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TreeSpirit: Illegal logging detection and alerting system using audio

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Conference Paper · December 2017

DOI: 10.1109/SKIMA.2017.8294127

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 TreeSpirit: Illegal Logging Detection and Alerting
System using Audio Identification over an IoT
Network
P.G. Kalhara∗ , V.D. Jayasinghearachchi† , A.H.A.T. Dias‡ , V.C. Ratnayake§ , C. Jayawardena¶
and N. Kuruwitaarachchik
∗†‡§ Department of Software Engineering, ¶k Department of Information Systems Engineering,
Sri Lanka Institute of Information Technology, New Kandy Road, Malabe 10115, Sri Lanka

Abstract—Illegal logging has been identified as a major prob- Forest Monitoring WSN Architecture that addresses the above
lem in the world, which may be minimized through effective concern.
monitoring of forest covered areas. In this paper, we propose and
describe the initial steps to build a new three-tier architecture for II. S YSTEM D ESIGN
Forest Monitoring based on Wireless Sensor Network and Chain-
saw Noise Identification using a Neural Network. In addition to
detection of chainsaw noises, we also propose methodologies to
localize the origin of the chainsaw noise. Listening Post Listening Post
(Arduino) (Arduino)
Index Terms—illegal logging, chainsaw, forest monitoring,
wireless sensor network, neural network, audio identification,
deforestation.
Listening Post Sound Processor in Forest Listening Post
I. I NTRODUCTION (Arduino) (Raspberry Pi) (Arduino)

Illegal logging has been identified as a major problem in

the world, with studies indicating that more than 100 million
cubic metres of timber are still being cut illegally each year.
Furthermore, destruction of the worlds forests contributes up
to 20% of global carbon dioxide gas emissions [1].
Forest Surveillance plays a key role in controlling Illegal
Logging. Various attempts have been made to carry out
effective surveillance of large forest-covered areas [2], [3] such
as, Satellite Image Processing, Local Video Surveillance and
Detection of Acoustic Signals of Chainsaws.
According to [2] and [4] currently, the optimal and the
only feasible surveillance strategy against Illegal Logging is
the detection of acoustic signals of chainsaw considering the
practicalities of the logging industry. This position is further
strengthened by the fact that many available studies as well as
existing commercial implementations focus on identification
of Illegal Logging through detection of acoustic signals of
chainsaws [2], [4]–[8].
Wireless Sensor Networks (WSN) play a key role in such
systems based on detection of acoustic signals of chainsaws.
Forest Monitoring WSNs consist of computing devices fixed
strategically in a forested area which are capable of sensing,
transmitting and in some instances processing the data cap-
tured [2], [4], [5], [7].
While we acknowledge the valuable contributions of previ-
ous and currently on-going work in this domain, we feel that
a cost-effective and practical long-range forest monitoring ar-
chitecture based on detection of acoustic signals of chainsaws
has not yet been realised. Hence this paper proposes a new
Base Station Computer
(Raspberry Pi)
Cloud API
Web Application SMS
Gateway
Fig. 1. High-level System Architecture
The proposed system consists of Listening Posts, Sound
Processors and a Base Station. The Listening Post continually
listens and transmits the sounds of the forest to the designated
Sound Processor. Each Sound Processor is allocated with three
Listening Posts. The Sound Processor will process the sound to
identify any signals that could potentially be chainsaw noise.
If it does detect such a sound, it will locate the origin of
the sound. Then this location information is transmitted to
the base station. The base station uploads such received data
to the Cloud Application Programming Interface (API) which
makes the data readily available through the associated web
application and Short Message Service (SMS) gateway.
The Listening Posts are based on the Arduino Uno (Fig. 2)
device while the Sound Processors and the Base station are
based on the Raspberry Pi 3 Model B (Fig. 3) device, attached
with additional hardware as listed in Table I. All devices are

Fig. 2. Arduino Uno
Fig. 3. Raspberry Pi 3 Model B
powered by solar panels with battery packs as backup power
systems.
TABLE I
H ARDWARE D EVICES U SED
Platform Additional Devices
nRF24L01+ WiFi Module
nRF24L01+ WiFi Module Base
DS3231 Real Time Clock Module
Arduino Uno
ADMP401 MEMS Microphone
Solar Panel
Battery Pack
nRF24L01+ WiFi Module
nRF24L01+ WiFi Module Base
DS3231 Real Time Clock Module
Raspberry Pi 3 Model B
Texas Instruments CC1310 Module
Solar Panel
Battery Pack
III. C OMMUNICATION A RCHITECTURE
Communication Architecture is a part of the System Archi-
tecture focused on utilizing different hardware and software
components to transmit and receive data on different IoT
platforms defined in the System Architecture. Ideally, such
data transmissions between devices should be reliable and have
considerable range while keeping associated legal/ licensing
concerns, power, and hardware costs to a minimum. To address
all these concerns successfully, a low-cost hardware device
which transmits in the Industrial, Scientific, Medical (ISM)
radio band has been selected as the communication platform.
Two types of wireless links are required in the proposed
system,
1) Short range, high speed link between each Listening Post
and the Sound Processor.
2) Long range, low speed link between each Sound Pro-
cessor and the Base Station.
Fig. 4. nRF24L01+ 2.4 GHz RF Transceiver
In case 1, a radio link must be constantly maintained
between each Listening Post and the Sound Processor to
exchange the sounds recorded for analysis and sound local-
ization. It is important that the quality of such transferred
sounds remain adequate for analysis during exchange. How-
ever, since Listening posts are low power devices, it is also
important that the selected communication platform has low
power consumption. Therefore, a low cost, low power, high
speed communication chip, based on the nRF24L01+ 2.4 GHz
RF Transceiver (Fig. 4) has been selected as the preferred
technology. This chip is capable of transmitting data at speeds
upto 2 Mbps [9] and has a range up to 1000 m line of sight
with improvements [10].
In case 2, a radio link must be occasionally established be-
tween a Sound Processor and the Base Station to transmit data
regarding any potential illegal logging activity that has been
detected by the Sound Processor. Since the requirement of this
link is transmission of a limited set of characters indicating
the type of illegal activity and the predicted coordinates of the
origin of the sound, it is not important that the link is high
speed. However, Sound Processors are located deep within a
forest while Base Station remains outside a forest, there is a
considerable distance between each Sound Processor and the
Base Station. Hence, the range becomes an important factor.
Again, Sound Processors are low powered devices, hence it
is important that the selected communication platform has
low power consumption. To fulfill these requirements Texas

Fig. 5. Texas Instruments CC1310 LaunchPad
Instruments CC1310 (Fig. 5 [11] communication module has
been selected as the preferred technology.
IV. S OFTWARE A RCHITECTURE
Software architecture of the proposed system can be divided
into four main sections: Sound Recognition Module, Sound
Localization Module, System Reliability and Real-time Web
Application. Fig. 6 represents a high-level view of the overall
software architecture of the proposed system.
A. Sound Recognition Module
1) Sound Identification Process: This operation identifies
the incoming sound signals from the environment. To fulfill
the requirement of the device we have tried two approaches
which are Audio Fingerprinting Mechanism and Deep Neural
Network for Sound Identification (DNN).
The first approach was an Audio Fingerprinting Mechanism
[12]. This mechanism repeatedly uses FFT over small windows
of time in the audio samples to create a spectrogram of the
audio. Based on these spectrograms, it stores combination
of frequency-time data into a database. By using this data-
set, it will compare the incoming sound signal. but the worst
case is it compares the exact value with the available data-
set. Therefore, in order to increase the accuracy of the audio
identification process, it needs to maintain a huge data-set
inside the database. Since the Sound Processor has a limited
storing capacity this approach was not a suitable solution for
the problem.
The second approach was to use a Deep Neural Network
(DNN) with a proper data-set [13]. This module mainly
focuses on identifying chainsaw sounds in the forest. There-
fore, using a properly trained DNN, the device can identify
chainsaw sounds with higher accuracy.
2) Data-sets: One of the main problems with training a
Deep Neural Network in a supervised manner is the amount
of computational effort and labeled data required for efficient
learning. But the authors were not able to find a proper data-set
of chainsaw sounds available publicly. Therefore, the authors
had to collect a proper data-set. Our team was able to gather
more than 100 chainsaw audio clips from an actual rain-forest
at various distances and directions with natural obstacles in-
between.
3) Experiment Setup: We used Fourier’s Transform to
convert our audio data to the frequency domain. This allows
for a much simpler and compact representation of the data,
which will be exported as a spectrogram (Fig. 7). This process
will give us an image file containing the evolution of all the
frequencies of audio through time.
Time is on the x-axis, and frequency on the y-axis. The
highest frequencies are at the top and the lowest at the bottom.
The scaled amplitude of the frequency is shown in grey-scale,
with white being the maximum and black the minimum.
The next thing we must do is to deal with the length of the
spectrogram. We can create fixed length slices of the spectro-
gram, and consider them as independent samples representing
the audio. We can use square slices for convenience, which
means that we will cut down the spectrogram into 128 x 128
pixel slices. This represents 2.56 s worth of data in each slice.
After slicing all audio into square spectral images, We can
now train a Deep Neural Network to identify these samples.
For this purpose, we have used TensorFlow [14].
The CNN takes in wave pattern as spectrogram and it does
so based on ”weights” and ”biases” that need to have a correct
value for the prediction to work well. Each ”neuron” in a
CNN does a weighted sum of all of its inputs, adds a constant
called the ”bias” and then feeds the result through some non-
linear activation function. In this CNN we used softmax as an
activation function for the last layer.
The training operation is happened on the CNN using
100 sound clips in a batch. There were three Convolutional
layers, three ReLU layers, three Pooling layers and one Fully
Connected layer with 50% dropout.
Considering the available hardware resources and power
consumption limitations I decided to use Berry Conda package
environment management system [15] to deploy this module
inside the Raspberry Pi. Berry Conda is a conda based Python
distribution for the Raspberry Pi. With it, we can install and
manage a scientific or Pydata stack on our Raspberry Pi using
conda, a package and environment management system. All
this can be done without compiling a single package.
4) Test Results: The module was evaluated with test data
to check the accuracy of the module.
Fig. 9 and Fig. 10 depicts the loss and accuracy of the audio
identification module. The lower the loss, the better a model
(unless the model has over-fitted to the training data). The loss
is calculated on training and validation and its interpretation
 REST API Web Application

Remote
Sound Localization Module Database

Audio
2D Audio Identification
TDOA
Multilateration Module
Module
Module

Cloud Pusher
NRF24L01+ Radio
Module Transmitter
Web Socket API
Python Interpreter

Main Module

C++ ↔ Python Interface Raspberry Pi 3 (Base Station)

NRF24L01+
Module

Message
Composer
Radio
Transmitter
Protothreading Module

NRF24L01+ Radio
Received Audio Module Transmitter Audio Transmitter
(WAV Format)
+ Timestamp

Main Module Main Module

Raspberry Pi 3 (Sound Processor) Arduino Uno (Listening Post)

Fig. 6. Overall Software Architecture

Loss
2.6

2.4

Fig. 7. Fourier’s Transformation Spectrogram

2.2

2.0

1.8
Loss

1.6

Fig. 8. Square Slices 1.4

1.2

is how well the model is doing for these data. Loss is not in
1.0
percentage as opposed to accuracy and it is a summation of
the errors made for each sample in training or validation sets. 0.8
0 50 100 150 200 250 300 350
The loss graph approximately gives the value to 0.98. Epoch
According to the regression line, the module can further reduce
the loss function value by more epochs. However, the trained Fig. 9. Plot with Loss
module accuracy level is very high. Out of 50 test samples,
the module was able to identify 48 samples correctly.
is happening. To give the solution we tried out new methods
B. Sound Localization and available methods to find out which is fitted best for
Sound source localization in outdoor Environment without our purpose and architecture. Finally came up with TDOA
using microphone arrays and for specific architecture. Output combined with Multilateration as the solution for the SSL.
of this component is to point out the location where logging TDOA is basically about the time difference of received
 Accuracy
1.0
0.9
0.8
0.7
Accuracy
0.6
0.5
0.4
0.3
0 50 100 150 200 250
Epoch
Fig. 10. Plot with Accuracy
signals of given microphones. To get that we use FFT like
methods to calculate phase difference between two signals and
low band noise removal to remove the noise and increase the
accuracy. Component is placed between audio fingerprinting
module and cloud. TDOA depends heavily on environmental
factors like noise. Selected method is accurate up to about 10
meters distance between mics without optimization. Because
of this is the only possible method must go with this method
and accuracy will be decreased along with the distance. It is
recommended to be used low noise and less obstacles.
1) TDOA: To get the time difference of arrival between
two distanced microphones, phase property of the wave is
used. to do that GCC-PHAT algorithm, Fast Fourier Transform
algorithm is used with some filtering techniques to remove
unwanted noise.
Fig. 11. Fast Fourier Transformation
2) Fast Fourier Transformation: To increase the perfor-
mance of the algorithm Fast Fourier Transform is used. It
simplifies the wave. A FFT algorithm computes the discrete
Fourier transform (DFT) of a sequence, or its inverse (IFFT).
Fourier analysis converts a signal from its original domain (of-
ten time or space) to a representation in the frequency domain
and vice versa. An FFT rapidly computes such transformations
by factorizing the DFT matrix into a product of sparse (mostly
zero) factors. As a result, it manages to reduce the complexity
of computing the DFT from O(n2 ), which arises if one simply
applies the definition of DFT, to O(n log n), where n is the
data size figure.
3) Noise Filtering: Basically, it reduces selected frequen-
cies 6 decibel times than initial intensity. High pass filter is
used in pydub library [16] which based on scipy library [17].
4) 2D Audio Multilateration: Sound Localization is not
available/ only discussed briefly in the existing solutions. Most
available solutions notify which zone of the forest the signal
is detected from but this could be inadequate as a zone could
be a large area comprising of acres of dense forest. However,
the author considers Sound Localization as an important part
of a Illegal logging detection systems for tropical rainforests
300 350
as tropical rainforests are dense forests and unless one knows
where one is headed, it is easy to get disorientated and lost.
Therefore, to make the task of detection easy we introduce
sound localization. Sound localization is based on a cluster.
Originally a technique called Triangulation was considered for
the Sound Localization. It required an array of two, directional
microphones attached to each Listening Post. Due to cost,
technical and practical reasons, it was decided that only a
single MEMS digital Omnidirectional microphone will be
attached to each Listening Post. Triangulation is not possible
when using a single microphone. Hence a new technique called
Multilateration [18], [19] is now used to locate the origins of
a Sound.
Fig. 12 shows an explanation of the Multilateration algo-
rithm that has been used in this system.
Y
b(100,0) d(100,100)
R3
R1 Sound Source
(x,y,z)
50m
R0
R2
a(0,0) 50m c(0,100) X
Fig. 12. Multilateriation Algorithm
Let a, b, c, d be four Sound Listening Posts situated on a
100m × 100m grid. Let each nodes location be denoted by
→
−
P m = (xm , ym , zm ); where m = (0, 1, 2, 3) and Zm = 0 as
it is a 2-D Multilateration. Let the unknown location of the
→
−
Sound Source be denoted by, E = (x, y, z); where Z = 0 as
it is a 2-D Multilateration.
By using the Distance Formula,
→
− →
− p
Rm = | P m − E |] = (xm − x)2 + (ym − y)2 + (zm − z)2
(1)
 By defining a (0, 0) as the origin, this could be simplified xm 2 + ym 2 + zm 2 x1 2 + y1 2 + z1 2
Dm = vTm − vT1 − +
to, vTm vT1
p (8)
R0 = x2 + y 2 + z 2 (2)
Using Eq. 8 on nodes m = (2, 3) will result in two
Let τm = Time difference between the time the sound was simultaneous equations which could be solved easily giving
emitted by the emitter (Te ) and received by the receiver m. the (x, y) position of the unknown source relative to the origin
Let v = the speed of Sound waves in air in m/s. Let Tm = a(0, 0).
Time the receiver m received the sound. B2
( B3 D3 − D2)
However, the issue is Te is an unknown quantity. To x=
overcome this, consider the node a(0, 0) as the origin and (A2 − B2.A3
B2 )
calculate the Time Difference of Arrival of each node relative −(A3x + D3)
to node a(0, 0). y=
B3
vτm = vTm − vT0 Given that the position of the origin a(0, 0) is known in
latitude and longitude values, the position of the sound source
vτm = Rm − R0 (3) can be estimated correctly.
Solving equations 1, 2 and 3 will provide the x, y location of C. System Reliability
the unknown sound source. However, doing so would require A forest monitoring system can be considered as an high
solving several non-linear simultaneous equations which could availability system which should be reliable and stable. To
be time and resource intensive for an embedded system. achieve high system reliability, a protocol suite has been
Therefore, it should be simplified mathematically to reduce introduced.
the complexity.
D. Data Integrity Protocol
2 2 Automatic hibernation of devices during very low power
Rm = (vTm + R0 ) •
to prevent data corruption in memory card.
Rm 2 = (vTm )2 + 2vTm R0 + R0 2 • Message Queuing Mechanism to store and resend mes-

0 = (vTm )2 + 2vTm R0 + R0 2 − Rm 2 sages that failed to deliver due to connection failures.

1) Keep-alive Protocol: A keep-alive protocol relying on
(R0 2 − Rm 2 ) exchanging periodical keep-alive or heartbeat messages is used
0 = vTm + 2R0 + (4)
vTm to detect radio link failures.
Remove 2R0 term by substituting the m = 1 term. 2) Time Synchronization Protocol: This protocol ensures
maintaining a correct clock time between all connected de-
(R0 2 − R1 2 ) vices. If actual local time could not be determined, all devices
0 = vT1 + 2R0 + (5)
vT1 of a single node are set to have a single relative time irre-
By Eq. 4 - Eq. 5, spective of the actual local time which eliminates the error
introduced by incorrect time.
(R0 2 − Rm 2 ) (R0 2 − R1 2 ) 3) System Stability Protocol: This protocol ensures to min-
0 = vTm − vT1 + − (6)
vTm vT1 imize the impact occurred by device malfunctions which can
R0 2 − Rm 2 can be replaced by using equation 2. be identified by keep-alive protocol.
• When a Sound Processor is malfunctioning all the con-
Rm 2 = xm 2 +ym 2 +zm 2 −x2xm −y2ym −z2zm +x2 +y 2 +z 2 nected Listening Posts are switched to connection es-
tablishment mode upon informing the agents about the
Rm 2 = xm 2 + ym 2 + zm 2 − x2xm − y2ym − z2zm + R0 2
incident. If the Listening Post can detect any nearby
R0 2 −Rm 2 = −xm 2 −ym 2 −zm 2 +x2xm −y2ym +z2zm (7) Sound Processor a new connection will be automatically
established until the malfunctioning device is replaced.
Combining equations 6 and 7 will result in a set of linear • When a Listening Post is malfunctioning the sound detec-
equations. tion and localization algorithms are adjusted accordingly
to minimize the impact.
0 = xAm + yBm + zCm + Dm
E. Real-time Web Application
2xm 2x1
Am = − Numerous functions are carried out by the real-time web
vTm vT1
application, including, but not limited to, managing the de-
2ym 2y1 vices connected to the network, assigning Listening Posts to
Bm = −
vTm vT1 Sound Processors, alerting users about significant incidents
2zm 2z1 and deploying software updates. The front-end of the web
Cm = − application is developed using Angular [20], which is one of
vTm vT1
the most-widely used JavaScript-based frameworks. All the
system data are stored in a document-oriented database and
exposed to front-end through JSON based Representational
State Transfer (REST) API which is developed using expressJS
[21], a rich JavaScript Framework for NodeJS [22] for building
applications and services.
V. F UTURE W ORK
Sound Localization refers to a listener’s ability to identify
the location or origin of a detected sound in direction and
distance. Due to the unavailability of a matching Sound
Localization mechanism [23]–[26] suitable for the proposed
architecture, a new method has been developed based on the
principles of Acoustic Multilateration which will approximate
the origin of a sound. Further practical tests should be carried
out on this module to test and improve its accuracy.
VI. C ONCLUSION
A new architecture for monitoring forests to detect illegal
logging using chainsaw noise identification on a Wireless Sen-
sor Network (WSN) has been proposed in this paper as well
as a methodology based on a Neural Network to accurately
identify the chainsaw acoustic signals. The noise recognition
aspect along with the localization aspect of the system require
continued research for a successful implementation in a real
environment.
ACKNOWLEDGMENT
This work was initiated as a component of the final year
research project at the Sri Lanka Institute of Information
Technology (SLIIT), Malabe, Sri Lanka. The authors would
like to express their sincere gratitude to the institute for
providing excellent guidance throughout the project.
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