PS-1 Project Report

On

Design of SDR-based HD video communication

BY

Vinayak Tulsyan 2020AAPS0442H

Rahul Varshney 2020AAPS1756H

Joshitha Marripudi 2020A3PS0629H

Under the supervision of

Prof. Mithun Mondal

from BITS Pilani Hyderabad Campus
and

SE (Dr) V Mahesh Babu

from MCEME, Secunderabad

BIRLA INSTITUTE OF TECHNOLOGY AND SCIENCE, PILANI (RAJASTHAN)

HYDERABAD CAMPUS

(June 2022)
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 ACKNOWLEDGMENTS

We would like to thank BITS Pilani for giving us this once-in-a-lifetime

opportunity of working with such a prestigious institute Military College of
Electronics and Mechanical Engineering (MCEME), Secunderabad.

We would also like to thank the PS division for giving us this chance to experience
the field of SDR. Special thanks to MCEME for allowing us to work with them.
Thanks to our PS mentor, SE (Dr) V Mahesh Babu, for being so supportive and
giving us the opportunity to work under his expert guidance and for taking time out
of his busy schedule to help us. Special thanks to our PS instructor Prof. Mithun
Mondal for his advice and support.

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 ABSTRACT
Software-defined radio (SDR) is a radio communication system where components
that have been traditionally implemented in hardware (e.g., mixers, filters,
amplifiers, modulators/demodulators, decoders, etc.) are instead implemented by
means of software on a personal computer or embedded system. An SDR performs
significant amounts of signal processing in a general-purpose computer or a
reconfigurable piece of digital electronics. The goal of this design is to produce a
radio that can receive and transmit a new form of radio protocol just by running the
new software. SDRs have significant utility for cell phone services, which must
serve a wide variety of changing radio protocols in real-time.

Using SDR, various tools of communication systems were developed and

simulated by us, and finally, an FM receiver and a video transmitter/receiver were
made.

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 CONTENTS

Description Page Nos

Title page 1

Acknowledgement 2

Abstract 3

Objectives 5

Literature Review 6

Key deliverables 25

Potential outcomes 36

Time Table 45

Bibliography 46

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OBJECTIVES

1. Study of video compression tech - AVC/H.264, Digital video transmission tech, data

encryption methods, etc.

2. Design & Simulation of Tx and Rx, for secure and high data rate transmission (use of

MIMO-OFDM, encryption & compression algos) using GNU RADIO/ MATLAB.

3. Analysis of parameters: diversity order, OFDM channels, data rate, latency, etc.

4. Hardware realization of a transceiver using SDR(USRP).

5. Performance evaluation in terms of video quality, range of comn and analysis of increasing

the range by designing suitable wideband power amplifier.

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 LITERATURE REVIEW

Noise:
1. Laplacian Noise
Laplace distribution is mathematically defined as a function that provides the probabilities of
occurrence of different possible outcomes in an experiment. Hence it is called a continuous
probability distribution. The Laplace Noise simply defines the addition of variable Y with input
variable X having Laplace distribution.

F(x) = X(x) + Y(x)

Where Y(x) represent a variable containing Laplace distribution.

2. Gaussian Noise
Gaussian Noise is a type of Statistical Noise having a probability density function equal to
Normal distribution (I.e., Gaussian noise results in the continuous probability distribution). The
statistical function is a fraction of the variance of a dependent variable that cannot be predicted
by the given independent variable. Gaussian Noise is also called White Noise because if a white
Noise is passed through the linear filter, it still results in the Gaussian Noise. The parameters like
variance and means change.

3. Impulse Noise
Impulse Noise is a type of acoustic noise generated because of unwanted instantaneous sharp
sounds, for example, noise generated from scratching any object, utensils falling, sounds coming
while tapping on the wall, etc.

Quantization:
Consider a continuous form of analog audio signals. Before transmitting, these input wave signals
need to be sampled down to discrete-time signals. Further, to digitalize these discrete values, the
amplitude of the signals needs to be approximated between 0 and 1.

Hence Quantization process plays an important role. Quantization is a process of rounding off or
approximation of a continuous analog signal. It is the process of converting a continuous range
of values into a finite range of discrete values.

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The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels, it is also called sub-division levels. For example, in the figure mentioned
below, there are 2-bit quantized (Consider positive half or either negative half of cosine wave
signals) levels. The quality of a Quantizer output depends upon the number of quantization levels.
The higher the Quantizer levels, the better is the Quantization process.

Sampling:
Nyquist theorem was developed by Harry Nyquist in the year 1928 and states that “ an analog
signal waveform is converted to digital format and can be reconstructed back to original analog
signal if and only if the sampling rate is greater than or equal to, twice the highest frequency
component in the analog signal “.

The Nyquist theorem is mathematically expressed as

fs ≥ 2fmax,
Where fs= sampling frequency or sample rate.
fmax = Maximum frequency.

The sample rate determines the number of samples taken per second. The sample rate is inversely
proportional to the sample period, which is mathematically expressed as Sampling frequency =
fs = 1/ft.
Consider a bandlimited Signal with x (w) as an input signal, and signal is bandlimited to [-w, w].

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Filters:
1. LOW PASS FILTERS:
A Low pass filter is defined as a passive type of filter which is used in order to reject all the
unwanted high frequencies signals generated from the system, thus allowing only low-frequency
signals through it. The cut-off frequency, which is denoted as fc is an important parameter as it
defines the boundary condition required in the frequency response of a system. Hence in the case
of a Low pass filter, the energy beyond the boundary level will get attenuated or reduced.
Therefore a low pass filter is also defined as a filter allowing only those frequency signals which
are lower than the cut-off frequency signals. In a simple way, a Low pass filter can also be
explained as a filter that removes the short-term fluctuation and thus provides a smoothening of
signals as an output. It is also called a corner frequency or a break frequency.

2. High pass filter:

A High pass filter is a type of filter which is designed in order to reject all the low-frequency
signals, thus allowing only high-frequency signals to pass through it. It allows only those
signals which are greater than the cut-off frequency signals, thus attenuating frequency signals
lower than the cut-off frequency. The cut-off frequency is mathematically calculated as

fc =1/2πRC

Where, R= Resistor, C= Capacitor, fc= Cut-off frequency.

A high pass filter is also called a low cut filter.

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3. Bandpass:
A Low pass filter is used to filter out the unwanted high-frequency signals, thus allowing
the low-frequency signals. A High pass filter is a type of filter which is designed in order
to reject all the low frequencies signals, thus allowing only high-frequency signals to pass
through it. Filter circuits need to be designed in such a way that a filter can combine the
properties of the LPF (low pass filter) and HPF (high pass filter) into a single filter, which
is known as a bandpass filter. Band Pass Filter shows the ability to pass frequencies
relatively unattenuated over a specified frequency band.

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SOFTWARE-DEFINED RADIO (SDR)

Radio is any kind of device that wirelessly transmits or receives signals in the radio frequency
(RF) part of the electromagnetic spectrum to facilitate the transfer of information. With the
exponential growth in the ways and means by which people need to communicate - data
communications, voice communications, video communications, broadcast messaging,
command and control communications, emergency response communications, etc. – modifying
radio devices easily and cost-effectively has become business-critical. Software-defined radio
(SDR) technology brings the flexibility, cost efficiency, and power to drive communications
forward, with wide-reaching benefits realized by service providers and product developers
through to end-users.
Software-defined radio (SDR) is a radio communication system where components that have
been traditionally implemented in hardware (e.g., mixers, filters, amplifiers,
modulators/demodulators, decoders, etc.) are instead implemented by means of software on a
personal computer or embedded system. An SDR performs significant amounts of signal
processing in a general-purpose computer or a reconfigurable piece of digital electronics. The
goal of this design is to produce a radio that can receive and transmit a new form of radio protocol
just by running the new software. SDRs have significant utility for cell phone services, which
must serve a wide variety of changing radio protocols in real-time.
The hardware of a software-defined radio typically consists of a superheterodyne RF front end
which converts RF signals from and to analog RF signals, analog to digital converters, and digital
to analog converters which are used to convert digitized intermediate frequency (IF) signals from
and to analog form, respectively. Software-defined radio can currently be used to implement
simple radio modem technologies. In the long run, SDR is expected to become the dominant
technology in radio communications. The following are some of the things that SDR can do that
haven't been possible before

1. Software-defined radios can be reconfigured “on-the-fly,” i.e., the universal

communication device would reconfigure itself appropriately for the environment. It
could be a cordless phone one minute, a cell phone the next, a wireless Internet gadget
the next, and a GPS receiver the next.
2. Software-defined radios can be quickly and easily upgraded with enhanced features.
In fact, the upgrade could be delivered over the air.
3. Software-defined radios can talk and listen to multiple channels at the same time.
4. Remote reprogramming allows bug fixes to occur while the radio is in service, thus
reducing the time and costs associated with operation and maintenance.
5. New kinds of radios can be built that have never before existed. Smart radios or
cognitive radios (CRs) can look at the utilization of the RF spectrum in their
immediate neighborhood and configure themselves for the best performance.

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Listed below are some of the popularly used SDRs:
1. Hack RF One Software Defined Radio (SDR), ANT500 & SMA Antenna Adapter Bundle

2. Ubertooth One SDR

3. YARD Stick One USB Transceiver & 915MHz Antenna

4. Seeedstudio KiwiSDR Kit Software Defined Radio with BeagleBone Green

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5. NESDR Mini 2+ 0.5PPM TCXO RTL-SDR & ADS-B USB Receiver Set w/ Antenna

6. NESDR Nano 2+ Tiny Black RTL-SDR USB Set

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SDR HARDWARE

The baseband section includes the host system with Gnu radio companion and UHD installed.
UHD is abbreviated as USRP Hardware driver, which is used to interface Gnu radio with SDR.
The combination of the IF section and RF section together results in SDR

IF section includes FPGA, which can up-convert and down-convert the given signals in order to
convert the signals into intermediate frequency signals.

Once the signal is converted to an intermediate frequency signal, it is further converted to digital
signals. This digital signal has to be converted to a high radio frequency signal before the
transmission through the Antenna, which can be probably achieved using RF Daughterboard

USRP
Universal Software Radio Peripheral (USRP) is a range of software-defined radios designed
and sold by Ettus Research and its parent company, National Instruments.

USRP SDRs are ideal for developing and prototyping complex wireless designs. After you design
and simulate your digital signal processing (DSP) algorithms, you need to prototype them in a
real-world environment.

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 Detailed View of the USRP-2944 10 MHz–6 GHz SDR

Key Features
Wide Selection of RF I/O

USRP SDRs offer frequency ranges between 10 MHz and 6 GHz and instantaneous bandwidth
up to 160 MHz

Scalable for Multichannel Operation

Easier FPGA Programming

USRP SDRs make FPGA programming easier by abstracting many of the low-level design tasks
associated with custom hardware design.

14
Various SDR Hardware offered by National instruments.

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VIDEO COMPRESSION:
Let us consider a video a combination of different images in order-time sequence. To send video
from one remote location to another, frame by frame incurs a lot of costs and computational
power, so we must find a way to reduce both.
Therefore a technique to meet the above requirement was developed called video compression.

Steps of video compression:

● Motion estimation, motion vector search

● Motion compression-based prediction
● Derive prediction error

Consider a video where a girl is moving in front of a bush, and we divide the video into various
frames in an ordered sequence of time. A very intuitive way of compressing the video would be
to predict the future frame from the current frame or even the current frame using the previous
frames. We need not code every image independently but use the existing information to get a
new image. But, a problem arises that we must redundantly do the process if two frames are the
same in the time sequence, which is called temporal redundancy.

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Let us consider the above image of a simple circle moving in a block. In the previous picture, we
can see that the ball is in the centre and in the present picture, it moves a little bit to the right.
Hence we take the difference picture that is the absolute difference between each element in the
matrix of images to encode. To decode we use difference image and add the previous image to
get the present image.

We use the difference picture because it has lower entropy and hence it is not calculation
intensive to encode or decode the complete video.

To bring non 0 values close to zero in difference picture we sample the frames in high frequencies

We use/ calculate motion compensation:

How many pixels are different is our current image different from previous image?

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Motion Compensation:

We divide the image into NxN blocks, Usually by default it is 16x16 but we use N=8 for
chrominance is chroma subsampling is 4:2:0.
The motion compensation is performed at a macro-block level. Let us consider the target frame
(TF) or the target image (TI) as the current frame, and the reference frame (RF) as the previous
image or the future image of the video.
We presume a vector as a motion vector that defined the position of the desired part of the
macroblock of the reference image with respect to the target image. If we consider our reference
frame as a future frame is called backward prediction and if the reference frame is the previous
image we call it forward prediction.

Motion Vector and MAD:

As mentioned earlier the motion vector is used to predict the frame around our target image and
thus we must find the reference image in a certain range of values. Usually, the MV is confined
in a small region [-p, p], which makes the window size 2p+1.

Here the vector that connects (x0, y0 ) and (x, y) is our Motion Vector MV.

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The difference between two macroblocks can be measured by Mean Absolute Difference
(MAD)

N Size of MB

k&l Indices of pixels in MB

i&j Horizontal and vertical differences

C(x+k, y+l) Pixels of MB in TF

R(x+i+k, y+j+l) Pixels of MB in RF

The complete goal of this search find the motion vector and Minimum MAD.

There are different types of search algorithms that we generally use in video compression:

6. Sequential search:
Here, every block is searched sequentially exactly like the above algorithm, it
undertakes (2p+1)x(2p+1) MAD searches. It is also called a full search, we take our
macroblock in the centre of the window and search the other macroblock accordingly
within the frame.

The motion vector that offers the least(i,j) values are designated as the macroblock in
the TF. This search method is very costly to perform.

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2. 2D Logarithmic Search:
Cheaper version, suboptimal yet effective. We perform several operations to get an optimal
solution like in binary search.

Initially, only 9 locations in the search window are used seeds for MAD-based search nd are
marked as 1. The one that yields the minimum MAD is located. Move the centre of the search to
your new region.

Reduce the step size to half.

In the next iteration Mark the new locations/ seeds of MAD-based search as seeds and proceed
so on we proceed to get our motion vector.

3. Hierarchy Search:
Initially estimate the motion vector for a reduced resolution of the image and then tune the motion
vector accordingly to find the target frame at a higher resolution. P proportionally reduced
Benefit of this method is we can have a multi-resolution approach and also the number of samples is
greatly reduced.

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Here is a comparison of the 3different search algorithm that has been discussed above:

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Compression Standards and Committee:

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Frame sequence:
In the field of video compression, a video frame is compressed using different algorithms with
different advantages and disadvantages, centred mainly around the amount of data compression.
These different algorithms for video frames are called picture types or frame types. The three
major picture types used in the different video algorithms are I, P and B. They are different in
the following characteristics:

I-frame:
An I-frame, keyframe, or intraframe consists only of macroblocks that use intra-prediction. Each
macroblock in an I-frame can only reference other macroblocks in the same frame. That is, you
can only use "spatial redundancy" within the frame for compression. Spatial redundancy is a term
used to refer to similarities between pixels in a single frame. I-frames appear in different avatars
of different video codecs such as IDR, CRA, and BLA frames, but the essence of these types of
I-frames is the same. Time predictions in I-frames are not allowed. I-frames have many uses and
will be discussed after the introduction of P-frames and B-frames.

P-frame:

P-frame is an abbreviation for Predicted Frame, which enables compression of macroblocks by

time prediction in addition to spatial prediction. P-frames use previously encoded frames for
motion estimation. Basically, each macroblock can be a time prediction or a P-frame.

B-frame:
B frame is a frame that can refer to the frame that occurs both before and after. For this reason,
B stands for Bi-Directional. If the video codec uses macroblock-based compression (like H.264
/ AVC), each macroblock in the B frame can be predicted using backward prediction (using
future frames). increase.
● Predict by forwarding prediction (using frames that occurred in the past)

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● Prediction without interim prediction-Intra only
● Completely skipped (using intra or inter prediction).
● B-frames also have the option of referencing and interpolating two (or more) frames
that occur before and after (in time dimensions), so B-frames reduce frame size and
video quality at the same time. Very efficient. time. They are very useful for video
compression because they can take advantage of both spatial and temporal redundancy
(future and past frames).

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KEY DELIVERABLES
Using the knowledge of different concepts of communication systems, we were able to simulate
the following tools using a GNU radio companion that works on Linux OS.

Laplace noise:

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Gaussian Noise

26
Impulse Noise:

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Low pass filter

High pass filter

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Band pass filter:

Different types of analog modulation techniques such as amplitude modulation, phase

modulation, and frequency modulation were performed.

Digital modulation techniques like ASK, FSK, and PSK were also tested.

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Some standard video compression techniques:
Video compression techniques are used to reduce the number of bits required to represent digital
video data while maintaining an acceptable fidelity or video quality. Their ability to perform this
task is quantified by the compression ratio. The higher the compression ratio is the smaller the
bandwidth consumption is.
Video is a sequence of images that are displayed in order. Each of these images is called a frame.
Since we cannot notice small changes in the frames like a slight difference in color, video
compression standards do not encode all the details in the video; some of the details are actually
lost. This is called lossy compression. It is possible to get very high compression ratios when
lossy compression is used. Whereas there are some compressions techniques that are reversible
or non-destructive compression. It is guaranteed that the decompression image is identical to the
original image. This is an important requirement for some applications where high quality is
demanded. This is called lossless compression.

When used to convey multimedia transmissions, video streams contain a huge amount of data
that requires a large bandwidth and subsequent storage space. As a result of the huge bandwidth
and storage requirements, digital video is compressed in order to reduce its storage or
transmitting capacity. This technology (video compression) reduces redundancies in spatial and
temporal directions. Spatial reduction physically reduces the size of the video data by selectively
discarding up to a fourth or more of unneeded parts of the original data in a frame. Temporal
reduction, Inter-frame delta compression, or motion compression, significantly reduces the
amount of data needed to store a video frame by encoding only the pixels that change between
consecutive frames in a sequence. Several important standards like the Moving Picture Experts
Group (MPEG) standard, H.261, 263, and 264 standards are the most commonly used techniques
for video compression.

H.261:
It was developed in 1990 by the International Telecommunication Union (ITU). H.261 standard
is used for data rates that are multiples of 64 Kbps. H.261 standard uses motion compensated
temporal prediction. It supports two resolutions, namely, Common Interface Format (CIF) with
a frame size of 352x288 and quarter CIF (QCIF) with a frame size of 172x144. The coding
algorithm is a hybrid of the following:

Inter-picture prediction:
It removes temporal redundancy transform coding, removes spatial redundancy motion
compensation, and uses motion vectors to compensate.

A macro block, the basic unit of temporal coding, is used to represent a 16x16 pixel region. Each
macro block is encoded using intra (I-coding) or predictive) P-coding. Motion prediction uses
only the previous picture to minimize the delay. H.261 is intended for carrying video over ISDN
in teleconferencing applications such as videoconferencing and videophone conversations.
H.261 is not suitable for usage in general digital video coding.

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H.263:
It was developed by the International Telecommunication Union (ITU) in 1996. It uses an
encoding algorithm called test model (TMN), which is similar to that used by H.261 but with
improved performance and error recovery leading to higher efficiency. It is optimized for coding
at low bit rates. H.263 provides the same quality as H.261 but with half the number of bits. A
block motion-compensated structure is used for encoding each picture into macroblocks. The
functionality of H.263 is enhanced by features like bi-directionally encoded B-frames,
overlapped-block motion compensation on 8x8 blocks instead of 16x16 macroblocks,
unrestricted motion vector range outside the picture boundary, arithmetic encoding, and
fractional-pixel motion-vector accuracy. H.263 supports three other resolutions in addition to
QCIF and CIF:

• SQCIF: Approximately half the resolution of QCIF

• 4CIF and 16CIF: 4 and 16 times the resolution of CIF

H.263 is like H.261, is not suitable for usage in general digital video coding. However, H.261
and 263 are a bit contradictory since they both lack some of the more advanced techniques to
really provide efficient bandwidth use.

H.263+:
It is an extension of H.263 with higher efficiency, improved error resilience, and reduced delay.
It allows negotiable additional modes, spatial and temporal scalability. H.263+ has enhanced
features like

• Reference picture re-sampling motion compensation and picture prediction

• Reduced resolution update mode that permits a high frame rate during rapid motion

• Independent segment decoding mode that prevents the propagation of errors from corrupt
frames

• Modified quantization mode improves bit rate control by controlling step size to detect errors
and reduce decoding complexity

MPEG-1:
The first public standard for the Moving Picture Experts Group (MPEG) Committee was the
MPEG-1. MPEG-1 was approved in November 1991 and its first parts were released in 1993. It
has no direct provision for interlaced video applications. MPEG frames are encoded in three
different ways:

• Intra-coded (I-frames): Encoded as discrete frames (still frames), independent of adjacent

frames

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 • Predictive-coded (P-frames): Encoded by prediction from a past I-frame or P-frame,
resulting in a better compression ratio (smaller frame)

• Bi-directional-predictive-coded (B-frame): Encoded by prediction using a previous and a

future frame of either I-frames or P-frames; offer the highest degree of compression

MPEG-1 decoding can be done in real time using a 350 MHz Pentium processor. It is also suitable
for playback from CD-ROM.

MPEG-2:
The MPEG-2 project was approved in November 1994, focused on extending the compression
technique of MPEG-1 to cover larger pictures and higher quality at the expense of higher
bandwidth usage. MPEG-2 is designed for digital television broadcasting applications that
require a bit rate typically between 4 and 15 Mbps (up to 100 Mbps), such as Digital high-
definition TV (HDTV), Interactive Storage Media (ISM) and cable TV (CATV). Profiles and
levels were introduced in MPEG-2. The profile defines the bit-stream scalability and the color
space resolution. With scalability, it is possible to extract a lower bit stream to get a lower
resolution or frame rate. The level defines the image resolution, the Y (Luminance) samples/sec,
the number of video and audio layers for scalable profiles, and the maximum bit-rate per profile.
The MPEG compatibilities include upward (decode from lower resolution), downward (decode
from higher resolution), forward (decode from previous generation encoding), and backward
(decode from new generation encoding). The MPEG-2 input data is interlaced, making it
compatible with the television scanning pattern that is interlaced.

The MPEG-2 is suitable for TV broadcast applications and high-quality archiving applications.
It is not, however designed for the internet, as it requires too much bandwidth.

MPEG-4:
It was approved in October 1998, and it enables multimedia in low bit-rate networks and allows
the user to interact with the objects. The objects represent aural, visual, or audio-visual content
that can be synthetic like interactive graphics applications or natural like in digital television.
These objects can then be combined to form compound objects and multiplexed and
synchronized to provide QoS during transmission. Media objects can be in any place in the
coordinate system. Streamed data can be applied to media objects to change their attributes.

The MPEG-4 compression methods are used for texture mapping of 2-D and 3-D meshes,
compression of time-varying streams, and algorithms for spatial, temporal, and quality
scalability, images, and video. Scalability is required for video transmission over heterogeneous
networks so that the receiver obtains a full resolution display. The MPEG-4 provides a high
coding efficiency for storage and transmission of audio-visual data at very low bit-rates. About
5-64 Kbps is used for mobile or PSTN video applications and up to 2 Mbps for TV/film
applications.

MPEG-7:

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It was approved in July 2001 to standardize a language to specify description schemes. The
MPEG-7 is a different kind of standard as it is a multimedia content description standard and
does not deal with the actual encoding of moving pictures and audio. With MPEG- 7, the content
of the video is described and associated with the content itself, for example to allow fast and
efficient searching in the material.

The MPEG-7 uses XML to store metadata and it can be attached to a timecode in order to tag
particular events in a stream. Although, MPEG-7 is independent of the actual encoding technique
of the multimedia, the representation that is defined within MPEG-4, i.e., the representation of
audio-visual data in terms of objects, is very well suited to the MPEG-7 standard. The MPEG-7
is relevant for video surveillance since it could be used for example to tag the contents and events
of video streams for more intelligent processing in video management software or video analytics
applications.

H.264/AVC:
In early 1998, the Video Coding Experts Group (VCEG) ITU-T issued a call for proposals on a
project called H.26L, with a target of doubling the coding efficiency in comparison to any other
existing video coding standards for various applications. The Moving Picture Expert Group
(MPEG) and the Video Coding Expert Group (VCEG) have developed a new and outstanding
standard that promises to outperform the earlier MPEG-4 and H.263 standard. Even though the
first draft design for the new standard was adopted in October 1999, it provides the most current
balance between the coding efficiency, cost and implementation complexity.

It has been finalized by the Joint Video Team (JVT) as the draft of the new coding standard for
formal approval submission referred to as H.264/AVC and was approved by ITU-T in March
2003 (known also as MPEG-4 part 10). The standard is further designed to give lower latency as
well as better quality for higher latency. In addition, all these improvements compared to previous
standards were to come without increasing the complexity of design so much that it would be
impractical or expensive to build applications and systems. An additional goal was to provide
enough flexibility to allow the standard to be applied to a wide variety of applications: for both
low and high bit rates, for low- and high-resolution video and with high and low demands on
latency.

The main features that improve coding efficiency are the following:

• Variable block-size motion compensation with the block size as small as 4x4 pixels

• Quarter-sample motion vector accuracy

• Motion vectors over picture boundaries

• Multiple reference picture motion compensation

• In-the-loop deblocking filtering

• Small block-size transformation (4x4 block transform)

• Enhanced entropy coding methods (Context- Adaptive Variable-Length Coding

(CAVLC) and Context Adaptive Binary Arithmetic Coding (CABAC))
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COMPARISON OF VIDEO COMPRESSION METHODS
Video compression standards provide a number of benefits, of which the foremost is ensuring
interoperability, or communication between encoders and decoders made by different people or
different companies. In this way standards lower the risk for both consumer and manufacturer
and this can lead to quicker acceptance and widespread use. In addition, these standards are
designed for a large variety of applications and the resulting economies of scale lead to reduced
cost and further widespread use. The well-known families of video compression standards, are
shown in Table 1 (Current and Emerging Video Compression Standards) performed under the
auspices of the International Telecommunications Union-Telecommunications (ITU-T, formerly
the International Telegraph and Telephone Consultative Committee, CCITT), the International
Organization for Standardization (ISO) and the Moving Pictures Expert Group (MPEG) which
was established by the ISO in 1988 to develop a standard for compressing moving pictures
(video) and associated audio on digital storage media. The first video compression standard to
gain widespread acceptance was the H.261 standard. The H.261 and 263 standards are suitable
for carrying video over ISDN. They are used for video delivery over low bandwidths. The MPEG
standards provide a range of compression formats that are suitable for applications that require
higher bit rates. The MPEG-1 provides compression for standard VHS quality video
compression. The MPEG-2 meets the requirements of applications with bit rates up to 100 Mbps
and can easily cater for digital television broadcasting applications.

Table 1: Current and emerging video compression standards

Table 2: Comparison of main coding tools in MPEG-2, MPEG-4 Part 2 and H.264/
AVC

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MPEG-1 and 2 are used for broadcast and CD-ROM applications, but unsuitable for the Internet.
The MPEG-4 is suitable for low bit-rate applications such as video conferencing as it provides a
high coding efficiency for storage and transmission. The MPEG-4 applications include Internet
multimedia, interactive video, video conferencing, videophone, wireless multimedia and
database services over ATM networks. H.263 and MPEG-4 are used for video delivery over low
bandwidths. To cater for the high bandwidth requirements for the Internet, codes must have high
bandwidth scalability, lower complexity and tolerance to losses, as well as lower latency for
interactive applications. MPEG-7 addresses this problem as it caters for both real-time and non-
real time applications and enables retrieval of multimedia data files from the Internet. If the
available network bandwidth is limited, or if a video is to be recorded at a high frame rate and
there are storage space restraints, MPEG may be the preferred option. It provides a relatively
high image quality at a lower bit-rate (bandwidth usage). Still, the lower bandwidth demands
come at the cost of higher complexity in encoding and decoding, which in turn contributes to a
higher latency when compared to motion H.264/AVC. H.264/AVC is now a widely adopted
standard and represents the first time that the ITU, ISO and IEC have come together on a
common, international standard for video compression. H.264 entails significant improvements
in coding efficiency, latency, complexity and robustness. It provides new possibilities for
creating better video encoders and decoders that provide higher quality video streams at
maintained bit-rates (compared to previous standards), or, conversely, the same quality video at
a lower bit-rate. Table 2 shows a comparison of the main coding tools in MPEG-2, MPEG-4 Part
2 and H.264/ AVC.

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CONCLUSIONS
Video compression is gaining popularity since storage, and network bandwidth requirements are
able to be reduced with compression. Many algorithms for video compression which are designed
with a different target in mind have been proposed. This study explained the standardization
efforts for video compression such as H.261, 263, and 263+, MPEG-1, 2, 4, 7, and H.264. Most
recent efforts on video compression for video have focused on scalable video coding. The
primary objectives of ongoing research on scalable video coding are to achieve high compression
efficiency, high flexibility (bandwidth scalability), and/or low complexity. Due to the conflicting
nature of efficiency, flexibility, and complexity, each scalable video coding scheme seeks trade-
offs on the three factors. Designers of video services need to choose an appropriate scalable video
coding scheme, which meets the target efficiency and flexibility at an affordable cost and
complexity.

POTENTIAL OUTCOMES
Using the above filters and modulations, an FM receiver and Video transmitter and
receiver were simulated as well.

FM RECEIVER:
Frequency modulation is a type of radio modulation technique where information is added to the
carrier signals and changed by varying the frequency of a transmitted signal. Information is
delivered in the form of voice or music. It is commonly used for radio signals greater than 30
MHz. In general, frequency modulation (FM) is defined as modulation techniques where the
information-containing signals are encoded in a carrier wave by varying the instantaneous
frequency of the wave.

Blocks used:

1. Options - It is used to select the GUI standard, whether QT or WX.

2. Variable - It is used in order to set the sample rate constant through each block.

3. UHD: USRP Source - UHD stands for USRP hardware driver software which specifies
various properties. The input data to the UHD USRP Source can be a float or a complex
data type depending on the data streams.

36
4. Low Pass Filter - It is used to avoid the passage of unwanted high-frequency signals.

5. WBFM Receive – It performs FM demodulation.

6. Rational Resampler – It is used to either increase or decrease the sampling rate.

7. Audio Sink – It is used as a speaker, which helps in the proper audibility of audio signals.

8. QT QUI Sink – It provides a complete representation of signals in terms of time domain,

frequency domain, waterfall display, and constellation display.

9. QT GUI Range – It creates a variable with a slider. The variable must be a value with a
real integer number.

37
GNU Radio Companion Flowgraphs:

38
Results:

Frequency spectrum of the received signal

Things to be noted while performing this experiment:

1. The proper positioning of the antenna is needed when receiving signals from the FM
station.
2. Experiment must be performed in a network area zone.
3. Tune the FM station which is currently working or active.

VIDEO TRANSMITTER AND RECEIVER:

We will be using gnu radio companion to perform a simple simulation on transmission and
receiving the video file from one remote place to the other.

39
Model-1(Without using USRP):

40
Model-2(Using USRP):

41
This experiment's system model is broken into two sections.

The first half of the GNU implementation is about the transmitter side, and the second portion is
about the receiving side.

Software for radio stations. The system's input is a live video feed.

GNU Radio software is used to process the signal. The code is developed in the Python
programming language, which is used to power the GNU Radio's building components. The
signal is wirelessly conveyed.

Dipole antennas are used. These antennas work in the range of the frequency ranges between 0.4
and 4 GHz where both the transmitter and the receiver are linked to the USRP device side of the
receiver.

Transmission End:

Receiver End:

42
Results:

The distance between the transmitter and receiver antenna is 1 meter, and the data speed is 1000
kbps. The packet error rate = 2.21%. The RF gain of the received signal is 6db, and the carrier
frequency is 1.2345 GHz.

43
Live Video Transmission and Video Compression
Implementation
We have Used the Gstreamer tool for Live video capturing and Compressing it.
GStreamer is a library for constructing graphs of media-handling components. It is a pipeline-
based multimedia framework that links together a wide variety of media processing systems to
complete complex workflows

In this We are taking Live Video from webcam in HD format (1280*720) and used H.264 codec
compressing simultaneously it to output.ts

Then this same Output file is used as input for our transceiver in GNU Radio Companion.

We can clearly see that Video size has decreased significantly with minimal change in quality of
video.

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TIME TABLE

From 01-06-2022 to 07-06-2022 We went through the resources shared.

From 08-06-2022 to 15-06-2022 We Implemented different modulation techniques.

From 16-06-2022 to 21-06-2022 We implemented video transmitter and receiver

From 22-06-2022 to 20-07-2022 We implemented H.264 compression algorithm

45
BIBLIOGRAPHY
https://www.gnuradio.org/grcon/grcon18/presentations/GNU_Radio_Support_for_Real-
time_Video_Streaming_over_a_DSA_Network/Debashri_DSA.pdf

https://research.ijcaonline.org/icaet2016/number4/icaet056.pdf

http://www.ettus.com/

https://web.archive.org/web/20190128075519/https://wiki.gnuradio.org/index.php/Main_Page

Software-defined radio - Wikipedia

Software Defined Radio - an overview | Science Direct Topics

10 Popular Software Defined Radios (SDRs) of 2022 (bliley.com)

https://www.vocal.com/video/video-compression-technology/

https://homepages.inf.ed.ac.uk/rbf/CVonline/LOCAL_COPIES/AV0506/s0561282.pdf

Video Coding Standards - from H.261 to MPEG1,2,4,7 - to H.265 MPEG-H - YouTube

https://ottverse.com/i-p-b-frames-idr-keyframes-differences-usecases/

https://en.wikipedia.org/wiki/Video_compression_picture_types#:~:text=An%20I%E2%80%91
frame%20(Intra%2D,movements%20need%20to%20be%20encoded .

H.264 over GNU Radio and USRP

https://ffmpeg.org/

Video Compression Techniques: An Overview (scialert.net)

Books:
1. Communication Systems 4e by Simon Haykins.
2. Practical Approach to Software Defined Radios by Ram and Amitesh Pandey

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