Analog and telecommunication

electronics

D5 - Special A/D converters

» Differential converters
» Oversampling, noise shaping
» Logarithmic conversion
» Approximation, A and μ laws
» Model encoding

Analog and telecom. electronics D5 - Special A/D converters 1 / 63

 D5 - Special A/D converters
Differential converters
♦Adaptive converters, Sigma-delta converters
♦Oversampling, Noise shaping

Voice conversion, SNRq and dynamic range

Logarithmic conversion
♦Piecewise approximation, A and μ laws

Waveform encoding and model encoding

♦Voice LPC
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 Radio systems: where are ADC/DAC ?
Services
♦ V battery, TX power, …..
Baseband chain
♦ A/D e D/A for voice signals
Receiver chain:
♦ A/D conversion of I/Q components in the IF channel
Transmitter chain
♦ D/A conversion of synthesized I/Q components

Software Defined Radio architectures

♦ Most functions by digital/programmable circuits
A/D or D/A conversion very close to antenna
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 A/D and D/A conversion: where ?

A/D and D/A converters

for voice signal.

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 ADC and DAC system goals
Improve cost/performance figure
♦ Cost factors
• Complexity, Bit rate, ….
♦ Performance parameters
• Bandwidth, Precision, ….
Signals with known features
♦ Amplitude distribution
♦ Statistic parameters
♦ Source model
ADC/DAC optimized for specific applications
♦ Low cost (no high precision component)
♦ Low bit rate
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 Tracking converter
The tracking ADC is a differential converter
♦The serial bit flow from the comparator output
represents the sign of A - A’ (current value –
previous value)

SERIAL
DATA
A + CK
- U/D counter
A’

DAC D

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 Differential converters
Quantization of difference between previous and
current values
♦Dynamic reduction
♦1-bit A/D conversion (comparator)
♦Serial flow of uniform bits
CODER DECODER

counter

counter

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 Δ (Delta/differential) converter
Integrating differential converter
♦Counter + DAC replaced by SW + integrator
♦L is a sequence of + or – pulses, with rate Fck = 1/Tck
♦The recovered signal is R = Σ(L)
♦On each pulse R changes of one step γ (+ or -)

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 Signals in the Δ converter
L is a sequence of positive or negative pulses, with rate Fck = 1/Tck
The recovered signal is S(L)
On each pulse R changes of one step γ (positive or negative).

TCK

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 Delta ADC dynamic

Minimum signal (IDLE state)

♦Peak level γ/2; idle noise
Maximum tracked signal
♦Slew rate γ/Tck overload
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 Characteristic of Delta (Δ) ADC
A differential converter
♦ Does not require high precision devices
♦ Does not require formatting of serial output data

Provides limited dynamic range

♦ Low bound: idle noise
♦ High bound: overload
♦ SNRq depends on bit rate (clock)
♦ Higher SNRq  bit flow with higher rate

Operates in oversampling mode

♦ Sampling rate far higher than Nyquist limit
♦ Higher bit flow (compared with Nyquist minimum)
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 Oversampling
Sampling at a rate far higher than the Nyquist limit
♦ Example: 3 kHz audio signal (Nyquist = 6 kS/s)
8 kS/s  Nyquist sampling; 1 MS/s  Oversampling
Oversampling sends aliased spectra far from baseband
♦ Reduced aliasing noise (spectrum folded from first alias)
♦ Relaxed specifications on the anti-alias input filter
Quantization noise is spread over a wider band (0 - Fs)
♦ Reduced spectral density of quantization noise
Higher bit rate (more samples/s)
♦ Can be reduced with digital filtering
Move complexity from analog  digital domain

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 Oversampling vs. Nyquist
Main spectrum
Nyquist (baseband) First alias Second alias

X(ω)
f

0 FS1 2FS1
Quantization noise (0-Fs1 band)

Oversampling

X(ω) First alias

0 FS2
Quantization noise (0-Fs2 band)

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 Filtering oversampled signals
Nyquist Steep filter
X(ω)
f

0 FS1 2FS1
Different filters:
same quantization noise power
(after reconstruction filter)
Oversampling

X(ω) Smooth filter

f
0 FS2

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 Oversampling vs. Nyquist noise
Nyquist Steep filter
X(ω)
f

0 FS1 2FS1
Same filter, higher sampling rate:
- Reduced quantization noise power
(after reconstruction filter)  higher SNRq
Oversampling
Steep filter
Removed quantization noise
X(ω)
f
0 FS2
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 Which is the actual limit ?
Actual Nyquist rule:
♦ A signal must be sampled at least
twice the signal BANDWIDTH
♦ Example:
a 1 GHz carrier, 100 kHz BW signal
can be safely sampled at Fs > 200 ks/s
♦ Spectrum is folded around K Fs/2
Less stringent specs for RF A/D converters
♦ Sampling rate related with bandwidth, not carrier
Tight specs for the S/H
♦ sampling jitter related with carrier, not bandwidth

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 Filter for Nyquist sampling

Steep antialias filter,

to limit aliasing noise Spectrum segment
folded to baseband
X(ω) (aliasing noise)
f

0 FS 2FS
FS/2

A/D

Complex analog LP filter

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 Oversampling: simpler filter
Complex, steep digital filter: Alias is far away;
- reduce noise antialias analog
- reduce bit rate (decimation) filter can be
simple

X(ω)
ω

0 2πFS2

A/D

Complex digital filter

Simple analog filter Can reduce the bit rate
(decimation)

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 Nyquist vs oversampling
NYQUIST  Complex analog LP filter

A/D

OVERSAMPLING  Simple analog LP filter

A/D

Complex digital filter

Move complexity from the Can reduce the bit rate
analog to the digital domain (decimation)

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 Δ converter input dynamic range
Range of input signals correctly handled
♦ γ corresponds to the quantization step AD in a standard ADC
♦ Input dynamic range: Fck/πFs
• Does not depend on γ
To increase input dynamic range
♦ γ constant  change Tck (Fck)  higher bit rate
♦ γ variable (adaptive converters)
• Minimum in idle condition (output sequence 0-1-0-1-0-…)
• Maximum near overload (output sequences 000… or 111... )
♦Remove dependency from signal frequency (ω)
• Σ−∆ converters
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 Adaptive converters
Two integrators in the loop
♦ Stability problems
♦ Integrator + predictor (pole/zero)
Variable step γ , depending from
♦ Signal level (power estimation)
• Syllabic adaptation
♦ Error sign sequences
• Real-time adaptation
Adaptation circuits must use the line signal
♦ idle: alternated 0-1-0-1… sequence at output
♦ overload: continuous streams 0000… or 1111...

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 Adaptive converters

Power
estimation

DAC uses only line signal

Power
estimation

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 Differential converter architectures
The differential converter can operate on many
bits
♦The comparator is replaced by an ADC
♦A DAC drives the integrator

Integrator

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 Digital differential converters
Integration can occur in the digital domain

♦Integrator becomes accumulator

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 Σ∆ converters
Input dynamic range is limited by signal slew rate
♦For wider dynamic: limit slew rate
Integrator on input signal
♦Decrease amplitude as frequency goes up (integrator)
 constant slew rate

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 Σ∆ converters
Input dynamic range is limited by signal slew rate

Integrator on input signal

To correctly rebuild the signal: derive the output

Standard differential chain

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 Sigma-Delta ADC and DAC
Move integrators on adder input
 single integrator at the output
Remove the integrator-derivator in DAC

ADC DAC

Keep antialias input and reconstruction output

filters (not shown)
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 Quantization noise in Σ−∆
In the Σ−∆ ADC quantization noise εq is
generated after integraton

Y/N transfer function

is highpass

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 Noise shaping
Noise is shifted towards high frequencies

Noise power spectrum density is higher at high

frequencies:
♦Noise shaping

Noise power spectrum density in baseband is

reduced

Further reduction to output noise power

♦Or simpler reconstruction filter
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 Oversampling vs. Nyquist noise

Oversampling
Reconstruction Flat quantization noise
X(ω) filter

f
0 FS2
Noise shaping
Shaped quantization noise
X(ω)
f
0 FS2
Noise power is moved to HF,
lower power density in baseband

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 Complete Σ−∆ conversion chain
Anti aliasing filter
• Oversampling allows simple filters
ADC Σ−∆ order 1, 2, … N
• Produces a high speed, non-weighted bit stream
Decimator
• Changes the high speed bit rate in low rate words
♦ ---- Channel -----
Interpolator
• Recreates the high speed serial flow
Σ−∆ DAC
• Rebuilds analog signal
Reconstruction filter
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 Bit rate reduction

Σ−∆ DECIMATOR
A

A D serial,
filtered High rate

D parallel,
Low rate

A’
INTERPOLATOR Σ−∆

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 D5: special A/D converters
Differential converters
♦Adaptive converters,
♦Sigma-delta converters
♦Oversampling, Noise shaping
Voice conversion, SNRq and dynamic range
Logarithmic conversion
♦Piecewise approximation,
♦A and μ laws
Waveform encoding and model encoding
♦Voice LPC
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 Voice signal conversion
Voice signal
♦exponential amplitude distribution
• more dense at lower levels
♦wide dynamic range
• SNRq low and variable with signal level
ADC can exploit these characteristics
♦Minimize noise for low-level signals
Logarithmic analog to digital conversion
♦constant SNRq over a wide signal dynamic range
♦fewer bits for the same SNRq
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 Linear and nonlinear A/D conversion
Linear A/D conversion
♦ all AD intervals have same amplitude
• quantization error does not depend on signal level
♦ poor results with signals at low levels for most time (voice)
• high quantization noise power, low signal power
Nonlinear A/D conversion
♦ different AD intervals
• quantization error changes with signal level
• the nonlinear relation can be chosen to optimize SNRq for a
specific signal type (PDF, amplitude distribution)
♦ for voice signals (exponential distribution)
• logarithmic law

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 Linear quantization
AD intervals with
constant width
Constant D
quantization noise
power
SNRq varies with
signal level
(worse for low-level A
signals)

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 Nonlinear quantization
AD intervals with
variable width
Quantization noise
power related D

with signal level

SNRq independent
from signal level
A

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 Standard conversion
The A/D conversion adds εq noise to analog signal
♦D = A + εq
♦AD is constant, therefore
• constant absolute error on D
• % error (SNRq) is related with signal level A

A D

εq

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 Logarithmic conversion
Conversion of signal logarithm:
♦D = log A + εq
♦sum of logs  log of product
• D = log A + εq = log K A (εq = log K)
• multiplying error (1 - K)
• constant % error, independent from signal level A

A log D

log A
εq
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 Nominal SNRq
Log quantization causes a constant relative error
♦constant SNRq
SNRq
log

lin Level

Full scale

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 A and µ approximation
♦Audio signals are bipolar
♦the log curve must be replicated in the III quadrant
• symmetric curve from I quadrant
Log 0 is undefined

near 0 the log curve can be only approximated

♦ µ law: translate the positive and negative
branches to get a continuous curve in (0,0)
♦A law: replace the curves near 0 with a straight
line (crossing 0,0)
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 A and μ approximation - graphs
Translation (μ law) Replacement (A law)

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 SNRq near 0
A law: signals less than 1/A --> linear quantization
♦SNRq depends on signal level (6 dB/octave)
μ law: almost linear quantization at low levels
♦similar effect: SNRq drop
SNRq

Level

1/A Full scale (S)

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 SNRq with A and μ law

μ law

A law

linear

Linear behavior Log behavior Overload

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 Log A/D approximation
Obtaining calibrated continuous nonlinear
behavior  complex & expensive analog circuits
Piecewise approximation
The log curve is divided in linear segments
♦due to log scale, the same ratio of input signal
corresponds to the same shift in horizontal axis
♦slope and starting point of each segment are sequenced
as 2 powers (2, 4, 8, 16, ….)
♦linear coding inside each segment

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 Piecewise approximation
Compressed signal

Continuous log law

slope

slope

slope

slope

Input signal

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 Log PCM format
Each sample is coded on 8 bit

♦MSB (bit 7): sign

♦bit 6, 5, 4: segment

♦bit 3, 2, 1, 0: level within the segment

7 6 5 4 3 2 1 0

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 Piecewise approximation: SNRq
Within each segment
♦ quantization error εq remains constant
♦ signal level changes  signal power changes
♦ SNRq changes with unity slope
From each segment to the next one (from S to 0)
♦ quantization error εq is divided by 2
♦ signal level is divided by 2
♦ SNRq constant
Near 0 same behavior as linear quantization
♦ constant εq
♦ signal level changes

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 SNRq with A and µ approximation

μ 255 law
A (87.6) law

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 Log conversion techniques
Analog log circuit, followed by A/D
♦poor precision and stability in the analog circuit
♦high cost
High resolution A/D conversion, followed by
digital log encoding
♦ makes available both the linear and log conversion
Intrinsic log A/D converter
♦nonlinear law A/D or D/A conversion
♦suitable for any type of nonlinear transfer function
(DAC for DDS)
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 A/D logarithmic converter
A logarithmic A/D converter can use the D/A feedback
technique: comparator-approximation logic - D/A loop
♦ the D/A must have exponential transfer function
How to build an exponential D/A (bipolar):
♦ sign bit: inverts the D/A reference voltage
♦ segment bits: provide a voltage with 2N steps
• segment bits are decoded into linear code (3-8 decoder)
• the 8 bit feed a linear 8-bit D/A
• each segment generates outputs with a ratio 2 towards
adjacent ones
♦ level bits: fed directly to a linear D/A

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 Logarithmic ADC
Sign bit  inverts the reference voltage Vr
Segment bits  voltage Vs scaled with 2N steps (1, 2, 4, 8, …)
Level bits  fed directly to a linear DAC using Vs as reference

Vs

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 Nonlinear DAC
Structure for nonlinear DAC and ADC with piecewise
approximation
♦ Segment bit decoder
♦ Standard DAC + lookup table
♦ Decoded DAC uniform elements
• To build starting point and slope of each segment
♦ Linear coding within each segment (level bits)
♦ Output adder
• Shifts the segment starting point
Technique used for DACs in DDS (sine generators)
♦ Sine conversion law (piecewise approximation)

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 Nonlinear DAC block diagram
Piecewise nonlinear characteristic
♦Da: segment bits
♦Db: level bits

Da Db

DECODER/LOOKUP

+Vr SEGMENT LEVEL VO

-Vr SLOPE DAC DAC
+

SEGMENT
START DAC

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 D5: special A/D converters
Differential converters
♦ Sigma-delta converters
♦ Oversampling
♦ Noise shaping
Logarithmic conversion
♦ Piecewise approximation
♦ A and μ laws
♦ Logarithmic converters
Waveform encoding and model encoding
♦ Voice LPC
♦ Comparison quality/bit rate

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 Model encoding vs waveform encoding
Waveform encoding:
♦ Sequence of number which represent the sequence of values
generated by sampling the time varying signal
♦ Example: sine tone
• Values of the sine signal at sampling times
Model encoding:
♦ Define a “source model”
♦ Model parameters are derived from the signal
♦ The signal is rebuilt from parameters using the model
♦ Example: sine tone
• Model: sine generator
• Parameters: amplitude, frequency, and phase
• Rebuilt using a properly set signal generator
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 Waveform encoding
Sequence of samples
♦example: sine tone
• Values of A sinωt for t = K Ts

Ts = 0,2 ms
10 V
2
1 t
(ms)

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 Model and parameters

Phase θ Period T

Peak
value V 2
t
1 [ms]

Model: v(t) = V sen (ω t + θ)

Parameters: V = 10 V
ω = 2πf = 2π/T = 5.2 krad/s
θ = 0,3π = 0,9 rad
6 decimal digits

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 SNR for model encoding
Which factors influence SNR?

Waveform encoding
♦Sampling rate
♦Resolution of samples (bit number)

Model encoding
♦Model accuracy
♦Correctness and resolution of parameters

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 Example of model encoding
LPC (Linear Predictive Coding) for voice signals
♦Based on a vocal segment model (larinx)
♦Signal is divided in frames (10-30 ms)
♦For each frame:
• voiced/unvoiced decision
• evaluation periodicity step (pitch)
• Evaluation of adapted filter coefficients
♦Voiced: complex waveforms repeated
• Pulse generator at pitch rate
• Filter to generate the waveform
♦Unvoiced: filtered noise
• Noise generator + filter
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 Block diagram of LPC decoder

PITCH

FILTER
PARAMETERS
PULSE
GENERATOR VOICED

FILTER

UNVOICED
NOISE
GENERATOR

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 Model encoding: performance
Standard criteria:
♦speaker recognition (A)
♦speech understanding (B)
Waveform encoding speed (kbit/s)
♦log PCM 64/32
♦Differential 32/16
♦Adaptive Differential (ADPCM) 4 (only B)
Model encoding
♦LPT (GSM phones) 9,6
♦Frequency slots vocoder 4,8
♦LPC 2,4
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