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Multiple Descriptions Coding inMELP Coder for Voice over IP
M.SAIDI, B.BoudraaLaboratoire de communication parle et de
traitement du
signal (LCPTS), universit USTHBAlgiers, Algeria
[email protected], [email protected]
M.Bouzid, M.BoudraaLaboratoire de communication parle et de
traitement du
signal (LCPTS), universit USTHBAlgiers, Algeria
[email protected], [email protected]
Abstract In VoIP systems, CELP coders, such as G.729, are
commonly used as they offer good speech quality in the absence of
packet loss. However, harmonic coders such as MELP may be a good
alternative for VoIP due to their higher resilience to packet loss.
In this paper we examine the problem of the packet loss in the VoIP
application using MELP coders. The proposed packetization scheme
based on Multiple Description Coding (MDC) applied to the MELP
coder is presented. A packet will contain information on two MELP
coders operating at 2.4 and 1.2 Kbps respectively. The
packetization is achieved using 135 bits in 22.5 ms corresponding
to a total rate of 6 kbps. The results show that under typical VOIP
operating conditions, the method performs well and outperforms CELP
coders operating without MDC at 8 kbps.
Keyword; VoIP, MELP, MDC, packet loss.
I. INTRODUCTION
Voice over Internet (VoIP) is achieving by sending and receiving
packets. At the receiver, some packets are missing because of
delays, congestions or transmission errors. This packet loss
degrades the quality of service (QoS) at the receiver. As the voice
is transmitted in real time, the receiver cannot request the
retransmission of lost packets due to the large transfer time
induced. Concealing loss techniques are then used at the
transmitter or the receiver in order to recover the loss packets.
These techniques are called redundant descriptions [1]. Among these
techniques, Multiple Description Coding (MDC), based on information
redundancy, increases the robustness against packet loss. The goal
is to keep a certain quality and intelligibility of speech when the
packet loss rate increases. Real-time Transport Protocol (RTP) is
often used.In this work, we have developed and implemented a
packetization scheme made frame by frame and applied on a harmonic
coder designed for voice over IP. Hence, this packetization is
achieved on a mixed excitation linear prediction (MELP) [2]. Each
packet contains major information on the current frame and its
future neighbouring. A received packet can recover until four lost
frames.
This paper is organized as follow. First, we proceed in the
voice communication over IP, then we propose a method for
recovering lost packets. Afterward a Packetization scheme is
proposed, and then a packet recovery is presented. Finally, a
simulation results are shown and evaluation.
II. VOICE OVER INTERNET PROTOCOL (VOIP)
It is widely accepted that VoIP is a technology emerging on the
internet and will dominate the field of voice communications. It
has begun to take place as a viable alternative to the traditional
voice communication systems. In this technique, the analog call is
converted to digital and compressed to be then transformed into
packets for transmission over an IP network. At the other end, the
process is reversed: As the packets of information circulating on
the Internet take different paths and arrive frequently out of
order, the packets are firstly stored in buffers to be re-sequenced
and then decompressed and transformed into a sound signal before
being routed back through the ordinary telephone equipment [5, 6].
Figure 1shows the block diagram of a voice communication system
over IP network.
Fig. 1: Block diagram of the transmission of voice over IP
IP network
Packetization
De-packetizationDecoding
CodingOriginal speech
Synthetic speech
978-1-4673-1591-3/12/$31.00 2012 IEEE
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III. OUR PROPOSED METHOD
In this work, we propose a method for combating efficiently
packet loss. The method is based on MDC technique, including
redundancy. Indeed, we code the signal on two descriptions in the
same packet. The first uses a MELP coder and encodes the current
frame Fn at 2.4 kbps while the second uses another MELP encoder
running at 1.2 kbps to encode the three frames Fn+1, Fn+2, Fn+3,
following Fn. Obviously, Fn is used to reconstruct the signal with
a good quality while Fn+1, Fn+2and Fn +3 are used to recover the
eventual loss packets. Indeed, the second description thus formed
contributes to reconstruct the speech when one, two, three or even
four successive packets are lost. This redundant information has
not the same quality of the extracted one from Fn as it is roughly
quantified. It only helps to reconstruct an intelligible speech
when packets are lost. The packetization scheme is shown in figure
2. Note that the MELP 2.4 operates on a frame of 22.5ms, while the
MELP 1.2 operations are achieved on a 67.5 ms frame [3, 4].
Fig. 2: Packetization using two descriptions
IV. PACKETIZATION
MELP coder parameters are given in table 1. The two coders
encode the fundamental frequency (pitch), the flag of the
aperiodicity, the five bands of voicing, two gains corresponding to
the energy of two half-frames, ten LPC coefficients converted into
LSF and spectral amplitudes of ten harmonics of the pitch.
A fine description at 2.4 kbps is required in order to provide
good speech quality in no-error conditions. The configuration
allows also a coarse description at 1.2 kbps with reasonable
quality to recover until three successive packet losses for larger
bursts. The packetisation scheme is shown in Figure 2. A packet
will contain both mentioned MELP coders and will be coded using 135
bits (54 + 81) corresponding to a rate of 6kbps. Hence, formation
of a packet requires the presence of four successive frames of 22.5
ms each. In a transmission without packet loss, only the first 54
bits will be used to decode the signal. Then, a packet is
attributed to the current frame corresponding to the MELP 2.4
(Figure 2). This causes a delay of 22.5 ms. Note that forming and
sending the first packet request a delay of 90 ms. Afterwards,
every 22.5 ms a packet is sent.
Bit Rate MELP 2.4 kbps MELP 1.2 kbps
Sampling frequency Size of frame
Bit Rates of frame
8 kHz180 samples (22.5 ms)44,44 frames/seconde
8 kHz3*180 samples (67.5 ms)14.8148 frames/seconde
Mode of voicing V N/V VVVUVVVUV VVU
UUV UVU VUU
UUU
10 LSFs 25 25 43 43 39 43 27Pitch 7 7 12 12 12 12 1210 Fourier
amplitudes 8 - 8 8 8 8 05 Bands of voicing 4 - 6 4 4 2 02 Gains 8 8
10 10 10 10 10
Flag 1 0 1 1 1 1 0Protection - 13 0 2 6 4 31Synchronisation 1 1
1 1 1 1 1
Total bits per frame 54 bits 81bitsTotal 54*44,44= 2400 bps
81*14.8148 = 1200 bps
Table I: Bit allocation encoder MELP 2.4 kbps and 1.2 kbps
Packet
MELP1.2kbps
MELP 2.4 kbps
Fn Fn+1 Fn+2 Fn+3
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V. PACKET RECOVERY
Figure 3 shows how our MDC system allows recovering lost
packets. Using this scheme, three successive lost packets can be
easily recovered. Even the fourth frame can be retrieved by
adopting a method of extrapolation [2]. From left to right on the
same figure, the respective cases of loss of a single packet, 2
packets, 3 packets and finally 4 packets are shown.
Explanation: The first case corresponds to a single packet loss
(F2 frame).The current packet is lost and the MELP 2.4 is unable
toprovide us with a good quality of the speech. The system falls
back to the previous received packet containing the information the
lost current frame. This frame is then reconstructed with a coarse
quality using the 81 bits corresponding to the 1.2 MELP coder; i.e.
to the three framesF2, F3 and F4.
The second case corresponds to two successive loss packets,
namely F4 and F5 (figure 3). As in the first case, we cannot
reconstruct the signal using a 2.4 MELP. So, we proceed with
recovery using the previous packet which was received
correctly.
In the third case, three successive packets corresponding toF7,
F8 and F9 are lost. The decoding of the signal is obviously
achieved from the last received packet.
In the latter case, when four successive packets (F11, F12,
F13and F14) are lost, then F11, F12 and F13 will be directly
recovered from the packet No. 10, as this packet contains
bothinformation on the frame 10 at 2.4 kbps and information
aboutsuccessive frames F11, F12 and F13 which were coded at 1.2kbps
in this packet. The frame F14 will be retrieved using an
extrapolation method as in [2].
Fig. 3: Process of recovery of lost packets based on the
proposed MDC
22.5 ms Original speech
Synthetic speech
2.4 kbps 1.2 kbps 2.4 kbps 1.2 kbps 1.2 kbps 2.4 kbps 1.2 kbps
1.2 kbps 1.2 kbps 2.4 kbps
F1 F2 F3 F4 F5 F6
F2 F3 F4 F3 F4 F5 F4 F5 F6 F5 F6 F7 F6 F7 F8 F7 F8 F9
F1 F2 F3 F4 F5 F6
Received Lost Received Received Lost Received
2.4 kbps
1.2 kbps
2.4 kbps
1.2 kbps
2.4 kbps
1.2 kbps
F7 F8 F9 F10
F8 F9 F10 F9 F10 F11 F10F11 F12 F11F12 F13
F7 F8 F9 F10
Lost Lost Lost
2.4 kbps
1.2 kbps
Lost
Extrapolation
2.4 kbps 1.2 kbps 1.2 kbps 1.2 kbps
F10
F11F12 F13
F10
Received Received
2.4 kbps
1.2 kbps
F11 F12 F14 F15
F12 F13F14 F13F14 F15 F14F15 F16 F16F17 F18
F11 F12 F13 F14
Lost Lost Lost
2.4 kbps
1.2 kbps
F14
F15F16 F17
F15
2.4 kbps
Lost
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VI. SIMULATION RESULTS AND EVALUATION
First, we evaluate the performance of the two MELP coders
implemented separately; the aim is to quantify the perceptual
quality of these coders before they are implemented using MDC. A
second evaluation will be conducted using MDC.
A. EVALUATION CORPUS
To assess and validate our method, we used a multilingual corpus
combining Arabic, French and English. The first record is composed
of Arabic sentences phonetically balanced [7] developed in our
laboratory. This corpus contains a total of 60sentences, 10
sentences spoken by 3 male and 3 female speakers. For French and
English, we used the known phrases phonetically balanced, la bise
et le soleil and The wind and sun .
B. EVALUATION OF CODERS
The evaluation of the performance of the two MELP coders
implemented separately were designed using the Recommendation P.862
of the ITU-T (International Telecommunication Union) [8] called
PESQ (Perceptual Evaluation of Speech Quality). This method
describes an objective method for predicting the subjective quality
for telephony and for voice coders. It is intended to evaluate the
influence of factors such as packet loss, the variable delay and
distortion due to channel errors that is poorly evaluated by
conventional methods. The PESQ is designed to compare a reference
version (original) to that obtained by synthesizing this reference
or after transmission and have been adversely affected. The results
are shown in Table 2.
Concerning the assessment of the simulation of the technique
using MDC, we used different rates of packet loss using the model
random process of lost [9] and tested the robustness of the MELP
coder using the MDC. In this case, we found an enhancement in the
robustness of the system against packet loss. The results from the
objective tests of our simulation are shown in Figure 5. They
showed that our method based on a loss concealment technique for
VoIP application raises significant quality loss rates for up to
30%. These results show that the encoding technique used by
multiple descriptions provides a significant improvement in speech
quality, especially when the loss rate increases. We give in Figure
6, Figure 7 and Figure 8 a sample result which shows the
reconstructed signal after a packet loss when one, two and three
frames consecutives were lost respectively. We observe the
correction of lost frames for different cases of packet loss. For
areas where voiced and after the correction, the voicing is
preserved.
VII. CONDUCT OF TESTS
This section presents the simulations used to perform our tests.
We simulated various losses to introduce degradation in the
synthetic signal. These losses were simulated randomly by use the
function that follows a uniform distribution.
MELP 2.4 Kbps MELP 1.2 Kbps
PESQ 3.20 2.71
Table II: Results of objective tests of two MELP coders
Simulation of loss
PESQ Score
Coder Decoder
Evaluation
Original Signal
Reference signal
Synthetic signalcorrected by MDC
Signal with loss
Fig. 4: Results of objective tests of two MELP coders
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0 5 10 15 20 25 300
0.5
1
1.5
2
2.5
3
3.5
4
4.5
5
Packets loss (%)
PESQ
Original signalMELPMELP with MDCG.729
1.02 1.04 1.06 1.08 1.1 1.12 1.14 1.16 1.18x 104
1
-0.5
0
0.5
1.02 1.04 1.06 1.08 1.1 1.12 1.14 1.16 1.18 1.2x 104
-0.5
0
0.5
MELP signal at 2.4 kbps with packet loss
1.02 1.04 1.06 1.08 1.1 1.12 1.14 1.16 1.18 1.2x 10 4
1
-0.5
0
0.5
MELP signal with MDC
1
1.2
Am
plitu
de
Number of samples
Original signal
Am
plitu
de
Am
plitu
de
Fig. 5: Results of objective tests using PESQ
Fig. 6: Sample output showing the correction made by the MDC
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8200 8400 8600 880 9000 9200 940 9600 9800-1
-0.5
0
0.5
8200 8400 8600 8800 9000 9200 9400 9600 9800 10000
-0.5
0
0.5
8000 8200 8400 8600 8800 9000 9200 9400 9600 9800 10000
-0.5
0
0.5
Number of samples
Synthetic signal with MDC
Synthetic signal at 2.4 kbps with loss
Original signal
Am
plitu
de
Am
plitu
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Fig. 8: Sample output showing the correction made by the MDC
8000 8200 8400 8600 8800 9000 9200 9400 9600 9800 10000
-0.5
0
0.5
8000 8200 8400 8600 8800 9000 9200 9400 9600 9800 10000
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8000 8200 8400 8600 8800 9000 9200 9400 9600 9800 10000-1
-0.5
0
0.5
Am
plitu
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Am
plitu
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Synthetic signal at 2.4 kbps with loss
Synthetic signal with MDC
Number of samples
Original signal
Fig. 7: Sample output showing the correction made by the MDC
Am
plitu
de
Am
plitu
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VIII. CONCLUSION
In this work, we have presented an original method using two
harmonics MELP coders, the first for the transmission over an IP
network speech encoded at 2.4 kbps. The second operating at 1.2
kbps is added to the first in the same package in order to
compensate of the lost packets using a technique called multiple
description or MDC. The results of our simulations for a VoIP
application showed that our method based on this concealment method
enhances the quality for loss rates up to 30%. The results show
that the proposed system of concealment is effective and provides a
significant improvement in speech quality, especially outperforms
when the packet losses exceeded 15%. We have proved that
theredundancy added by MDC can ensure properly good quality speech
for any loss of packets. In this study, we obtained an encoder that
operates at 135 bits / frame of 22.5 mscorresponding to a total
rate of 6 kbps. So, we recommend the use of this solution to
replace advantageously the CELP G.729standard currently used in
VoIP applications and that has aflow rate of 8 kbps without
MDC.
REFERENCES[1] M. Rui and F. Labeau, Error-Resilient Multiple
Description
Coding. Proceedings of the IEEE, Vol 56, No 8, 2008
[2] E. Orozco, E. Orozco, and A.M.Kondoz. Multiple Description
Coding for Voice over IP using Sinusoidal Speech Coding. In Proc.
of IEEE- ICASSP, 2006.
[3] McCree, K. Truong, E. B. George, T. P. Barnwell, V.
Viswanathan. A 2.4 kbits/s MELP Coder Candidate for the New U.S.
Federal Standard. In Proc. of IEEE-ICASSP, 1996.
[4] T. Wang, K. Koishida, V. Cuperman and A. Gersho. A 1200bps
Speech Coder Based on MELP. Proc. IEEE. Inter. Conf. Acoustics.
Speech and Signal Processing, 2000.
[5] A. Nagle. Enrichissement de la Confrence audio en Voix sur
IP au travers de l'amlioration de la qualit et de la spatialisation
sonore. Paris Tech, France, 2008.
[6] L. Ouakil et G. Pujolle, Tlphonie sur IP . Edition groupe
Eyrolles, 2008.
[7] M. Boudraa, B. Boudraa, B. Gurin. Twenty lists of Ten Arabic
Sentenses for assessment. Acustica, vol. 86, pp.870-882, 2000.
[8]
[9]
ITU-T, Perceptual evaluation for speech quality (PESQ), an
objective method for end-to-end speech quality assessment of
narrow-band telephone networks and speech codecs, Recommendation
P.862, International Telecommunications Union, 2001.M. Yajnik, S.
Moon, J. Kurose, and D. Towsley. Measurement and modeling of the
temporal dependence in packet loss. In INFOCOM 99. Eighteenth
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Societies. Proceedings. IEEE, Vol 1, P. 345-352, 1999.
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