pulses of varying width or position, respectively. In this respect, PCM bears little resemblance to these other forms of signal encoding, except that all can be used in time-division multiplexing, and the numbers of the PCM codes are represented as electrical pulses. Analog-to-digital converter In electronics , an analog-to-digital converter ( ADC , A/D , or A-to-D )
4582-449: The late 1940s and early 1950s used a cathode-ray coding tube with a plate electrode having encoding perforations. As in an oscilloscope , the beam was swept horizontally at the sample rate while the vertical deflection was controlled by the input analog signal, causing the beam to pass through higher or lower portions of the perforated plate. The plate collected or passed the beam, producing current variations in binary code, one bit at
4661-494: The logarithm of the resolution, i.e. the number of bits. Flash ADCs are certainly the fastest type of the three; The conversion is basically performed in a single parallel step. There is a potential tradeoff between speed and precision. Flash ADCs have drifts and uncertainties associated with the comparator levels results in poor linearity. To a lesser extent, poor linearity can also be an issue for successive-approximation ADCs. Here, nonlinearity arises from accumulating errors from
4740-769: The original analog signal: the sampling rate , which is the number of times per second that samples are taken; and the bit depth , which determines the number of possible digital values that can be used to represent each sample. Early electrical communications started to sample signals in order to multiplex samples from multiple telegraphy sources and to convey them over a single telegraph cable. The American inventor Moses G. Farmer conceived telegraph time-division multiplexing (TDM) as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanical commutator for time-division multiplexing multiple telegraph signals; he also applied this technology to telephony . He obtained intelligible speech from channels sampled at
4819-518: The output but are considered unlikely enough to allow reliable synchronization. In other cases, the long term DC value of the modulated signal is important, as building up a DC bias will tend to move communications circuits out of their operating range. In this case, special measures are taken to keep a count of the cumulative DC bias and to modify the codes if necessary to make the DC bias always tend back to zero. Many of these codes are bipolar codes , where
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#17328697632854898-410: The performance of the ADC can be greatly increased at little or no cost. Furthermore, as any aliased signals are also typically out of band, aliasing can often be eliminated using very low cost filters. The speed of an ADC varies by type. The Wilkinson ADC is limited by the clock rate which is processable by current digital circuits. For a successive-approximation ADC , the conversion time scales with
4977-540: The pulses can be positive, negative or absent. In the typical alternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes. The word pulse in the term pulse-code modulation refers to the pulses to be found in the transmission line. This perhaps is a natural consequence of this technique having evolved alongside two analog methods, pulse-width modulation and pulse-position modulation , in which
5056-411: The quantization levels match the true analog signal), aliasing and jitter . The SNDR of an ADC is often summarized in terms of its effective number of bits (ENOB), the number of bits of each measure it returns that are on average not noise . An ideal ADC has an ENOB equal to its resolution. ADCs are chosen to match the bandwidth and required SNDR of the signal to be digitized. If an ADC operates at
5135-410: The reverse function; it converts a digital signal into an analog signal. An ADC converts a continuous-time and continuous-amplitude analog signal to a discrete-time and discrete-amplitude digital signal . The conversion involves quantization of the input, so it necessarily introduces a small amount of quantization error . Furthermore, instead of continuously performing the conversion, an ADC does
5214-514: The same title was filed by John R. Pierce in 1945, and issued in 1948: U.S. patent 2,437,707 . The three of them published "The Philosophy of PCM" in 1948. The T-carrier system, introduced in 1961, uses two twisted-pair transmission lines to carry 24 PCM telephone calls sampled at 8 kHz and 8-bit resolution. This development improved capacity and call quality compared to the previous frequency-division multiplexing schemes. In 1973, adaptive differential pulse-code modulation (ADPCM)
5293-499: The signal can be reconstructed. If frequencies above half the Nyquist rate are sampled, they are incorrectly detected as lower frequencies, a process referred to as aliasing. Aliasing occurs because instantaneously sampling a function at two or fewer times per cycle results in missed cycles, and therefore the appearance of an incorrectly lower frequency. For example, a 2 kHz sine wave being sampled at 1.5 kHz would be reconstructed as
5372-406: The subtraction processes. Wilkinson ADCs have the best linearity of the three. The sliding scale or randomizing method can be employed to greatly improve the linearity of any type of ADC, but especially flash and successive approximation types. For any ADC the mapping from input voltage to digital output value is not exactly a floor or ceiling function as it should be. Under normal conditions,
5451-415: The time it takes to charge (and/or discharge) its capacitor from 1 ⁄ 3 V supply to 2 ⁄ 3 V supply . By sending this pulse into a microcontroller with an accurate clock, the duration of the pulse can be measured and converted using the capacitor charging equation to produce the value of the unknown resistance or capacitance. Larger resistances and capacitances will take
5530-550: The time to charge the capacitance from a known starting voltage to another known ending voltage through the resistance from a known voltage supply, the value of the unknown resistance or capacitance can be determined using the capacitor charging equation: V capacitor ( t ) = V supply ( 1 − e − t R C ) {\displaystyle V_{\text{capacitor}}(t)=V_{\text{supply}}\left(1-e^{-{\frac {t}{RC}}}\right)} and solving for
5609-411: The unknown input voltage to the input of an integrator and allows the voltage to ramp for a fixed time period (the run-up period). Then a known reference voltage of opposite polarity is applied to the integrator and is allowed to ramp until the integrator output returns to zero (the run-down period). The input voltage is computed as a function of the reference voltage, the constant run-up time period, and
5688-464: The unknown resistance or capacitance using those starting and ending datapoints. This is similar but contrasts to the Wilkinson ADC which measures an unknown voltage with a known resistance and capacitance, by instead measuring an unknown resistance or capacitance with a known voltage. For example, the positive (and/or negative) pulse width from a 555 Timer IC in monostable or astable mode represents
5767-565: The use of PCM for voice communication in 1937 while working for International Telephone and Telegraph in France. He described the theory and its advantages, but no practical application resulted. Reeves filed for a French patent in 1938, and his US patent was granted in 1943. By this time Reeves had started working at the Telecommunications Research Establishment . The first transmission of speech by digital techniques,
5846-418: The useful resolution of a converter is limited by the signal-to-noise ratio (SNR) and other errors in the overall system expressed as an ENOB. Quantization error is introduced by the quantization inherent in an ideal ADC. It is a rounding error between the analog input voltage to the ADC and the output digitized value. The error is nonlinear and signal-dependent. In an ideal ADC, where the quantization error
5925-692: The voice signal even further. An ADPCM algorithm is used to map a series of 8-bit μ-law or A-law PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard. Audio coding formats and audio codecs have been developed to achieve further compression. Some of these techniques have been standardized and patented. Advanced compression techniques, such as modified discrete cosine transform (MDCT) and linear predictive coding (LPC), are now widely used in mobile phones , voice over IP (VoIP) and streaming media . PCM can be either return-to-zero (RZ) or non-return-to-zero (NRZ). For
6004-452: The voice signal, telephony applications therefore typically use an 8000 Hz sampling frequency which is more than twice the highest usable voice frequency. Regardless, there are potential sources of impairment implicit in any PCM system: Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the analog-to-digital process; newer implementations do so in
6083-433: The whole passband of the converter. If a signal is sampled at a rate much higher than the Nyquist rate and then digitally filtered to limit it to the signal bandwidth produces the following advantages: Oversampling is typically used in audio frequency ADCs where the required sampling rate (typically 44.1 or 48 kHz) is very low compared to the clock speed of typical transistor circuits (>1 MHz). In this case,
6162-560: Was developed, by P. Cummiskey, Nikil Jayant and James L. Flanagan . In 1967, the first PCM recorder was developed by NHK 's research facilities in Japan. The 30 kHz 12-bit device used a compander (similar to DBX Noise Reduction ) to extend the dynamic range, and stored the signals on a video tape recorder . In 1969, NHK expanded the system's capabilities to 2-channel stereo and 32 kHz 13-bit resolution. In January 1971, using NHK's PCM recording system, engineers at Denon recorded
6241-438: Was enabled by metal–oxide–semiconductor (MOS) switched capacitor (SC) circuit technology, developed in the early 1970s. This led to the development of PCM codec-filter chips in the late 1970s. The silicon-gate CMOS (complementary MOS) PCM codec-filter chip, developed by David A. Hodges and W.C. Black in 1980, has since been the industry standard for digital telephony. By the 1990s, telecommunication networks such as
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