Time interleaved (TI) ADCs are Analog-to-Digital Converters (ADCs) that involve M converters working in parallel. [1] Each of the M converters is referred to as sub-ADC, channel or slice in the literature. The time interleaving technique, akin to multithreading in computing, involves using multiple converters in parallel to sample the input signal at staggered intervals, increasing the overall sampling rate and improving performance without overburdening the single ADCs.
The concept of time interleaving can be traced back to the 1960s. One of the earliest mentions of using multiple ADCs to increase sampling rates appeared in the work of Bernard M. Oliver and Claude E. Shannon. Their pioneering work on communication theory and sampling laid the groundwork for the theoretical basis of time interleaving. However, practical implementations were limited by the technology of the time.
In the 1980s, significant advancements were made: W. C. Black and D. A. Hodges from the Berkeley University successfully implemented the first prototype of a time interleaved ADC. In particular, they designed a 4-way interleaved converter working at 2.5 MSample/s. Each slice of the converter was a 7-stage SAR pipeline ADC running at 625 kSample/s. An effective number of bits (ENOB) equal to 6.2 was measured for the proposed converter with a probing input signal at 100 kHz. The work was presented at ISSCC 1980 and the paper was focused on the practical challenges of implementing TI ADCs, including the synchronization and calibration of multiple channels to reduce mismatches. [2]
In 1987, Ken Poulton and other researchers of the HP Labs developed the first product based on Time Interleaved ADCs: the HP 54111D digital oscilloscope. [2]
In the 1990s, the TI ADC technology saw further advancements driven by the increasing demand for high-speed data conversion in telecommunications and other fields. A notable project during this period was the development of high-speed ADCs for digital oscilloscopes by Tektronix. Engineers at Tektronix implemented TI ADCs to achieve the high sampling rates necessary for capturing fast transient signals in test and measurement equipment. As a result of this work, the Tektronix TDS350, a two-channel, 200 MHz, 1 GSample/s digital storage scope, was commercialized in 1991. [2]
By the late 1990s, TI ADCs had become commercially viable. One of the key projects that showcased the potential of TI ADCs was the development of the GSM (Global System for Mobile Communications) standard, where high-speed ADCs were essential for digital signal processing in mobile phones. Companies like Analog Devices and Texas Instruments began to offer TI ADCs as standard products, enabling widespread adoption in various applications. [2]
The 21st century has seen continued innovation in TI ADC technology. Researchers and engineers have focused on further improving the performance and integration of TI ADCs to meet the growing demands of digital systems. Key figures in this era include Boris Murmann and his colleagues at Stanford University, who have contributed to the development of advanced calibration techniques and low-power design methods for TI ADCs.
Today, TI ADCs are used in a wide range of applications, from 5G telecommunications to high-resolution medical imaging. The future of TI ADCs looks promising, with ongoing research focusing on further improving their performance and expanding their application areas. Emerging technologies such as autonomous vehicles, advanced radar systems, and artificial intelligence-driven signal processing will continue to drive the demand for high-speed, high-resolution ADCs.
In a time-interleaved system, the conversion time required by each sub-ADC is equal to . If the outputs of the multiple channels are properly combined, the overall system can be considered as a single converter operating at a sampling period equal to , where represents the number of channels or sub-ADCs in the TI system.
To illustrate this concept, let us delve into the conversion process of a TI ADC with reference to the first figure of this paragraph. The figure shows the time diagram of a data converter that employs four interleaved channels. The input signal (depicted as a blue waveform) is a sinusoidal wave at frequency . Here, is the clock frequency, which is the reciprocal of , the overall sampling period of the TI ADC. This relationship aligns with the Shannon-Nyquist sampling theorem, which states that the sampling rate must be at least twice the highest frequency present in the input signal to accurately reconstruct the signal without aliasing. [3]
In a TI ADC, every , one of the channels acquires a sample of the input signal. The conversion operation performed by each sub-ADC takes seconds and, after the conversion, a digital multiplexer sequentially selects the output from one of the sub-ADCs. This selection occurs in a specific order, typically from the first sub-ADC to the sub-ADC, and then the cycle repeats. [1]
At any given moment, each channel is engaged in converting different samples. Consequently, the aggregate data rate of the system is faster than the data rate of a single sub-ADC by a factor of . This is because the TI system essentially parallelizes the conversion process across multiple sub-ADCs. The factor , representing the number of interleaved channels, thus represents the increase in the overall sampling rate of the entire system. [1]
To conclude, the time-interleaving method effectively increases the conversion speed of each sub-ADC by times. As a result, even though each sub-ADC operates at a relatively slow pace, the combined output of the TI system is characterized by a higher sampling rate. Time interleaving is therefore a powerful technique in the design and implementation of data converters since it enables the creation of high-speed ADCs using components that individually have much lower performance capabilities in terms of speed.
Two architectures are possible to implement a time interleaved ADC. [4] The first architecture is depicted in the first figure of the paragraph and it is characterized by the presence of a single Sample and Hold (S&H) circuit for the entire structure. The sampler operates at a frequency and acquires the samples for all the channels of the TI ADC. Once a sample is acquired, an analog demultiplexer distributes it to the correspondent sub-ADC. This approach centralizes the sampling process, ensuring uniformity in the acquired samples. However, it places stringent speed requirements on the S&H circuit since it must operate at the full sampling rate of the ADC system.
In contrast, the second architecture, illustrated in the second figure of the paragraph, employs different S&H circuits for each channel, each operating at a reduced frequency , where is once again the number of interleaved channels. This solution significantly relaxes the speed requirements for each S&H circuit, as they only need to operate at a fraction of the overall sampling rate. This approach mitigates the challenge of high-speed operation of the first architecture. However, this benefit comes with trade-offs, namely, increased area occupation and higher power dissipation due to the additional circuitry required to implement multiple S&H circuits.
The choice between these two architectures depends on the specific requirements and constraints of the application. The single S&H circuit architecture offers a compact and potentially lower-power solution, as it eliminates the redundancy of multiple S&H circuits. The centralized sampling can also reduce mismatches between channels, as all samples are derived from a single source. However, the high-speed requirement of the single S&H circuit can be a significant challenge, particularly at very high sampling rates where achieving the necessary performance may require more advanced and costly technology. [4]
On the other hand, the multiple S&H circuit architecture distributes the sampling load, allowing each S&H circuit to operate at a lower speed. This can be advantageous in applications where high-speed circuits are difficult or expensive to implement. Additionally, this architecture can offer improved flexibility in managing timing and gain mismatches between channels. Each S&H circuit can be independently optimized for its specific operating conditions, potentially leading to better overall performance. The trade-offs include a larger footprint on the integrated circuit and increased power consumption, which may be critical factors in power-sensitive or space-constrained applications. [4]
In practical implementations, the choice between these architectures is influenced by several factors, including the required sampling rate, power budget, available silicon area, and the acceptable level of complexity in calibration and error correction. For instance, in high-speed communication systems the single S&H circuit architecture might be preferred despite its stringent speed requirements, due to its compact design and potentially lower power consumption. Conversely, in applications where power is less of a concern but achieving ultra-high speeds is challenging, the multiple S&H circuit architecture might be more suitable.
Ideally, all the sub-ADCs are identical. In practice, however, they end up being slightly different due to process, voltage and temperature (PVT) variations. If not taken care of, sub-ADC mismatches can jeopardize the performance of TI ADCs since they show up in the output spectrum as spectral tones. [5]
Offset mismatches (i.e., different input-referred offsets for each sub-ADC) are superimposed to the converted signal as a sequence of period , affecting the output spectrum of the ADC with spurious tones, whose power depends on the magnitude of the offsets, located at frequencies , where M represents the number of channels and k is an arbitrary integer number from to . [5]
Gain errors affect the amplitude of the converted signal and are transferred to the output as an amplitude modulation (AM) of the input signal with a sequence of period . As a matter of fact, this mechanism introduces spurious harmonics at frequencies , whose power depends both on the amplitude of the input signal and on the magnitude of the gain error sequence. [5]
Finally, skew mismatches are due to the channels being timed by different phases of the same clock signal. If one timing signal is skewed with respect to the others, spurious harmonics will be generated in the output spectrum. It can be demonstrated that these spurs are located at the frequencies . Moreover, their power depends both on the magnitude of the skew between the control phases and on the value of the input signal frequency. [5]
Channel mismatches in a TI ADC can seriously degrade its Spurious-Free Dynamic Range (SFDR) and its Signal-to-Noise-and-Distortion Ratio (SNDR). To recover the spectral purity of the converter, the proven solution consists of compensating these non-idealities with digitally implemented corrections. Despite being able to recover the overall spectral purity by suppressing the mismatch spurs, digital calibrations can significantly increase the overall power consumption of the receiver and may not be as effective when the input signal is broadband. For this reason, methods to provide higher stability and usability in real-world cases should be actively researched. [5]
As cellular communications systems evolve, the performance of the receivers becomes more and more demanding. For example, the channel bandwidth offered by the 4G network can be as high as 20 MHz, whereas it can range from 400 MHz up to 1 GHz in the current 5G NR network. [6] On top of that, the complexity of signal modulation also increased from 64-QAM in 4G to 256-QAM in 5G NR.
The tighter requirements impose new design challenges on modern receivers, whose performance relies on the analog-to-digital interface provided by ADCs. In 4G receivers, the data conversion is performed by Delta-Sigma Modulators (DSMs), [7] as they are easily reconfigurable: It is sufficient to modify the oversampling ratio (OSR), the loop order or the quantizer resolution to adjust the bandwidth of the data converter according to the need. [7] This is a desirable feature of an ADC in receivers supporting multiple wireless communication standards.
In 5G receivers, instead, DSMs are not the preferred choice: The bandwidth of the receiver has to be higher than a few hundreds of MHz, whereas the signal bandwidth of a DSM is only a fraction of half of the sampling frequency . In mathematical terms, . Thus, in practice, it is hard if not impossible to achieve the required sampling frequency with DSMs. For this reason, 5G receivers typically rely on Nyquist ADCs, in which the signal bandwidth can be as high as , according to the Shannon-Nyquist sampling theorem.
The ADCs employed in 5G receivers do not only require a high sampling rate to deal with large channel bandwidths, but also a reasonable number of bits. A high resolution is necessary for the data converter to enable the use of the high-order modulation schemes, which are fundamental to achieve high throughputs with an efficient bandwidth utilization. The resolution of a data converter is defined as the minimum voltage value that it can resolve, i.e., its Least Significant Bit (LSB). The latter parameter depends on the number of physical bits (N) of the converter as (where FSR is the full scale range of the ADC). Hence, the larger the number of levels, the finer the conversion will be. In practice, however, noise (e.g., jitter and thermal noise) poses a fundamental limit on the achievable resolution, which is lower than the physical number of bits and it is typically expressed in terms of ENOB.
Usually, for 5G receivers, ADCs with an ENOB of 12 bits and bandiwdth up to the GHz are the favorable choice. [6] Time interleaved ADCs are frequently employed for this application since they are capable of meeting the above-mentioned requirements. In fact, TI ADCs utilize multiple ADC channels operating in parallel and this technique effectively increases the overall sampling rate, allowing the receiver to handle the wide bandwidths required by 5G network.
A receiver is one of the essential components of a communication system. In particular, a receiver is responsible for the conversion of radio signals in digital words to allow the signal to be further processed by electronic devices. Typically, a receiver include an antenna, a pre-selector filter, a low-noise amplifier (LNA), a mixer, a local oscillator, an intermediate frequency (IF) filter, a demodulator and an analog-to-digital converter.
The antenna is the first component in a receiver system; it captures electromagnetic waves from the air and it converts these radio waves into electrical signals. These signals are then filtered by the pre-selector, which guarantees that only the desired frequency range from the signals captured by the antenna are passed to the next stages of the receiver. The signal is then amplified by an LNA. The amplification action ensures that the signal is strong enough to be processed effectively by the subsequent stages of the system. The amplified signal is then mixed with a stable signal from the local oscillator to produce an intermediate frequency (IF) signal. This process, known as heterodyning, shifts the frequency of the received signal to a lower, more manageable IF. The IF signal undergoes further filtering to remove any remaining unwanted signals and noise. Finally, a demodulator extracts the original information signal from the modulated carrier wave. Precisely, the demodulator converts the IF signal back into the baseband signal, which contains the transmitted information. Different demodulation techniques can be used depending on the type of modulation employed (e.g., amplitude modulation (AM), frequency modulation (FM), or phase modulation (PM)). As a last step, an ADC converts the continuous analog signal into a discrete digital signal, which can be processed by digital signal processors (DSPs) or microcontrollers. This step is crucial for enabling advanced digital signal processing techniques.
To further improve the power efficiency and cost of a receiver, the paradigm of Direct RF Sampling is emerging. According to this technique, the analog signal at radio frequency is simply fed to the ADC, avoiding the downconversion to an intermediate frequency altogether. [8]
Direct RF Sampling has significant advantages in terms of system design and performance. By removing the downconversion stage, the design complexity is reduced, leading to lower power consumption and cost. Additionally, the absence of the mixer and local oscillator means there are fewer components that can introduce noise and distortion, potentially improving the signal-to-noise ratio (SNR) and linearity of the receiver. [8]
However, directly sampling the radio-frequency signal imposes stringent requirements on the performance of the ADC. The signal bandwidth of the ADC in the receiver must be a few GHz to handle the high-frequency signals directly. Achieving such high values with a single ADC is challenging due to limitations in speed, power consumption and resolution. [8]
To meet these demanding requirements, Time interleaved ADC systems are typically adopted. In fact, TI ADCs utilize multiple slower sub-ADCs operating in parallel, each sampling the input signal at different time intervals. By interleaving the sampling process, the effective sampling rate of the overall system is increased, allowing it to handle the high bandwidths required for direct RF sampling.
Bandwidth is the difference between the upper and lower frequencies in a continuous band of frequencies. It is typically measured in unit of hertz.
In telecommunications, orthogonal frequency-division multiplexing (OFDM) is a type of digital transmission used in digital modulation for encoding digital (binary) data on multiple carrier frequencies. OFDM has developed into a popular scheme for wideband digital communication, used in applications such as digital television and audio broadcasting, DSL internet access, wireless networks, power line networks, and 4G/5G mobile communications.
In radio communications, single-sideband modulation (SSB) or single-sideband suppressed-carrier modulation (SSB-SC) is a type of modulation used to transmit information, such as an audio signal, by radio waves. A refinement of amplitude modulation, it uses transmitter power and bandwidth more efficiently. Amplitude modulation produces an output signal the bandwidth of which is twice the maximum frequency of the original baseband signal. Single-sideband modulation avoids this bandwidth increase, and the power wasted on a carrier, at the cost of increased device complexity and more difficult tuning at the receiver.
The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample rate must be at least twice the bandwidth of the signal to avoid aliasing. In practice, it is used to select band-limiting filters to keep aliasing below an acceptable amount when an analog signal is sampled or when sample rates are changed within a digital signal processing function.
In electronics, an analog-to-digital converter 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 is proportional to the input, but there are other possibilities.
Delta modulation is an analog-to-digital and digital-to-analog signal conversion technique used for transmission of voice information where quality is not of primary importance. DM is the simplest form of differential pulse-code modulation (DPCM) where the difference between successive samples is encoded into n-bit data streams. In delta modulation, the transmitted data are reduced to a 1-bit data stream representing either up (↗) or down (↘). Its main features are:
Signal-to-noise ratio is a measure used in science and engineering that compares the level of a desired signal to the level of background noise. SNR is defined as the ratio of signal power to noise power, often expressed in decibels. A ratio higher than 1:1 indicates more signal than noise.
In electronics, a digital-to-analog converter is a system that converts a digital signal into an analog signal. An analog-to-digital converter (ADC) performs the reverse function.
In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or space; this definition differs from the term's usage in statistics, which refers to a set of such values.
In signal processing, undersampling or bandpass sampling is a technique where one samples a bandpass-filtered signal at a sample rate below its Nyquist rate, but is still able to reconstruct the signal.
In signal processing, oversampling is the process of sampling a signal at a sampling frequency significantly higher than the Nyquist rate. Theoretically, a bandwidth-limited signal can be perfectly reconstructed if sampled at the Nyquist rate or above it. The Nyquist rate is defined as twice the bandwidth of the signal. Oversampling is capable of improving resolution and signal-to-noise ratio, and can be helpful in avoiding aliasing and phase distortion by relaxing anti-aliasing filter performance requirements.
Continuous-wave radar is a type of radar system where a known stable frequency continuous wave radio energy is transmitted and then received from any reflecting objects. Individual objects can be detected using the Doppler effect, which causes the received signal to have a different frequency from the transmitted signal, allowing it to be detected by filtering out the transmitted frequency.
Delta-sigma modulation is an oversampling method for encoding signals into low bit depth digital signals at a very high sample-frequency as part of the process of delta-sigma analog-to-digital converters (ADCs) and digital-to-analog converters (DACs). Delta-sigma modulation achieves high quality by utilizing a negative feedback loop during quantization to the lower bit depth that continuously corrects quantization errors and moves quantization noise to higher frequencies well above the original signal's bandwidth. Subsequent low-pass filtering for demodulation easily removes this high frequency noise and time averages to achieve high accuracy in amplitude which can be ultimately encoded as pulse-code modulation (PCM).
A flash ADC is a type of analog-to-digital converter that uses a linear voltage ladder with a comparator at each "rung" of the ladder to compare the input voltage to successive reference voltages. Often these reference ladders are constructed of many resistors; however, modern implementations show that capacitive voltage division is also possible. The output of these comparators is generally fed into a digital encoder, which converts the inputs into a binary value.
Effective number of bits (ENOB) is a measure of the dynamic range of an analog-to-digital converter (ADC), digital-to-analog converter, or their associated circuitry. The resolution of an ADC is specified by the number of bits used to represent the analog value. Ideally, a 12-bit ADC will have an effective number of bits of almost 12. However, real signals have noise, and real circuits are imperfect and introduce additional noise and distortion. Those imperfections reduce the number of bits of accuracy in the ADC. The ENOB describes the effective resolution of the system in bits. An ADC may have a 12-bit resolution, but the effective number of bits, when used in a system, may be 9.5.
A fully differential amplifier (FDA) is a DC-coupled high-gain electronic voltage amplifier with differential inputs and differential outputs. In its ordinary usage, the output of the FDA is controlled by two feedback paths which, because of the amplifier's high gain, almost completely determine the output voltage for any given input.
The time-stretch analog-to-digital converter (TS-ADC), also known as the time-stretch enhanced recorder (TiSER), is an analog-to-digital converter (ADC) system that has the capability of digitizing very high bandwidth signals that cannot be captured by conventional electronic ADCs. Alternatively, it is also known as the photonic time-stretch (PTS) digitizer, since it uses an optical frontend. It relies on the process of time-stretch, which effectively slows down the analog signal in time before it can be digitized by a standard electronic ADC.
The Serial Low-power Inter-chip Media Bus (SLIMbus) is a standard interface between baseband or application processors and peripheral components in mobile terminals. It was developed within the MIPI Alliance, founded by ARM, Nokia, STMicroelectronics and Texas Instruments. The interface supports many digital audio components simultaneously, and carries multiple digital audio data streams at differing sample rates and bit widths.
An integrating ADC is a type of analog-to-digital converter that converts an unknown input voltage into a digital representation through the use of an integrator. In its basic implementation, the dual-slope converter, the unknown input voltage is applied to the input of the integrator and allowed to ramp for a fixed time 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 input voltage is computed as a function of the reference voltage, the constant run-up time period, and the measured run-down time period. The run-down time measurement is usually made in units of the converter's clock, so longer integration times allow for higher resolutions. Likewise, the speed of the converter can be improved by sacrificing resolution.
An RF chain is a cascade of electronic components and sub-units which may include amplifiers, filters, mixers, attenuators and detectors. It can take many forms, for example, as a wide-band receiver-detector for electronic warfare (EW) applications, as a tunable narrow-band receiver for communications purposes, as a repeater in signal distribution systems, or as an amplifier and up-converters for a transmitter-driver. In this article, the term RF covers the frequency range "Medium Frequencies" up to "Microwave Frequencies", i.e. from 100 kHz to 20 GHz.