Digital Signal Processing Methodologies
Digital Signal Processing Methodologies
DOI: 10.5923/j.ajbe.20130301.03
1
Biomedical Engineering Department, Sudan University of Science & Technology, Khartoum, Sudan
2
Systems & Biomedical Engineering Department, Cairo University, Giza, Egypt
Abstract Ultrasound imaging is an efficient, noninvasive, method for med ical diagnosis. A commonly used approach
to image acquisition in ultrasound system is digital beamforming. Digital beamforming, as applied to the medical
ultrasound, is defined as phase align ment and summation of signals that are generated fro m a co mmon source, by received
at different times by a mu lti-elements ultrasound transducer. In this paper first: we tested all signal processing
methodologies for digital beamforming which included: the effect of over samp ling techniques, single trans mit focusing
and their limitations, the apodization technique and its effect to reduce the sidelobes, the analytical envelope detection
using digital finite impulse response (FIR) filter appro ximations for the Hilbert transformat ion and how to co mpress the
dynamic range to achieve the desired dynamic range for display (8 bits). Here the image was reconstructed using physical
array elements and virtual array elements for linear and phase array probe. The results shown that virtual array elements
were given well results in linear array image reconstruction than physical array elements, because it provides additional
number of lines. Ho wever, physical array elements shown a good results in linear phase array reconstruction (steering) than
virtual array elements, because the active elements number (Aperture) is less than in physical array elements. We checked
the quality of the image using quantitative entropy. Second: a modular FPGA-based 16 channel digital u ltrasound
beamforming with embedded DSP for ultrasound imaging is presented. The system is imp lemented in Virtex-5 FPGA
(Xilin x, Inc.). The system consists of: t wo 8 channels block, the DSP wh ich co mposed of the FIR Hilbert filter b lock to
obtain the quadrature components, the fractional delay filter block (in-phase filter) to co mpensate the delay when we were
used a high FIR order, and the envelope detection block to compute the envelope of the in-phase and quadrature
components. The Hilbert filter is imp lemented in the form whereby the zero tap coefficients were not computed and
therefore an order L filter used only L/2 mu ltiplications. This reduced the computational time by a half. Fro m the
implementation result the total estimated power consumption equals 4732.87 mW and the device utilization was acceptable.
It is possible for the system to accept other devices for further processing. Also it is possible to build 16-,32-, and
64-channel beamformer. The hardware architecture of the design provided flexib ility for beamforming.
Keywords Medical Ult rasound, Dig ital Beamforming, FIR Hilbert Transform Filter, FPGA, Embedded DSP
point within the body, from which location of the returning 2.1. Over Sampling Techni que
echo is derived. Over samp ling is used to achieve high delay resolution.
A commonly used approach to image acquisition in However, this increase the data volu me has to acquire. This
ultrasound system is digital beamforming because the analog is usually avoided by sampling just above the Nyquist rate
delay lines impose significant limitations on beamformer and interpolating to achieve the required delay resolution.
performance and more expensive than digital implementatio Radial sampling resolution is a relat ionship between the
ns. Dig ital beamforming, as applied to the medical depth and the number of delay values and it equals to the
ultrasound, is defined as phase align ment and summat ion[2] speed of sound over twice the samp ling frequency. Fro m the
of signals that are generated from a co mmon source, by literature[7], wide band transducer required delay resolution
received at different times by a mu lti-elements ultrasound in order of 1/ 16 the signal period.
transducer[6].
The beamforming process needs a high delay resolution to 2.2. Delay Equati on
avoid the deteriorating effects of the delay quantization lobes
RF signal N
on the image dynamic range and signal to noise ratio
RF signal 3
RF signal 2
RF signal 1
(SNR)[7]. If oversampling is used to achieve this timing
resolution[8], a huge data volu me has to be acquired and
process in real time. This is usually avoided by sampling
just above the Nyquist rate and interpolating to achieve the
Delay Stage
Over Sampling Technique
required delay resolution[7].
Beamforming required Apodization weighted to
decreasing the relative excitation near the edges of the
Single Transmit Focusing
radiating surface of the transducer during transmit or
receiving, in order to reduce side lobes. Just as a time side
lobes in a pulse can appear to be false echoes.
After delay and sum the envelope of the signals is detected. The Apodization Technique
The envelope then compressed logarithmically to reduce the
dynamic range because; the maximu m dynamic range of the
human eye is in the order of 30 dB[9]. The actual dynamic Physical Array Virtual Array
range of the received signal depends on the ADC bits, the Elements Elements
Summation Stage
time gain co mpensation (TGC) amp lifier used in the front
end, and the depth of penetration. The signal is compressed
to fit the dynamic range used for display (usually 7 or 8 b its). Linear Array Linear Phase
It is typical to use a log co mpressor to achieve the desired Reconstruction Array
dynamic range for display[10]. Reconstruction
With the growing availability of high-end integrated
analog front-end circu its, distinction between different FIR Hilbert Transform
digital ult rasound imag ing systems is determined almost Filter Design
exclusively by their software co mponent.
We develop a compact low-cost digital ultrasound
imaging system that has almost all of its processing steps Real Quadrature
done on the PC side. In this paper first: we tested all signal Components Components
processing methodologies for dig ital beamforming which
DSP Stage
transducer array. After a wave-front is trans mit into the summed together after phase shift and some signal
med iu m an echo wave propagates back from the focal points conditioning to produce a single output. This reconstruction
(P) to the transducer. The distance from P to the origin is technique divides the field of view (FOV) into different point
equal to the Euclid ian distance between the spatial point targets (raster points), P(i,j).
(xi ,y i ) ( is the center for the physical element number i) and Each point represented as an image p ixel, which is
the focal point (xf,y f) (is the position of the focal point). The separated laterally and axially by small d istances. Each
original time, t i, , equal the distance from P over the speed of target is considered as a point source that transmits signals to
sound. A point is selected on the whole aperture (AP) as a the aperture elements as in figure 4. The beamfo rming timing
reference (xc,y c) for the imag ing process. The propagation is then calculated for each point based on the distance R
time (tc) for this was calculated as the above, but the distance between the point and the receiving element, and the velocity
here fro m P to the reference (xc ,y c). The delay to use on each of ultrasonic beam in the media. Then the samples
element of the array is the difference between t c and ti . corresponding to the focal point are synchronized and added
to complete the beamforming as the follo wing:
y (axial) PD (i, j) = ∑Nn=1 Xn �K ij �, (1)
(xc ,yc) (xi ,yi ) where PD (i,j) is the signal value at the point whose its
coordinates are (i,j), and Xn (Kij ) is the sample
x (Lateral)
corresponding to the target point in the signal Xn received
by the element number n. The sample number Kij which is
equivalent to the time delay is calculated using the equation
below:
P (x ,y )
f f
R n (i,j )
K ij = . (2)
T∗c
Scan Line Scan Line Here Rn (i, j) is the distance from the center of the element
to the point target, c is the acoustic velocity via the media,
Figure 2. Geometry of a focused transducer array and T is the sampling period of the signal data.
2.3. physical Array Elements y
128/64. Then one elements shift was applied to the virtual 2 sin 2 (π (n −α) ⁄2)
,n ≠ α
and physical aperture and this process is repeated till the h[ n] = �π n −α , (4)
factor equal 128/128. 0 ,n = α
64 virtual elements 64 physical elements α = (N − 1)/2 . We chose the filter length equal
16,20,24,28 and 32-tap with a Hamming window used to
reduce the sidelobe effects. According to the normalized
1 2 … 63 64 1 2 … 63 64 … 128 root mean square error (RMSE) between the designed FIR
Hilbert filter and ideal Hilbert transform filter. The values
of the RMSE fo r the five FIR filters are shown in table 1.
Center element # 1 for image line #1
We selected 24-tap FIR filter because it provided a good
result for the qadrature components compared to ideal
Figure 5. Linear virtual array elements
Hilbert transform filter.
2.4.2. Linear Phase Array Reconstruction
Table 1. The Normalized RMSR for FIR Hilbert filters
We used the same techniques as in section (2.3.2).
FIR Hilbert filter order Normalized RMSE value
2.5. Apodizati on
16 0.0109
Apodization is amplitude weighting of normal velocity 20 0.0096
across the aperture[9][11], one of the main reasons for
24 0.0092
apodization is to lo wer the side lobes on either side of the
28 0.0091
main beam[12]. Just as a time side lobes in a pulse can
appear to be false echoes[9]. Aperture function needed to 32 0.0090
have rounded edges that taper toward zero at the ends of the
aperture to create low side lobes levels. We used windowing 2.6.2. Co mpressed the Dynamic Range
functions (hamming, Blackman, and Kaiser (β=4)) as The envelope then compressed logarithmically to ach ieve
apodization functions to reduce the side lobes. There is the desired dynamic range for d isplay (8 bits). It is typical to
trade-off in selecting these functions: the main lobe of the use a log compressor to achieve the desired dynamic range
beam broadens as the side lobes lower[9]. for d isplay. Log transformation co mpressed the dynamic
range with a large variation in pixels values[18].
2.6. Envel ope Detection and Compressed The Dynamic
Range 2.7. Implementation Steps
2.6.1. FIR Hilbert Transform Filter Design
After delay and sum, the analytic envelope of the signal is
calculated as the square root of the sum of the squares of the
real and quadrature components[13]. The most accurate way
of obtaining the quadrature components was to pass the echo
signal through a Hilbert transform[14], because it provides
90-degree phase shift at all frequencies[15].
The Hilbert transformation filter acts like an ideal filter
that removes all the negative frequencies and leaves all
positive frequencies untouched. A number of authors
suggested the use of digital FIR filter appro ximations to
implement the Hilbert transformat ion. For linear time
invariant (LTI) a FIR filter can be described in this form[16]:
𝑦𝑦 [ n] = 𝑏𝑏0 𝑥𝑥[ 𝑛𝑛] + 𝑏𝑏1 𝑥𝑥[ 𝑛𝑛 − 1] + ⋯ + 𝑏𝑏𝑀𝑀 𝑥𝑥[ 𝑛𝑛 − 𝑀𝑀]
𝑀𝑀
= �𝑖𝑖=0 𝑏𝑏𝑖𝑖 𝑥𝑥[𝑛𝑛 − 𝑖𝑖] . (3)
Where 𝑥𝑥 is the input signal, 𝑦𝑦 is the output signal, and the
constants 𝑏𝑏𝑖𝑖 , 𝑖𝑖 = 0,1,2,…,M, are the coefficients. The
designed FIR Hilbert filter can be used to generate the
Hilbert transformed data of the received echo signal. The
impulse response of the Hilbert filter with length N (odd Figure 6. Architecture implementation of the modular FPGA-Based 16-
channel digital ultrasound receive beamformer blocks
number) is defined as[17]:
18 M awia Ahmed Hassan et al.: Digital Signal Processing M ethodologies for
Conventional Digital M edical Ultrasound Imaging System
(a)
(b)
Figure 7. The inside contents of the implementation blocks. (a) The 8 channel block, (b) the reconstructed line block
A typical architecture imp lementation of the modular Xilin x block to read the one d imension RF data fro m
FPGA -based, 16-channel digital u ltrasound receive workspace.
beamformer with embedded DSP fo r ult rasound imaging 2. The RF data then convert the double precision data type
was shown in figure 6. The system consist of: Two 8 to fixed po int numeric precision for hard ware efficiency.
channels block and reconstructed line b lock. The 3. Verified the fixed-point Model by co mpared the
beamfo mer is done by using Xilin x system generator fixed-point results to the floating-point results and determine
(Xilin x, Inc.) and MATLAB simu lin k (MathWorks, Inc.). if the quantizat ion error is acceptable.
The system is imp lemented in Virtex-5 FPGA. The 4. A fter verified the model, each channel data saved in
implementation steps are (Figure 7): memo ry b lock. The memory word size is determined by the
1. The RF data saved in MATLA B workspace and used bit width of the channel data. The memo ry controlled by wire
American Journal of Biomedical Engineering 2013, 3(1): 14-30 19
enable port with 1 indicates that the value of the channel data channels was 128 channels, and the A/D sampling rate was
should be written to the memory address pointed to by 13.8889 MSPS. Linear shape transducer was used to acquire
step-up counter. the data with center frequency of 3.5 M Hz, and element
5. The delay process is based on sampled delay focusing spacing of 0.22mm. Each u ltrasonic A-scan was saved in a
(SDF). The delays calculated using the same method in record consisted of 2048 RF samp les per line each
section 2.2. Then the delays converted to number of samp les represented in 2 bytes for the phantom data and 4 byte for the
by divided the delays by the sampling time. SDF consist of real data, and the signal averages was 8. The speed of the
addressable shift register (ASR) to delay the sampled signals, ultrasound was 1480 m/sec. The data were acquired for
and M-Code Xilin x block that contain the calculated delays. phantom within 6 p ins at different positions. The data was
The samples are delayed by the value in the address input of used to simulate the N-channel beamformer on receive as
the ASR. discussed in methodologies. The radio frequency (RF)
6. After delay ing each RF channel samples, the signals, A-scan, were recorded fro m every possible
summation is applied using M-Code block to summate the 8 combinations of transmitter and receiver for all elements in
channel signals. the 128 elements.
7. The summat ion of the two 8 channels is connected to
adder to reconstructed the final focus line. 3.2. Over Sampling Techni que
8. Modify the bit of the signal to 16 bit using the bit Table 2 was shown the effect of oversampling to the delay
modifier b lock. resolution. As can be shown when sampling ratio increased
9. The Reg ister block (data presented at the input will the delay resolution decreased according to the radial
appear at the output after one sample period). sampling resolution. Fro m the table the Delay resolution
10. We used the FIR Hilbert filter (with 24-tap as in equal to (1/(2×sampling ratio )).
section 2.6.1) b lock for applying the quadrature Fro m the table when the sampling ratio (F) equal 8 as an
components. interpolation factor, it gave better delay resolution (1/16)
11. The Fractional delay filter (in-phase filter) to the signal period. However, this increase the data volume
compensate the delay when we are being used a high FIR has to acquire.
order.
12. Then we mod ified the bit of the signals from step 10 3.3. Physical Elements Array
and 11 to 16 bit again using the bit modifier blocks.
13. The Envelope detection block wh ich was co mputed 3.1.1. Linear Array Image Reconstruction
the envelope of the two signals coming fro m step 10 and 11. In this situation we reconstructed an image using
14. In order to obtain performance and logic utilizat ion N-channel beamformer on receive where N= 4,8,16,32 and
figures for the suggestion architecture, it was implemented 64. Figure 8 was shown images reconstructed for six p in
in the hardware description language (VHDL) and phantom by linear technique (figure3). In figure 8 (b,c,d and
synthesized with Virtex-5 FPGA . e) the images was reconstructed with focusing using AP
equal 4,8,16,32and 64. In figure 8 (a) image was
reconstructed without focusing and AP=4 elements. When
3. Results and Discussions the AP size increased the lateral resolution was improved.
However, the FOV was reduced when the aperture size
3.1. The Ultrasound Data increased. Figure 9 shown the signal to noise ratio (SNR)
We used correct data obtained from the Bio med ical when aperture equal 4,8,16,32,and 64. As can be shown
Ultrasound Laboratory[19], University of Michigan; the when the size of A P increased that improved the SNR. A lso
phantom data set that was used to generate the results here is figure 10 described the effect of oversamp ling technique.
under "Acusonl7" and real data (cyst4_a100). The
parameters for this data set are as follo ws: the number of
Table 2. The effect of the oversampling to the delay resolution
Sampling Radial sampling resolution(=Ts/2×c)
Delay resolution
Sampling fre quency(Fs) period Delta-distance (Dd)
(1/(2×sampling ratio))
(Ts=1/Fs) c=1.480mm/µsec
Fs=13.8889
Ts=72 nsec Dd=0.0533 mm/samples 1/2
Sampling Ratio=1
Fs=2×13.8889=27.7778
Ts=36 nsec Dd=0.0267 mm/samples 1/4
Sampling Ratio=2
Fs=4×13.8889=15.5556
Ts=18 nsec Dd=0.0133 mm/samples 1/8
Sampling Ratio=4
Fs=8×13.8889=111.1112
Ts=9 nsec Dd=0.0067 mm/samples 1/16
Sampling Ratio=8
20 M awia Ahmed Hassan et al.: Digital Signal Processing M ethodologies for
Conventional Digital M edical Ultrasound Imaging System
(a)
(b)
Figure 10. The effect of oversampling technique to the SNR. Here we
used image with AP equal to 64 with F equal to 1,2,4,and 8
(e)
(a)
(f)
Figure 8. Physical Linear array image reconstruction, (a) Using AP=4 (b)
without focusing, (b) Using AP= 4 with focusing, (c) Using AP= 8with Figure 11. Physical Linear phase array image reconstruction, element
focusing, (d) Using AP= 16 with focusing, (e) Using AP= 32 with number 128 for transmitted and received with all 128 elements, (a) Using
focusing, (f) Using AP=64 with focusing F=4, (b) Using F=8
American Journal of Biomedical Engineering 2013, 3(1): 14-30 21
(a)
(a)
(b)
Figure 12. Physical Linear phase array image reconstruction, transmitted
and received with all 128 elements (a) Using F=4, (b) Using F=8
(b)
3.4. Virtual Array Elements
(a)
Figure 15. SNR for AP equal 16,32,and 64 using raster point technique for
virtual elements
(a)
Figure 17. The frequency response of apply 16-, 20-, 24-, 28-, and 32-tap
FIR Hilbert filter
(b)
Figure 18. The different between the FIR Hilbert filters and the ideal
Hilbert filte (c)
(a)
(a)
(b)
(b)
(c)
Figure 22. The compressed envelope images used log compression
(transmitted and received with all elements) (a) Compressed envelope
(c) image apodized with Hamming , (b) Compressed envelope image apodize
Figure 20. The envelope images used raster point technique (There were d with Blackman , (c) Compressed envelope image apodized with Kaiser
128 elements for transmits and receives) (a) Envelope image apodized with (β=4)
Hamming , (b) Envelope image apodized with Blackman , (c) Envelope
image apodized with Kaiser (β=4)
Figure 21. The analytical envelope Figure 23. The dynamic range without compression
American Journal of Biomedical Engineering 2013, 3(1): 14-30 25
(e)
(f)
(a)
(g)
Figure 25. Pin three in the phantom as a sub-image. (a) Without
apodization,(b) Apodized with Hamming , (c) Apodized with with
Blackman , (d) Apodized with Kaiser (β=4), (e) Frequency spectrum of
fig.24 (b), (f) Frequency spectrum of fig.24 (c), (e) Frequency spectrum of
fig.24 (d)
(b)
Figure 26. Image reconstructed using raster point technique for real data
(c) (Cyst)
Figure 29. Comparison between implemented the first line (after the pipe
line adder in Fig.6(b)) in image reconstruction and the simulated one
(a)
(a)
(b)
(b)
(c)
(d)
(e)
(d)
Figure 30. Hilbert filter apply to ultrasound line data. (a) Reconstructed
FIR Hilbert, (b) Ideal Hilbert, (c) the different between the reconstructed Figure 31. Image reconstruction and the PSF of the first pin. (a) Image
and ideal Hilbert, , (d) frequency spectrum of reconstructed FIR Hilbert,(e) after delay and sum, (b) PSF of the first pin in (a), (c) Image after applying
The ideal Hilbert filter frequency spectrum FIR Hilbert filter and envelope detection, (d) PSF of the first pin in (c)
28 M awia Ahmed Hassan et al.: Digital Signal Processing M ethodologies for
Conventional Digital M edical Ultrasound Imaging System
Slice Logic Utilization Used Available Utilization Slice Logic Utilization Used Available Utilization
Number of Slice
1,875 207,360 1% Number of route-thrus 29 - -
Registers
Number used as Flip Number using O6 output
1.875 - - 22 - -
Flops only
Number using O5 output
Number of Slice LUT s 2,283 207,360 1% 7 - -
only
Number used as logic 627 207,360 1% Number of occupied Slices 819 51,840 1%
Number using O6 Number of LUT Flip Flop
885 - - 2,473 - -
output only pairs used
Number using O5 Number with an unused Flip
19 - - 598 2,473 24%
output only Flop
Number using O5 and Number with an unused
23 - - 190 2,473 7%
O6 LUT
Number used as Number of fully used
1,653 54,720 1% 1,685 2,473 68%
Memory LUT-FF pairs
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