A Multi-stationary Computing-in-memory Processor with Scalable-precision ADC for Keyword
Spotting from Raw Audio Signal
DongwookKim1
Kyuho JasonLee2,*
-
Department of Electrical Engineering, Ulsan National Institute of Science and Technology,
Ulsan, Korea
-
School of Electrical and Electronic Engineering, Yonsei University, Seoul, Korea
Copyright © The Institute of Electronics and Information Engineers(IEIE)
Index terms
Keyword spotting (KWS), embedded DRAM (eDRAM), computing-in-memory (CIM), analog computing
I. INTRODUCTION
Recently, keyword spotting (KWS) has emerged as a key technology for convenient voice
interfaces in smart edge devices [1,
2]. As KWS must operate continuously in an always-on state, it imposes a stringent requirement
for low-energy processing to be deployed in edge devices [3-
7]. Computing-in-memory (CIM), a promising solution for energy-efficient artificial
intelligence (AI) computation, is particularly well suited for KWS systems [8-
11].
Fig. 1. Challenges in keyword spotting processing.
CIM achieves low energy consumption by integrating computation units into memory,
which reduces frequent external memory access. Nonetheless, prior CIM processors have
exposed their limitations in KWS processing.
Fig. 1 highlights the challenges in KWS processing. KWS models operate by applying convolutional
neural networks (CNNs) to voice signals to extract temporal features. In these models,
voice signal preprocessing entails complex computations, such as the fast Fourier
transform. Prior works [12,
13] adopted sinc convolution, which directly decomposes the 1-D signal into distinct
frequency bands without MFCC-based feature extraction. However, these approaches still
face some significant limitations to be applied to KWS processors. First, the Nyquist
criterion requires a 16-kHz sampling rate for human voice recognition [14]. As a result, convolution directly on the raw audio produces a significantly large
input feature map (IFM), leading to substantial computational cost and inflated model
complexity. Second, prior CIM processors demonstrated notably low hardware utilization
in depthwise convolution (DWCONV), as the inherent column-wise activation sharing
prevents effective parallel execution [15,
16]. Since temporal feature extraction from frequency-band signals requires depthwise
separable convolution (DSCONV) after the sinc convolution [12], this operation leads to a significant drop in the utilization of CIM processors.
Lastly, analog-to-digital converters (ADCs) remain a major energy bottleneck in CIM
processors, further limiting their efficiency for always-on KWS applications [17-
20].
This paper highlights an eDRAM CIM processor for KWS achieving high integration density
and low decision energy, enabled by three key features to address the challenges above:
(1) frequency-selective stride convolution (FSSC) for compact acoustic feature extraction;
(2) grouped depthwise convolution (GDCONV) to maximize CIM utilization; and (3) scalable-precision
analog-to-digital converter (SP-ADC) for low-energy CIM computation.
The rest of this paper is constructed as follows. Section II introduces the proposed
energy-efficient KWS CIM processor. The overall architecture of the processor, including
the raw audio processing algorithm, is presented in this Section. Implementation and
evaluation results are shown in Section III, followed by a conclusion in Section IV.
II. PROPOSED KWS PROCESSOR
1. Overall Architecture
Fig. 2(a) shows the overall architecture of the proposed KWS processor. It consists of two
CIM clusters, and each cluster integrates eight CIM cores. The processor also includes
a 16 KB input activation (IA) buffer, a weight buffer, a top controller, and a non-linear
activation single instruction multiple data (SIMD) core, all communicating through
an interconnect network. The SIMD core performs activation and batch normalization
between convolution layers. The CIM core consists of a $144\times 144$ 3T2C eDRAM
CIM array. Among the 144 rows, 16 rows are allocated to operate as the capacitive
digital-to-analog converters (CAP-DACs) used in the SP-ADCs, while 16 columns are
reserved for sign-bit computation [21,
22], where the MACs are shifted positive and offset-canceled to restore the signed result.
It also integrates a bitline (BL) and wordline (WL) driver, an ADC/core controller,
DW/PW switch, and a digital adder for partial sum accumulation. Fig. 2(b) shows the schematic of the 3T2C eDRAM bitcell. The access transistor P1 employs an
HVT-PMOS to minimize the leakage current while allowing the storage node (SN) to be
charged up to VDD from the write bitline (WBL). In addition, the read bitline (RBL)
is driven through charge coupling according to the read wordline (RWL) signal to generate
the output voltage. Fig. 2(c) shows the operand table of the 3T2C bitcell. During computation, the RBL is pre-discharged
to GND. Then, the RWL drives the RBL voltage rise through the activated N0 when SN
stores 1, executing the analog AND operation.
Fig. 2. (a) Overall architecture of the proposed KWS processor, (b) schematic, and
(c) operand table of the 3T2C eDRAM bitcell.
2. Frequency-selective Stride Feature Extraction
Fig. 3(a) depicts the operation of the FSSC. A one-second, one-dimensional raw audio signal
consisting of 16,384 samples is fed into a sinc convolution layer. The signal is decomposed
into amplitudes corresponding to different frequency bands for subsequent feature
extraction in this layer. In the low-frequency domain, the signal varies slowly over
time. Therefore, even with a large stride in convolution, the resulting accuracy degradation
remains negligible. On this basis, the FSSC selectively applies different stride sizes
to each frequency band to minimize the size of the input feature maps in the convolution
layer. The stride sizes that minimize accuracy degradation were determined as hyperparameters,
based on the kernels of the trained sinc convolution layer. The model is then fine-tuned
with the selected strides to minimize the accuracy loss.
Fig. 3. (a) Frequency-selective stride convolution, (b) proposed kernel fusion architecture,
and (c) performance comparison.
Although FSSC reduces IFM size, varying strides cause non-uniform IFM lengths that
complicate parallel processing. To address this, a kernel fusion architecture based
on structural re-parameterization is introduced, as shown in Fig. 3(b). During offline training, consecutive linear convolutions are mathematically collapsed
into a single equivalent kernel. By applying this fused kernel directly to the raw
input using the maximum stride, the network bypasses intermediate irregular feature
maps. Consequently, it directly outputs uniform IFMs, which enables hardware-friendly
parallel processing without additional computational overhead.
Fig. 3(c) illustrates the performance improvement achieved by the FSSC. As the stride size
increases in the sinc convolution layer, both the input IFM size and the accuracy
decrease, exhibiting a trade-off between computational efficiency and performance.
Nevertheless, when the stride is increased up to 128, the accuracy drop remains as
small as 0.8%, while the IFM size is reduced by 99.22%. As a result, the computational
load of the subsequent DSCONV layers becomes significantly smaller, effectively addressing
the challenges of raw audio processing.
3. Multi-stationary CIM Architecture
Fig. 4(a) depicts the throughput enhancement in the DWCONV layer enabled by the proposed multi-stationary
CIM architecture. In conventional CIM architectures, DWCONV utilizes only a single
memory column at a time, since each channel employs distinct weights. Fig. 4(b) introduces GDCONV to maximize DWCONV computation. The proposed architecture maximizes
weight reuse and CIM utilization by training DWCONV to share identical weights across
multiple channels. In particular, the frequency-domain features obtained via sinc
convolution exhibit high similarity across adjacent channels, which enables weight
sharing between consecutive channels while minimizing accuracy loss. To further improve
CIM utilization with this model, the processor adopts IA stationary mapping in DWCONV.
All IAs sharing identical kernels are mapped to a single CIM macro. The corresponding
weight is broadcast via WL, allowing simultaneous computation across all columns.
Fig. 4. (a) Multi-stationary CIM architecture, (b) operation of the GDCONV computation,
and (c) performance improvements in DWCONV layers.
Fig. 4(c) illustrates the performance improvement achieved by the multi-stationary CIM architecture.
GDCONV achieves weight sharing across 16 channels without reducing the channel dimension,
which constrains the accuracy loss to less than 2%. Furthermore, the IA-stationary
architecture results in a $16\times$ improvement in DWCONV throughput. Consequently,
GDCONV combined with the IA-stationary architecture leads to a $1.47\times$ improvement
in the KWS system throughput.
4. Layer Fusion for Minimized Memory Footprint
Fig. 5 depicts the proposed layer fusion architecture, which is designed to minimize the
memory footprint. In this work, the DSCONV network, following the sinc convolution,
has an IFM size of 128 points $\times$ 128 channels $\times$ 8 bits in DWCONV, and
a weight size of 128 input channels (IC) $\times$ 128 output channels (OC) $\times$
8 bits in PWCONV. Therefore, each CIM cluster is designed to provide full on-chip
storage for either the IFM or the weights, thereby eliminating the external memory
accesses. The PWCONV outputs are directly committed into the CIM array of the next
cluster through the same memory write path, without any column-level buffering or
data reshaping stalls. As a result, the transfer overhead between PWCONV and DWCONV
is substantially reduced. On the other hand, the DWCONV results are stored in the
IA buffer, whereas the other CIM cluster simultaneously updates its weights, thereby
enabling seamless pipelined execution. Finally, by applying layer fusion between the
DWCONV and PWCONV layers, the memory footprint is reduced by 47.5% compared to the
identical design without layer fusion.
Fig. 5. Proposed layer fusion CIM architecture.
5. Scalable-Precision ADC
Fig. 6(a) shows the hardware structure of the proposed SP-ADC. It consists of a CAP-DAC, a
sense amplifier (SA), a 7-bit register, an ADC controller, and a voltage integrator.
The SP-ADC incorporates distinct CAP-DACs dedicated to the conversion of the most
significant bits (MSBs) and the least significant bits (LSBs), respectively. The MSB
DAC is implemented with 48C (32C + 16C) to convert the two MSB bits, while the LSB
DAC employs 16C capacitors to convert the remaining bits. The MSB DAC is charged by
a voltage integrator circuit shared within the CIM core. The LSB DAC consists of sixteen
dummy bitcells to minimize area overhead. It is also incorporated into the reference
generation path, providing inherent noise calibration. The SP-ADC operates in a successive
approximation register (SAR) architecture. It sequentially reconstructs the multibit
multiply-and-accumulate (MAC) results from the analog voltage.
Fig. 6. (a) Hardware structure, (b) operational details, and (c) energy reduction
of the SP-ADC.
Fig. 6(b) illustrates the operational sequence of the proposed SP-ADC. After the MAC operation
in the first cycle, the sign bit is detected, and the subsequent three cycles are
dedicated to converting the two MSBs. Instead of rail-to-rail switching of the large
MSB capacitor, the SP-ADC employs an integrator-based ramping scheme. A shared voltage
integrator generates a stepwise ramp voltage that accumulates charge equivalent to
48C over three steps, which is distributed to each column via local current mirrors.
The current mirror is disabled once the target range is determined by the comparator,
thereby halting further charging of the MSB DAC. In inlier cases with a limited output
range, energy consumption is reduced by limiting the number of comparator evaluations
and restricting switching activity to the small LSB capacitances. Even for outlier
cases, conversion energy is reduced by early termination of the MSB ramp, leveraging
the output distribution of neural networks [22]. After the MSB conversion phase, the SAR controller activates the LSB DAC to convert
the remaining LSBs.
Fig. 6(c) shows the energy reduction of the SP-ADC. The integrator-based charging circuit reduces
capacitor-driving energy, which otherwise increases with bit precision in conventional
SAR ADCs. In particular, for outlier cases where the operation range exceeds $\pm
16$, the proposed SP-ADC reduces the average energy consumption of the CIM macro by
27.12% compared to the baseline [8]. For inlier cases within the $\pm 16$ range, the MSB-DAC switching energy is fully
eliminated, resulting in a total energy reduction of up to 75.71%. In conclusion,
the proposed SP-ADC effectively reduces the ADC conversion energy for KWS processing
by exploiting the output data distribution inherent to KWS models.
III. MEASUREMENT RESULTS
1. Comparison with Conventional Structures
Fig. 7 shows the chip layout and hardware performance specification of the proposed processor.
The processor is designed and verified via post-layout simulations in Cadence Virtuoso
and Innovus with a 28 nm CMOS process. The 3T2C eDRAM achieves a retention time of
70.1 $\mu$s, utilizing a 0.61 fF metal–oxide–metal capacitor stacked across the M3–M7
layers with a 0.46 $\mu\text{m}^2$. It consumes 117.2 $\mu$W of maximum power consumption
at 1.0 V operating voltage and 100 MHz frequency. It achieves 34.8 TOPS/W of macro
energy efficiency and 21.6 TOPS/W of system energy efficiency. The processor achieves
95.1% accuracy on the 10 keywords of the Google Speech Commands dataset [23] with unknown and silence. It also achieves 98.4% accuracy on the Hey Snips [24] wake word detection task and 91.3% accuracy on the UrbanSound8K dataset [25]. As a result, the proposed processor can operate robustly across various acoustic
signal processing tasks.
Fig. 7. Chip layout, performance specification, and benchmark results of the proposed
processor.
Fig. 8. Simulation results of (a) MAC linearity under Monte Carlo simulation, (b)
performance specification of the SP-ADC.
Fig. 8(a) illustrates the MAC linearity of the proposed CIM processor based on post-layout
simulations. To evaluate the impact of process variations, including those in the
3T2C eDRAM bitcells, post-layout Monte Carlo simulations were conducted. The results
show a 2-mV standard deviation in the MAC voltage, ensuring stable computation across
128 levels. Fig. 8(b) presents the integral nonlinearity (INL) and differential nonlinearity (DNL) of the
SP-ADC. Extracted from post-layout simulations at a 100-MHz clock frequency, the maximum
INL and DNL are 0.38 LSB and 0.26 LSB, respectively. Furthermore, the SP-ADC achieves
a 41.6-dB SNR and a 6.62-bit ENOB. As a result, the proposed CIM processor ensures
linear MAC operations and accurate data conversion across various practical operating
conditions.
Fig. 9. Normalized performance improvements and evaluated accuracy.
Fig. 9 compares the proposed design to a baseline that consists only of the same CIM core
count, without the proposed features. As the feature map size is significantly reduced
by the FSSC, the energy consumption and latency are reduced by 88% each. Furthermore,
according to the GDCONV and the SP-ADC, energy consumption and latency are reduced
by 99% and 95%, respectively, compared to the baseline without any optimization. On
the other hand, the processor shows a 2.3% accuracy loss due to the trade-off between
accuracy and hardware performance. In conclusion, the proposed processor significantly
reduces the energy consumption and latency of KWS processing while minimizing the
accuracy loss.
Table 1 summarizes the performance comparison with prior KWS processors. The proposed design
achieves 29.6% lower inference energy than the NPU-based KWS processor in [4] and demonstrates a $1.85\times$ improvement in energy efficiency compared to the
CIM-based architecture in [6]. Furthermore, by leveraging a sinc convolution network [12] to support direct feature extraction from raw audio signals, it attains the highest
accuracy on the Google Speech Commands dataset [23] among the referenced works. As a result, the proposed processor proves its effectiveness
for ultra-low-energy KWS operations.
Table 1. Performance comparisons with KWS processors.
|
|
JSSC'25 [3]
|
JSSC'24 [4]
|
JSSC'22 [5]
|
TETC'24 [6]
|
SOVC'19 [7]
|
This Work
|
|
Technology
|
28 nm
|
28 nm
|
65 nm
|
28 nm
|
65 nm
|
28 nm
|
|
HW Type
|
|
NPU
|
|
|
CIM
|
|
|
Bit Precision
|
4-b
|
8-b
|
8-b
|
8-b
|
1-b
|
8-b
|
|
Dataset
|
Google Speech Commands
|
|
*# of Keywords
|
12
|
7
|
12
|
12
|
10
|
12
|
|
Accuracy [%]
|
91.8
|
92.8
|
86.0
|
94.4
|
90.2
|
95.1
|
|
Area [mm2]
|
8
|
0.8
|
2.03
|
3.56
|
6.2
|
0.325
|
|
Energy Efficiency [TOPS/W]
|
N/A
|
N/A
|
N/A
|
5.4
|
11.7
|
21.6
|
|
Energy / Inference [$\mu$J / Decision]
|
1.73
|
0.88
|
17.40
|
1.65
|
3.36
|
0.62
|
IV. CONCLUSIONS
In conclusion, an ultra-low-energy end-to-end KWS processor is proposed with three
key features: 1) The FSSC algorithm reduces the length of the raw audio input by 99.22%,
thereby minimizing memory usage and computational complexity. 2) The multi-stationary
CIM architecture and GDCONV maximize hardware utilization in the DWCONV layer and
reduce the number of WL activations, thereby decreasing the MAC energy by 95.26%.
3) The SP-ADC optimizes the CAP-DAC switching energy according to the output range,
achieving up to 75.71% lower analog-to-digital conversion energy. As a result, the
processor achieves 95.1% accuracy on the 10 keywords of the Google Speech Commands
dataset [23], with a macro energy efficiency of 34.8 TOPS/W and a system energy efficiency of
21.6 TOPS/W. As a result, the proposed processor achieves an energy consumption of
0.62 $\mu$J/decision for KWS.
ACKNOWLEDGEMENT
This work was supported by the Yonsei University Research Fund of 2025-22-0142. The
EDA tool was supported by the IC Design Education Center (IDEC), Korea.
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Dongwook Kim received his B.S. degrees in electronic engineering and computer and
information engineering from Kwangwoon University, Seoul, South Korea, in 2024, and
the M.S. degree in Electrical Engineering from Ulsan National Institute of Science
and Technology, Ulsan, South Korea, in 2026. Since 2026, he has been with the Weapon
Integration Team, Korea Aerospace Industries, Sacheon, South Korea. His current research
interests include low-power and energy-efficient computing-in-memory for deep neural
networks and avionics systems.
Kyuho Jason Lee received the B.S., M.S., and Ph.D. degrees from the School of Electrical
Engineering, Korea Advanced Institute of Science and Technology (KAIST), Daejeon,
South Korea, in 2012, 2014, and 2017, respectively. Dr. Lee worked for Samsung Research
America as a Hardware Designer in 2016. From 2017 to 2018, he was a post-doctoral
researcher with the Information Engineering and Electronics Research Institute, KAIST.
Then, he worked as an Assistant/Associate Professor with the Department of Electrical
Engineering and the Graduate School of Artificial Intelligence, Ulsan National Institute
of Science and Technology (UNIST), Ulsan, South Korea, from 2018 to 2025. He is currently
an Associate Professor with the School of Electrical and Electronics Engineering,
Yonsei University, Seoul, South Korea. Dr. Lee has been serving as a TPC member for
the IEEE Asian Solid-State Circuits Conference (A-SSCC), the ACM/IEEE Design, Automation
and Test in Europe (DATE), ACM/IEEE Design Automation Conference (DAC), IEEE International
Conference on Electronics, Information, and Communication (ICEIC), IEEE International
SoC Design Conference (ISOCC), IEEE Asia and South Pacific Design Automation Conference
(ASP-DAC), IEEE Asia Pacific Conference on Circuits and Systems (APCCAS). He also
has served as an OC member for the IEEE International Conference on Artificial Intelligence
Circuits and Systems (AICAS), IEEE International Conference on Electronics, Information,
and Communication (ICEIC), IEEE International Conference on Consumer Electronics Asia
(ICCE-Asia), and IEEE Asian Solid-State Circuits Conference (A-SSCC). He was the TPC
Vice Chair of A-SSCC 2026 and Guest Editor for IEEE Journal of Solid-State Circuits
(JSSC).