Thesis
Defense
Olufunke Olaleye
Symbiotic Audio Communication
on
Interactive Transport
Technology Overview
 Today's Internet traffic contains both audio and
data packets.
 Sensitive traffic (audio) shares bandwidth with
other non-sensitive traffic as shown fig 1.1
 VoIP
 Audio Chat
 Online Radio
 Real time internet lecture
 Real time Internet conference
 Online Music service
 Internet News
Figure 1.1 Bandwidth Limited Network
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Growth of Audio
While, the technology has moved many businesses online, the
use of audio traffic over the Internet has growth exponentially.
 However, audio perception is highly
susceptible to disturbance in temporal
quality.
 Packet loss, delay and jitter affects
quality of service during congestion.
Global VOIP Growth
Subscribers
# of Subscribers
150000000
100000000
50000000
To maximize the audio/voice quality,
some algorithms proposed to adapt:
0
2005
2006
2007
2008
2009
Year
GLOBAL VoIP GROWTH
2005
Total
Subscribers
2006
2007
2008
24,043,303 47,346,874 81,618,331 111,209,271
Growth %
Net New
Subscribers
67
97
2009
133,633,938
72
36
20
9,682,349 23,303,571 34,271,457
29,590,940
22,424,668
 Size of the playout buffer
 Coding rate
 Packet path diversity [25]
Challenge  Receiving feedback from
the network about congestion.
Source: Infonetics Research, February 2006
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Research Goal of Thesis
 A well-engineered, end-to-end network is necessary to transmit audio over the Internet.
Goal of this thesis:
Reduce the delay and jitter faced by audio traffic in the network during congestion.
 An efficient solution: Sender’s end (encoder) ability to sense the state of the network and
react accordingly.
R &D:
 Developed symbiotic perceptual audio streaming mechanism that is capable of detecting the
underlying bandwidth of the network and modify the target bit rate to suit the network
condition.
 Combines the quantization technique to accurately represent the audio signals without
distortion.
Proposed TCP Interactive (iTCP)
--- Operationally state equivalent to the conventional TCP except applications.
--- Optionally subscribe and receive selected local end-point protocol events in real-time.
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The Human Auditory Perception
 Human ear perceives signal within the
range 20 and 20000 Hz.
 High sensitivity is between 2.5 and 5
kHz and decreases beyond and below this
frequency band.
 Two Principle Perception is based on
 Threshold in quiet --- Sensitivity.
 Masking threshold --Figure 2.2 Threshold in quiet and masking threshold
 Temporal
 Simultaneous masking
 Human auditory system perceives
signal in a non-equal width sub-band
called critical bands, it’s unit is barks
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Impact of Threshold in Quiet
Threshold in Quiet
Quantized values without
adaptation
 Signal strength
below the threshold in
quiet are inaudible to the
human ear.
Quantized values with 25%
adaptation
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Impact of Masking Threshold
Quantizes value with a single masker – 37.5%
Quantized values with multiple maskers ---50%
Final Output
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Effect of Simultaneous &
Temporal Masking
 Masking threshold -- Temporal
 Simultaneous masking
Effect of Multiple Signals
Effect of Multiple Signals
Sound
Presure (dB)
Sound
Presure (dB)
200
200
100
25
Time in ms
Signal A
Signal B
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Signal C
Signal D
Signal E
Signal A
375
325
225
175
Signal A
125
Signal C
275
375
325
275
225
175
75
25
Signal A
125
Signal C
0
Signal E
0
75
Signal E
Time in ms
Signal B
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Signal C
Signal D
Signal E
8
Audio Adaptation
 Audio signal simultaneously passes though the
hybrid filter bank and psychoacoustic model.
 The hybrid filter bank
 Divides the input signal into chucks
of 576 samples called granule and sub
bands of frequency.
 Provides a specified mapping in
time and frequency.
Figure 2.5 Block Diagram of the encoder
 The psychoacoustic model behaves like the human auditory system.
 Computes a just noticeable noise level in each subband.
 Determines the block (window) type --- (short/long).
 Computes the energy in each partitions band (threshold calculation partition).
 Convolves the partitioned energy by applying the spreading function (frequency spread of masking).
 Apply pre-echo control using some constants (2 or 16).
 Compares the threshold with the last threshold and quiet threshold and takes the maximum.
 Converts threshold calculation partition to scalefactor bands and calculates the signal to mask ratio.
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Tables for Threshold calculation
partitions
Threshold calculation partitions is computed with following parameters: width, minval, threshold in quiet, norm and bval:
(44.1kHz sampling rate---long)
no.
0
1
2
5
6
7
12
15
16
60
FFT-lines
1
1
1
1
1
1
1
2
2
36
minval
24.5
24.5
24.5
20
20
20
18
12
6
0
(44.1kHz sampling rate---short)
qthr
norm
bval
4.532
0.951
0
4.532
0.7
0.431
4.532
0.681
0.861
0.09
0.665
2.153
0.09
0.664
2.584
0.029
0.664
3.015
0.009
0.578
5.057
0.018
0.856
6.422
0.018
0.846
7.026
32.554 0.483 23.897
Parameters for computing the SNR, for short window, it is read from a
table.
Norm is normaling constant for each sub band
Bval is bark value.
For low freq, the strength of the masking is limited by minval
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no.
0
1
2
5
6
12
13
14
37
FFT-lines
1
1
1
1
1
1
1
1
7
qthr
4.532
0.904
0.029
0.009
0.009
0.009
0.009
0.009
6.33
norm SNR (db)
0.952
-8.24
0.7
-8.24
0.681
-8.24
0.665
-8.24
0.664
-8.24
0.578
-7.447
0.541
-7.447
0.575
-7.447
0.57
-5.229
bval
0
1.723
3.445
7.609
8.71
13.21
13.748
14.241
23.828
bark: a non-linear frequency scale.
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Tables for converting threshold
calculation partitions to scalefactor
bands
There are 21 bands at each sampling frequency for long windows and 12 bands each for short windows.
(44.1kHz sampling rate---long)
no. sb
0
1
2
5
6
12
13
14
20
cbw
3
3
4
1
3
4
3
3
2
bu
0
4
7
17
18
36
40
43
59
bo
4
7
11
18
21
40
43
46
61
w1
1
0.944
0.389
0.861
0.083
0.18
0.9
0.532
0.278
bo is the first index value of cbw
 bu is the last index value of cbw
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w2
0.056
0.611
0.167
0.917
0.583
0.1
0.468
0.623
0.96
no. sb
0
1
2
5
6
9
10
11
(44.1kHz sampling rate---short)
cbw
2
2
3
5
3
3
3
2
bu
0
3
5
15
20
30
33
36
bo
3
5
8
20
23
33
36
38
w1
1
0.833
0.167
0.833
0.75
0.625
0.7
0.833
w2
0.167
0.833
0.5
0.25
0.583
0.3
0.167
1
The number of partitions (cbw) converted to one
scalefactor band (excluding the first and the last partition).
The threshold calculation partitions are converted directly to
scalefactor bands. The first partition which is added to the
scalefactor band is weighted with w1, the last with w2
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Audio Adaptation
The noise allocation block
 Uses the output of the psychoacoustic
model, “signal to mask” ratio.
 Used the noise level in noise allocation
to determine the actual quantizers and
quantizer levels.
 Determines how to allocate the number
of code available for quantization of
subband signals.
It uses two nested iteration loops.
The spectrum (frequencies) are broken into "scalefactor bands".
Thes bands are determined by the MPEG ISO spec.
In the noise shaping/quantization code, we allocate bits among the
partition bands to achieve the best possible quality
 Distortion control loop (outer loop)
 Rate control loop – (inner loop)
quantize in an iterative process
 The inner loop quantizes the input signal and increases the quantizer step size until
the output can be coded with the available amount of bit.
 After completion of the inner loop, an outer loop checks the distortion of each
scalefactor band, if the allowed distortion is exceeded, it amplifies the scalefactor
band and calls the inner loop again.
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Audio Adaptation
 If the overall bit sum is less
than the available bits to encode
a frame.
 The best quantized values are
coded by Huffman coding to
further reduce their space
requirement.
The bitstream formatting block
assembles and formats quantized
subband signal using Huffman
code and other side information
into bitstream.
This is then passed into the
network for transmission.
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TCP Congestion Control

TCP provides a connection-oriented, reliable delivery of
data streams between two applications or hosts

TCP uses two mechanism to detect network congestion
i.
Retransmission timer time out.
ii.
Duplicate ACKs.
Congestion Control Algorithms

Slow-start and congestion- avoidance.

Fast-retransmit and fast-recovery.
Figure 3.1 Slow Start/Congestion
Avoidance (SSCA) mechanism.
Figure 3.2 Fast Retransmit/Fast Recovery
(FRFR) mechanism
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TCP Congestion Control Internal
Events
}
Subscribable
Events
Event
Denotation
Explanation
1
Retransmission time out
2
5
New ACK received
snd_cwnd reached the slow
start threshold ssthresh
Third duplicate ACK
received
Fourth (or more) duplicate
ACK received
Congested network / Lost segment.
Increment snd_cwnd exponential or
linearly.
Switch snd_cwnd increment from
exponential to linear.
6
New ACK received
3
4
SSCA
FRFR

Sub



Lost segment, execute fast retransmit.
A segment left the network; transmit a
new segment.
Retransmitted segment arrived at the
receiver and all out of order segments
buffered at the receiver are acknowledged





Table 3.1. TCP Congestion Control Internal Events
 In this thesis, for simplicity, we use event 1, retransmission
timer time out.
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iTCP: interactive TCP
User Space
 An application that has subscribed
to the kernel also binds a T-ware to a
selected TCP event through the
subscription API as shown by line 1
and 2 of figure 4.
7
Application
1
T-ware
T-ware
T-ware
[1]
[2]
[n]
5
TCP
Connection
Signal Handler
6a
4a
System
Socket
API
Probing API
Subscription
API
3a
2
6b
4b
TCP Kernel
3b
Event
Monitor
Event
Informatio
n
Connectio
n State
Figure 4. The iTCP extension and API.
 When the TCP state changes as a
result of congestion in the network, it
also causes an event to occur in the
TCP kernel.
 The event monitor is aware of the
changes that occurred; thus, it
responds instantly by sending a signal
in (3a) to the signal handler and also
stores the event information in (3b).
 The signal handler catches the
signal from the kernel and requests
the event type (4a, 4b) from the
kernel via the probing API.
 The appropriate T- Transientware
Modules (or T-ware) (5) is triggered
to serve the particular event.
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Symbiosis Throttling Model
 During congestion, detected by the time-out event ( ζ = 1 ),
the model reduces the target bit rate to considerable lesser or
minimum rate.
 bmax, target bitrate at a normal state
 bmin, the minimum acceptable rate
Figure 5.1 Symbiosis throttling Model (Input Rate)
 Reduction ratio = rate retraction ratio
= ρ = b min/ bmax
(5.1)
 “Running generation threshold” function = link between the
underlying TCP and the model.

1
gT (t  1) when   1
2
 gT (t  1) otherwise
gT (t ) 
(5.2)
Figure 5.2 Symbiosis throttling Model (Output Rate)
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Symbiosis Throttling Model
 Running control generation function” b(t) =
b(t )   . bmax when   1
(5.3)
1
gT (t  1)
2
, b(t  1)  1] when   1 & b (t  1)  gT (t  1)
when   1 & b(t ) 
 2 . b(t  1)
 min[ bmax
Figure 5.1 Symbiosis throttling Model (Input Rate)
 Rs = reservoir size
 Estimated Target Buffer Fullness

ETBF  0.9 * T (t )  R
s
 1
. 1152 / freq)  Z ) / 2 when   1
T (t )  ((bmax . 1152 / freq)  Z ) / 2 when
 ((  . bmax
(5.4)
(5.5)
 A = Actual number of bits per granule
Rs  T  A
Figure 5.2 Symbiosis throttling Model (Output Rate)
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Symbiosis Throttling Model
Input
Source audio
T-ware
xr
xr
Perceptual Model
Event
b
b (t), ρ
Reservoir
Rs
ratio(sb)
Noise Allocation
Compute Allowed
Distortion
T
T (t), xr
Estimated Target
Buffer Fullness
0.9 * T , Rs
xmin(sb)
0.9 * T (t), Rs
Amplify violated
(xmin – xfsf) of
masking threshold
scalefactor band j < 0
ok
Compute
Distortion in
Quantization
xfsf(sb)
ix
Quantization of
actual energy
xr  ix
ix
Quantization of
amplified energy
xr  ix
violated
xr (i) = xr (i) * ifqstep
xfsf(sb)
Code
Information
Quant Compare
Compute
Distortion in
Quantization
Rs
Output audio stream

The best quantization with allowed distortion to sharp the noise.
ifqstep = sqrt(2) ^ ((1 + scalefac_scale) * ifq(scalefactor band))
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Symbiosis Mechanism:
The T-ware
Signal handler
• Catch the signal from the kernel
and invoke the appropriate Tware.
• Encoder subscribe with the
“retransmit timer out" event only.
• T-ware calculates a frugal
state target bitrate base
on the retraction ratio
• Stores the reduced rate
in the “rate.par” file.
 The mechanism use to reduce
delay is called T-ware.
 It gives the transport layer the
ability to communicate with the
application layer (encoder).
 The key element is the loss event
handler that generates a signal.
 The signal information is used to
probe iTCP service.
 Couple with the retraction ratio
ρ , the rate is reduced to achieve the
objective of the thesis.
Recovery-T-ware
• The recovery T-ware kicks in at the
Trecovery time
• Returns the encoder bit rate to
normal rate.
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Figure 6a & 6b Pseudo code of the
Signal Handler and the Recovery
handler
20
Experiment Set Up
Figure 7.1 Experiment setup
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Experiment (Test Samples)
TEST ( 3 types of audio sound quality)
 High quality music (HighQmusic),
 Low or Poor quality music (LowQmusic)
 Speech mixed with music (SpeechMusic)
3 Running Mode
 iEXP
 iOFF
 Classic
Table 7.1 Experiment control flags and running modes
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Experiment and Performance
Analysis
The parameters used in the experiments:
(i)
Predetermined rate retraction ratio ( ρ = 0.50)
(ii)
Bit rate
To compute b(t), the rate retraction ratio is multiplied by the
current bit rate. (Bit rate * ρ )
 Frames information are collected for the first 800 up to
1200 audio frames of the encoder and the player.
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Performance Analysis
 Performance of iEXP is better than the
classic TCP.
 The delay buildup in Classic and iOFF
are much higher than that experienced by
the iEXP (iTCP) as indicated by the step
jump.
 The step jump of iEXP is much smaller
as a result of the rate retraction ρ
 iEXP was able to recover from delay
buildup in few seconds compare to the
Classic and iOFF.
Figure 7.2. Frame arrival delay on the three audio qualities
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Performance Analysis
Assuming a packet arrives at the receiver
end at time tj
 Expected arrival time for the packet is ej.
Referential jitter (refJitter(j)) = (tj - ej)
 refjitter(j) is negative if packet j arrives
early at the receiver end.
It can be buffered and played at the
actual time.
 refjitter(j) is positive, if packet j has
arrived late at the receiver end.
 The player will pause and wait for
packet that arrive late.
 The higher the delay, the higher the jitter
experience by the frames.
Figure 8.3. Referential Jitter on the three audio qualities
March 21, 2007
 The step jumps in the iEXP were much
smaller than those in TCP-classic and iOFF.
 Furthermore, this indicates that the
interactive TCP reduces jitter.
Symbiotic Audio Communication
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on Interactive Transport
Performance Analysis
 Symbiotic rate reduction that
occurred as a result of the rate
modification between the rate controller
of the encoder and the symbiosis unit
 Target bits and the actual bits
generated by the encoder for each
frame.
 The rate retraction ratio of the
symbiosis kicks in when a time out
event is triggered and reported by iTCP
during congestion.
 The effect is observed on the plots
as number of bits drop in accordance
with the rate retraction ratio.
Figure 7.4. Symbiotic Rate Reduction on the three audio qualities
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Performance Analysis
Comparison of frame delay and
acceptance ratio
Table 7.2 Average Frame Delay and acceptance ratio
 A delay tolerance of d = 2, 4, and 6
seconds are introduced to measure the
average frame delay and acceptance
ratio.
 iEXP mode experience a low delay
and high acceptance ratio.
 iTCP’s T-ware mechanism allowed
the application to use sophisticated
techniques to control the temporal
qualities of its traffic.
Low delay & High acceptance ratio.
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Performance Analysis
Overall stream compression
 The overall delivery bits in the iEXP
mode reduce to 80 – 90% of the original
bits
Table 7.3 Percentage of total bits delivered
for each mode
 iOFF and Classic cases shows no
adaptation.
 The file size of iEXP mode reduce
significantly compared to the other modes.
Table 7.4 Sizes of the audio files
 The audio quality between the sending
end and the receiving end is achieved
perceptually by the temporal and spectral
resolution.
 iTCP provides a means of tradeoff of
terrible frame delay for a satisfactory
reduction of quality.
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Performance Analysis
iOFF
iExp
 Aim of the iOFF mode --- Study the overhead introduced
by the event notification service.
classic
18
16
 Increase total transmission time for all modes.
14
12
10
 iOFF mode is higher than the classic TCP mode due to
event notification service enabled.
8
6
4
 iEXP mode is much smaller than the other modes
2
0
HighQmusic
Low Qmusic
SpeechMusic
Indicates the application level performance
outweighs the overhead.
Figure 7.5 Overhead of the interactivity
service
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Conclusions
Aim:
Reduce delay and jitter of audio traffic.
Solution:
Design an interactive and friendly application that receives the state
feedback from the network – Symbiotic encoder.
Achievements:
Dynamic reduction of jitter and delay of time sensitive traffic during
congestion.
 Network congestion reduction by reducing the traffic at the source.
 Trade off quality for delay and jitter.
 The approach is simple and does not alter any network dynamic to be
optimal (e.g fair queuing).
 Its effect is entirely on the application layer.
 It also further validates interactive transport control protocol (iTCP)
usefulness and the efficiency through the idea of the event notification.
Technically, this scheme is not applicable to non-elastic traffic such as simple file transfer.
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Future Work
Optimization:
The parameters in the scheme are user defined but can be optimized with the
symbiotic throttling model in [13].
The experiment can be performed using the other subsribable events(i.e.
duplicate ACK).
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Questions and Comments
March 21, 2007
Symbiotic Audio Communication
on Interactive Transport
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