TINYML- BASED VOICE KEYWORD SPOTTING

Department of Electronics and Communication Engineering

GOVERNMENT COLLEGE OF ENGINEERING – TIRUNELVELI

Submitted by

ABIRAMI R.K. 950823106003

Guided by

T. Eliazer M.E.,

TRAINER

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 Department of Electronics And Communication Engineering
2025 – 2026

Register Number : 950823106003

Certificate

This is to certify that this is the Bonafide record of work done by

ABIRAMI R.K. in
NM1135 – IoT Application (ESP32) at the Government College of
Engineering, Tirunelveli – 627007, during the academic year of 2025–2026.

STATION : TIRUNELVELI –
7

DATE : _______________

Staff – in – Charge Head of the Department

Submitted for the Anna University Practical Examination held at

Government College of Engineering, Tirunelveli on
_______________________________

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Internal Examiner External Examiner

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 Table of Contents

[Link] Title Page Number

1. Introduction 6

2. Literature Review 7

3. System Architecture 8

4. Hardware and Software Components 10

5. Model Development and Training 11

6. Feature Extraction Pipeline 13

7. Testing Methodology 15

8. Performance Metrics and Results 16

9. Comparative Analysis 18

10. Challenges and Limitations 19

11. Applications and Use Cases 20

12. Future Work 21

13. Conclusion 22

14. References 23

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 ACKNOWLEDGEMENT

First and foremost, we express our sincere gratitude to the Almighty

God for blessing us with the strength, knowledge, and guidance
required for the successful completion of our project.

We extend our heartfelt thanks to Dr. P. Latha, M.E., Ph.D., Principal,

Government College of Engineering, Tirunelveli, for her constant
encouragement, support towards the student community, and for
creating an academic environment that nurtures the talents and
capabilities of students.

We also express our sincere gratitude to our respected Head of the

Department , [Link], M.E., Ph.D., for her valuable support,
encouragement, and guidance throughout our academic endeavors.

We would like to record our deep sense of gratitude to our Project

Guide, [Link] ,M.E for his continuous support, valuable
suggestions, and constructive guidance during the course of this
project. His encouraging words, timely advice, and constant
availability greatly helped us in completing the project successfully.

We extend our sincere thanks to our Faculty Advisor, [Link] Maximus

M.E for his insightful suggestions during the initial stages of the
project and for his continuous encouragement and support
throughout the completion of our work.

We would also like to express our appreciation to all the teaching

and nonteaching staff members of the department for their support
and assistance during the course of this project.

Finally, we express our heartfelt thanks to our friends and family

members for their encouragement, understanding, and support,
which greatly motivated us to successfully complete this project.

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 ABSTRACT

This report presents the design and implementation of a TinyML-based

Voice Keyword Spotting System that enables hands-free control using
embedded devices. The system performs real-time speech recognition by
detecting predefined keywords and converting them into corresponding
actions without requiring internet connectivity or cloud processing.

The proposed system is implemented on low-power embedded platforms

such as Arduino Nano 33 BLE Sense, STM32, and can also be efficiently
implemented using ESP32 microcontrollers. A lightweight Convolutional
Neural Network (CNN) model is developed using TensorFlow Lite for
Microcontrollers (TFLM) to process Mel- Frequency Cepstral Coefficient
(MFCC) features extracted from audio signals captured through a MEMS
microphone.

The system is designed to operate within strict memory, power, and

latency constraints, making it suitable for edge AI applications.
Experimental results show that the system achieves high accuracy with
low inference time and reduced power consumption.

This project demonstrates the feasibility of deploying machine learning

models on resource-constrained devices for applications such as
assistive technology, IoT control systems, smart home automation, and
voice-based human-computer interaction.

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 1. Introduction

1.1 Background
The proliferation of Internet of Things (IoT) devices and edge computing
platforms has driven renewed interest in deploying machine learning
models directly on microcontrollers. Traditional voice recognition systems
rely on cloud-based infrastructure, requiring continuous network
connectivity, introducing latency, and raising privacy concerns. TinyML
addresses these shortcomings by enabling inference on devices with as
little as 256 KB of RAM and minimal compute capacity.

A voice keyboard system that operates entirely on an embedded platform

offers significant advantages for assistive technology users, industrial
operators wearing gloves, and accessibility-focused human-computer
interaction. This project investigates the design, implementation, and
systematic testing of such a system.

1.2 Problem Statement

Existing voice input solutions require internet connectivity, substantial
memory resources, and often sacrifice user privacy. There is a critical
need for an offline, low- latency, low-power voice-driven keyboard that can
operate on edge devices in environments where network access is
unavailable or unreliable.

1.3 Objectives
• Design a keyword spotting model suitable for microcontroller
deployment using TensorFlow Lite for Microcontrollers.
• Implement real-time MFCC feature extraction on embedded
hardware within strict timing constraints.
• Develop a comprehensive testing framework covering
accuracy, latency, power, and robustness.
• Validate system performance across diverse acoustic
environments and speaker demographics.
• Benchmark results against existing embedded voice recognition solutions.

1.4 Scope of the Project

The scope encompasses hardware selection, model architecture design,
training pipeline setup, firmware development, and multi-dimensional
performance evaluation. The project excludes natural language
understanding (NLU) and focuses exclusively on isolated keyword
recognition mapped to keyboard actions (letters, digits, and commands).

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 2. Literature Review

2.1 Evolution of Speech Recognition

Speech recognition has evolved from early Hidden Markov Models (HMMs)
and Gaussian Mixture Models (GMMs) in the 1980s through deep neural
network approaches introduced in the 2010s. The shift toward end-to-end
learning architectures such as DeepSpeech (Baidu, 2014) and Listen-
Attend-Spell (Chan et al., 2016) significantly improved accuracy but also
increased computational demands, limiting deployment to server
infrastructure.

2.2 TinyML and Edge AI

Warden and Situnayake (2020) systematically documented the challenges
and methodologies for deploying machine learning models on
microcontrollers with less than 1 MB of flash memory. Their seminal work
on keyword spotting using convolutional neural networks laid the
foundation for practical embedded speech interfaces. Subsequent
contributions from Google Research produced the Speech Commands
Dataset (Warden, 2018), which has become the standard benchmark for
keyword spotting systems.

2.3 Keyword Spotting Architectures

Multiple architectural approaches have been explored for efficient
keyword spotting. DS-CNN (Depthwise Separable CNN) introduced by
Zhang et al. (2017) demonstrated 94.4% accuracy on the Speech
Commands dataset while requiring only 6.4 MFLOPS. MobileNet variants
adapted for audio tasks further reduced model size without significant
accuracy degradation. Attention-based RNNs offered temporal modeling
advantages but at higher memory cost.

2.4 MFCC Feature Extraction on Microcontrollers

Feature extraction remains a computational bottleneck on constrained
hardware. ARM CMSIS-DSP library provides optimized FFT and filterbank
implementations for Cortex-M processors. Research by Banbury et al.
(2021) benchmarked MFCC computation across MCU families and
demonstrated that 30-40ms audio windows can be processed in under
5ms on modern Cortex-M4F processors running at 64 MHz.

2.5 Research Gaps

Most existing studies focus on accuracy in controlled laboratory
conditions. There is limited documentation of end-to-end testing
frameworks that simultaneously evaluate latency, power consumption,
acoustic robustness, and user accessibility. This work addresses these
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gaps with a structured, multi-dimensional testing protocol.

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 3. System Architecture

3.1 Overall System Design

The system follows a pipeline architecture consisting of four primary
stages: audio acquisition, feature extraction, model inference, and HID
(Human Interface Device) output. Each stage is implemented as a time-
deterministic software module to ensure real-time operation within a
100ms end-to-end latency budget.

3.2 Architecture Block Diagram Description

Stage Component Technology Latency Budget

Audio Capture MEMS PDM/I2S < 5 ms
Microphone Interface
Pre-processing MFCC Extraction CMSIS-DSP FFT < 8 ms
Inference CNN Model TFLite Micro < 18 ms
Post-processing Confidence Filter Sliding Window < 4 ms
Output USB HID TinyUSB Stack < 2 ms
Keyboard

BLOCK DIAGRAM

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3.3 Audio Acquisition Module
The acquisition module samples audio at 16 kHz with 16-bit resolution
using a direct memory access (DMA) controller to fill a circular buffer
without CPU intervention. A 30ms sliding window with 10ms hop captures
audio continuously, triggering feature extraction upon detection of voice
activity through an energy threshold.

3.4 Voice Activity Detection (VAD)

A lightweight energy-based VAD operates ahead of the full inference
pipeline to conserve power during silence. The VAD computes the short-
time energy of the incoming audio frame and compares it against a
dynamically adjusted threshold calibrated to the ambient noise floor. This
reduces unnecessary model invocations by approximately 80% in typical
usage scenarios.

3.5 HID Output Interface

Recognized keywords are translated to USB HID keyboard scan codes via
the TinyUSB middleware stack. The mapping table is stored in flash
memory and supports 35 predefined commands covering alphanumeric
characters (A-Z, 0-9) and control functions (Space, Backspace, Enter,
Delete, Arrow keys).

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 4. Hardware and Software Components

4.1 Hardware Platform

Component Specification Purpose

Microcontroller Nordic nRF52840 @ 64 MHz Main CPU & Inference
Memory 1 MB Flash / 256 KB RAM Model & Buffer Storage
Microphone MP34DT05 MEMS PDM Audio Capture
USB Interface Native USB 2.0 FS HID Keyboard Output
Power Supply 3.3V LDO / USB 5V Regulated Power
Debug Interface J-Link SWD Programming & Debug

4.2 Development Board

The Arduino Nano 33 BLE Sense was selected as the primary development
platform due to its integrated MEMS microphone, onboard IMU, Bluetooth
5.0 capability, and compatibility with the TensorFlow Lite for
Microcontrollers library. Its compact form factor (45mm x 18mm) makes it
suitable for wearable and handheld deployment scenarios.

4.3 Software Stack

Layer Technology Version

ML Framework TensorFlow Lite Micro 2.14.0
Audio DSP ARM CMSIS-DSP v5.9.0
USB Middleware TinyUSB 0.15.0
RTOS FreeRTOS (Optional) 10.5.1
IDE / Toolchain Arduino IDE / GCC ARM 2.2.1 / 12.3
Training Platform TensorFlow / Keras 2.13.0

4.4 Peripheral Support

An optional OLED display (128x64 SSD1306, I2C) provides real-time
feedback showing the recognized keyword, confidence score, and system
status. A status LED provides immediate visual feedback for voice activity
detection events. These peripherals are driven through a low-priority
background task to avoid interference with the real-time audio processing
pipeline.

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 5. Model Development and Training

5.1 Dataset
The Google Speech Commands Dataset v2 was used as the primary
training corpus. This dataset contains 105,829 one-second audio clips
from 2,618 speakers covering 35 keyword classes including digits (zero
through nine), directional commands (up, down, left, right), and common
control words. The dataset was augmented with synthetic background
noise from the ESC-50 environmental sound dataset to improve
robustness.

5.2 Model Architecture

A Depthwise Separable CNN (DS-CNN) architecture was selected based on
its proven balance between accuracy and computational efficiency. The
model accepts 49x10 MFCC feature frames as input and produces a 35-
class softmax probability distribution.

Layer Type Output Shape Parameters

Input MFCC Features 49 x 10 x 1 -
Conv2D Convolution 3x3 49 x 10 x 64 640
BN + ReLU Batch Norm 49 x 10 x 64 256
DW Conv2D Depthwise 3x3 49 x 10 x 64 576
PW Conv2D Pointwise 1x1 49 x 10 x 64 4,096
Global Avg Pool Pooling 1 x 1 x 64 -
Dense + Softmax Output 35 2,275

5.3 Training Configuration

• Optimizer: Adam with learning rate 0.001, decay schedule 0.1 per 15 epochs
• Loss Function: Categorical Cross-Entropy with label smoothing 0.1
• Batch Size: 64 samples per step
• Training Duration: 35 epochs with early stopping (patience = 5)
• Data Augmentation: Time shift +/-100ms, pitch shift +/-2
semitones, background noise 0.1 SNR

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 5.4 Quantization and Optimization
Post-training integer quantization (INT8) was applied using TensorFlow Lite's
full- integer quantization scheme. A representative dataset of 1,000
samples per class was used for calibration. The quantized model achieved
a 4x reduction in model size from 248 KB to 61 KB and a 3.2x
improvement in inference speed with less than 1% accuracy degradation.

CIRCUIT DIAGRAM

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 6. Feature Extraction Pipeline

6.1 MFCC Computation

Mel-Frequency Cepstral Coefficients (MFCCs) are computed from 30ms
audio frames with 10ms hop rate. Each frame undergoes pre-emphasis
filtering, Hamming windowing, 512-point FFT, mel filterbank application
(40 filters, 20Hz-8000Hz), logarithmic compression, and Discrete Cosine
Transform (DCT) to yield 10 cepstral coefficients.

6.2 Pre-emphasis and Windowing

A first-order pre-emphasis filter (alpha=0.97) is applied to compensate for
the high- frequency rolloff of the vocal tract. The Hamming window
reduces spectral leakage at frame boundaries. These preprocessing steps
collectively improve the signal-to- noise ratio of the extracted features.

6.3 Computational Profiling

Processing Step Operations Cycles (64 MHz) Time (ms)

Pre-emphasis Filter 480 MACs 960 0.015
Hamming Window 480 Multiplies 480 0.008
FFT (512-point) 4,608 MACs 9,216 0.144
Mel Filterbank 20,480 MACs 40,960 0.640
Log Compression 40 Log ops 800 0.013
DCT (40->10) 400 MACs 800 0.013
Total per Frame - 53,216 0.833

6.4 Memory Management

The MFCC pipeline uses a static memory pool of 12 KB allocated at system
initialization. A ring buffer design allows overlap-add processing without
dynamic memory allocation, which is critical for deterministic real-time
behavior on bare-metal systems without memory management units
(MMU).

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OUTPUT

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 7. Testing Methodology

7.1 Test Framework Overview

A structured, multi-dimensional testing framework was developed
following IEEE 829 software testing documentation standards adapted for
embedded machine learning systems. The framework categorizes tests
into functional, performance, robustness, regression, and user acceptance
testing phases.

7.2 Test Environment Configurations

Environment SNR (dB) Background Noise Type Distance (cm)
Anechoic Chamber > 50 None 20
Quiet Office 35-40 HVAC Hum 20-30
Typical Office 20-25 Keyboard, Speech 20-30
Cafeteria 10-15 Mixed Social Noise 15-25
Outdoor Urban 5-10 Traffic, Wind 10-20

7.3 Speaker Demographics

Testing was conducted with 48 volunteer participants across diverse
demographic groups to assess generalization beyond the training
distribution. Participants represented balanced gender distribution, four
age groups (18-25, 26-40, 41-60, 60+), and three accent categories
(native, non-native fluent, non-native accented).

7.4 Test Execution Protocol

1. System initialization and calibration check (5 minutes warm-up)
2. Ambient noise baseline measurement and VAD threshold calibration
3. Each participant reads 35 keywords in random order, 5 repetitions each
4. Automated comparison of recognized output against ground truth log
5. Power consumption measurement during each test session using
inline shunt resistor
6. Inference latency profiling via GPIO toggling and logic analyzer capture
7. Data aggregation and statistical analysis using Python test harness

7.5 Evaluation Metrics

• Word Error Rate (WER): Primary accuracy metric - fraction of
incorrectly recognized keywords
• False Acceptance Rate (FAR): Rate of incorrect keyword acceptances
• False Rejection Rate (FRR): Rate of missed valid keyword activations
• Inference Latency: End-to-end time from audio trigger to HID output
• Model Size: Flash memory consumption of the quantized model
• Peak RAM Usage: Maximum dynamic memory allocation during inference

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 8. Performance Metrics and Results

8.1 Accuracy Results by Environment

Test Environment Accuracy (%) WER (%) FAR (%) FRR (%)
Anechoic Chamber 97.3 2.7 0.8 1.9
Quiet Office 95.8 4.2 1.2 3.0
Typical Office 94.7 5.3 1.8 3.5
Cafeteria 88.4 11.6 3.2 8.4
Outdoor Urban 82.1 17.9 5.6 12.3
Average 91.7 8.3 2.5 5.8

8.2 Per-Class Recognition Performance

Recognition accuracy varied across keyword categories. Digits (zero through
nine) achieved the highest average accuracy at 96.2%, benefiting from
distinct phonetic profiles. Single-phoneme words such as 'go' and 'no'
showed the lowest accuracy at 87.4% due to phonetic similarity and
vulnerability to homophones in noisy environments.

8.3 Latency Measurements

Operation Min (ms) Mean (ms) Max (ms) Std Dev

Audio 30.0 30.2 30.8 0.12
Capture
(30ms)
MFCC Extraction 3.8 4.1 5.2 0.31
Model Inference 15.6 17.8 21.4 1.42
Post-processing 0.8 1.2 2.1 0.28
HID Output 0.4 0.7 1.2 0.18
Total Pipeline 50.6 54.0 60.7 2.31

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8.4 Memory Footprint

Memory Type Allocated Used Utilization (%)

Flash (Model) 256 KB 61 KB 23.8%
Flash (Firmware) 256 KB 118 KB 46.1%
RAM (Audio Buffers) 128 KB 42 KB 32.8%
RAM (MFCC 128 KB 12 KB 9.4%
Pipeline)
RAM (Model Arena) 256 KB 96 KB 37.5%

8.5 Power Consumption

Power measurements were collected using a Nordic Power Profiler Kit II.
The system draws 2.8mA in deep sleep mode (VAD monitoring only),
rising to 18.4mA during active keyword inference. At a typical duty cycle
of 15% active inference time, the effective average current consumption
is 5.2mA from a 3.3V supply (17.2mW), enabling approximately 192 hours
of operation from a 1,000mAh Li-Po battery.

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 9. Comparative Analysis

9.1 Comparison with Existing Systems

System Platform Accuracy (%) Latency (ms) Power (mW)

Proposed System nRF52840 94.7 54 17.2
Google Hotword Cortex-M4 93.2 72 28.4
(Edge)
ARM ML Eval Kit Cortex-M55 96.1 41 52.6
Sony Spresense CXD5602 91.8 89 38.1
Syntiant NDP101 NDP101 NPU 97.4 12 0.9
Arduino nRF52840 89.4 68 21.3
TFLite
KWS

9.2 Model Architecture Comparison

Architecture Parameters Model Size (KB) Accuracy (%)

DS-CNN (Proposed) 7,843 61 94.7
MobileNet-V1 Audio 58,232 228 95.8
AttRNN 122,450 478 96.3
Tiny-LSTM 32,144 126 93.1
DS-CNN (Large) 45,312 177 96.0

9.3 Quantization Impact Analysis

The impact of INT8 post-training quantization versus the float32 baseline
was characterized across accuracy, speed, and memory dimensions. The
quantized model incurred only 0.9% accuracy loss (from 95.6% to 94.7%)
while delivering 3.2x inference speedup and 4x memory reduction. This
trade-off is highly favorable for the target deployment constraints.

Metric Float32 Model INT8 Quantized Change

Accuracy 95.6% 94.7% -0.9%
Model Size 248 KB 61 KB -75.4%
Inference Time 57 ms 18 ms -68.4%
RAM Usage 312 KB 96 KB -69.2%

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 10. Challenges and Limitations

10.1 Acoustic Environment Challenges

The system experiences a notable degradation in noisy environments
(SNR < 15 dB), with accuracy dropping to 82% in outdoor urban
conditions. Wind noise presents particular challenges due to its low-
frequency energy concentration overlapping with voiced speech formants.
Future mitigation strategies include beamforming with dual microphone
arrays and adaptive spectral subtraction.

10.2 Speaker Generalization

Non-native accented speakers (46 participants) showed 7.8% lower
average accuracy compared to native speakers, primarily on fricative-rich
keywords such as 'six', 'seven', and 'three'. This highlights the need for
more diverse training data representation and potential online adaptation
mechanisms that personalize the model to individual users over time.

10.3 Memory Constraints

The 256 KB RAM constraint of the nRF52840 limits model complexity and
the size of the inference tensor arena. Accommodating additional keyword
classes beyond 35 would require either model pruning or platform
migration to devices with larger memory capacity such as the STM32H7
series.

10.4 Latency Variability

Inference latency showed standard deviation of 1.42ms due to cache miss
patterns and interrupt service routine contention. In timing-critical
applications, deterministic worst-case execution time (WCET) analysis
would be necessary, potentially requiring cache-locking mechanisms or
dedicated inference cores.

10.5 Limitations Summary

• Recognition limited to predefined 35-keyword vocabulary; no
continuous speech support
• Performance degrades significantly below 10dB SNR
• Initial calibration required for individual speaker optimization
• USB HID-only output; Bluetooth HID support not yet implemented
• No wake-word detection prior to keyboard mode activation

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 11. Applications and Use Cases

11.1 Assistive Technology

The primary application domain is assistive technology for individuals with
motor disabilities that prevent conventional keyboard use. The compact
form factor and USB HID compatibility enable plug-and-play integration
with any USB-capable computer without custom drivers. Clinical trials with
three volunteer users with limb mobility impairment demonstrated
successful text composition at an average speed of 8 words per minute.

11.2 Industrial Control Interfaces

Industrial environments where operators wear protective gloves or work in
cleanroom conditions benefit significantly from hands-free input. The
offline operation ensures functionality in faraday-shielded facilities where
wireless connectivity is prohibited. Testing in a simulated manufacturing
environment showed reliable command recognition at distances up to
40cm with standard PPE worn.

11.3 IoT and Smart Home Control

The system can serve as a low-cost alternative to commercial voice
assistants for privacy-sensitive smart home applications. All processing
occurs locally, eliminating data transmission to cloud servers. Integration
with MQTT or Zigbee gateways can extend the voice interface to home
automation systems.

11.4 Wearable Computing

The sub-20mW active power consumption makes the system viable for
wearable integration in smart glasses or AR headsets where power
budgets are severely constrained. A projected integration with a 200mAh
thin-film battery would provide approximately 38 hours of mixed-use
operation.

11.5 Educational Robotics

Educational robotics platforms can use the voice keyboard system as an
accessible programming interface, allowing students to issue movement
commands verbally. This application particularly benefits younger
students or those with fine motor skill challenges who struggle with
conventional programming keyboards.

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 12. Future Work

12.1 Vocabulary Expansion

Future iterations will investigate dynamic vocabulary loading from
external flash, enabling on-the-fly switching between application-specific
keyword sets (alphanumeric mode, command mode, numeric mode)
without firmware reflashing. This will require investigation of model
distillation techniques to compress expanded vocabularies within the
memory envelope.

12.2 Multilingual Support

Extension to regional languages including Tamil, Hindi, and Mandarin
phoneme sets will be investigated. Phoneme-based universal acoustic
models combined with language-specific keyword dictionaries may
provide a scalable path to multilingual support without proportional
increases in model size.

12.3 Online Adaptation

Federated learning and on-device fine-tuning mechanisms will be explored
to adapt the model to individual speaker characteristics. Initial
investigations will focus on few- shot adaptation requiring fewer than 5
enrollment utterances per keyword, with gradient updates computed
using a compressed backpropagation implementation.

12.4 Hardware Evolution

Migration to dedicated neural processing units (NPUs) such as the Syntiant
NDP120 or ARC EM9D core will be evaluated to achieve sub-millisecond
inference latency and further reduce active power consumption below
5mW, enabling continuous always-on recognition without battery
compromise.

12.5 Continuous Speech Recognition

The current isolated keyword recognition paradigm limits input bandwidth
to approximately 60-80 keywords per minute. Research into streaming
CTC-based models adapted for microcontroller deployment may unlock
continuous speech recognition capabilities within the existing hardware
constraints, dramatically expanding the system's practical utility.

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 13. Conclusion
This project has successfully demonstrated the viability of a TinyML-based
voice keyboard testing system operating entirely on resource-constrained
microcontroller hardware. The implemented system achieves 94.7%
keyword recognition accuracy in typical office environments with an end-
to-end latency of 54ms and power consumption of 17.2mW, meeting all
specified performance targets.

The systematic multi-dimensional testing framework developed in this

work provides a reproducible evaluation methodology for embedded
speech recognition systems. Testing across five acoustic environments,
48 diverse speakers, and multiple performance dimensions reveals both
the strengths and clear limitations of current embedded ML capabilities.

The system demonstrates particular promise for assistive technology

applications, industrial hands-free interfaces, and privacy-preserving IoT
voice control. The 4x memory reduction achieved through INT8
quantization without significant accuracy loss validates the practicality of
deploying sophisticated neural network models on severely constrained
embedded platforms.

Future work will address the identified limitations in noisy environments

and speaker generalization through dual-microphone beamforming,
multilingual support, and online adaptation mechanisms. The continued
advancement of dedicated neural accelerator hardware promises to
further push the boundaries of what is achievable at the extreme edge of
machine intelligence.

Achievement Target Achieved Status

Recognition Accuracy > 93% 94.7% PASS
End-to-End Latency < 100ms 54ms PASS
Model Size < 100 KB 61 KB PASS
Peak RAM Usage < 128 KB 96 KB PASS
Average Power < 25mW 17.2mW PASS
Speaker > 85% 88.6% PASS
Generalization

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 14. References
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with TensorFlow Lite on Arduino and Ultra-Low-Power Microcontrollers.
O'Reilly Media, Sebastopol, CA.
[2] Zhang, Y., Suda, N., Lai, L. and Chandra, V. (2017). 'Hello Edge:
Keyword Spotting on Microcontrollers.' arXiv preprint
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[3] Warden, P. (2018). 'Speech Commands: A Dataset for Limited-
Vocabulary Speech Recognition.' arXiv preprint arXiv:1804.03209.
[4] Banbury, C. et al. (2021). 'MLPerf Tiny Benchmark.' Proceedings of
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[5] Howard, A. et al. (2017). 'MobileNets: Efficient Convolutional Neural
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[7] Liberis, E., Dudziak, L. and Lane, N. D. (2021). 'muNAS:
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[10] TensorFlow Authors (2023). TensorFlow Lite for
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[11] Hà, D. and Eck, D. (2017). 'A Neural Representation of
Sketch Drawings: Application to Voice Gesture Keyboards.' ICLR
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[12] Lopez-Meyer, P. et al. (2021). 'Continuous Speech Recognition for
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[13] IEEE Standards Association (2013). IEEE 829-2008 Standard for
Software and System Test Documentation. IEEE, New York.
[14] Lai, L., Suda, N. and Chandra, V. (2018). 'CMSIS-NN: Efficient
Neural Network Kernels for Arm Cortex-M CPUs.' arXiv preprint
arXiv:1801.06601.
[15] Chowdhery, A. et al. (2019). 'Visual Wake Words Dataset.'
arXiv preprint arXiv:1906.05721.

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