Conference Presentations by Mohamad Raad
11th International Conference on Telecommunication System, Services and Application (TSSA), 2017
this paper presents a wideband C-shaped patch antenna for LibyaSat-1. The two parallel slots of t... more this paper presents a wideband C-shaped patch antenna for LibyaSat-1. The two parallel slots of the upper C-shaped path are incorporated to generate a second resonant frequency and hence broaden bandwidth. The folded patch technique is used to reduce the coaxial probe length and inductance at the feed section. In addition, the Quasi Newton method is used to achieve an operating frequency of 2.215 GHz (S-band). Our simulation results show that the antenna achieves a-10-dB impedance bandwidth of 1550 MHz (2.00-3.55 GHz), and has a total gain of 6.45 dB at 2.215GHz.
Papers by Mohamad Raad
Scalable to lossless audio compression based on perceptual set partitioning in hierarchical trees... more Scalable to lossless audio compression based on perceptual set partitioning in hierarchical trees (PSPIHT)
Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)
This paper describes the use of sorted sinusoidal parameters to produce a fixed rate, scalable, w... more This paper describes the use of sorted sinusoidal parameters to produce a fixed rate, scalable, wideband audio coder. The sorting technique relies on the perceptual significance of the sinusoidal parameters. Sinusoidal coding permits the representation of a given signal through the summation of sinusoids. The parameters of the sinusoids (the amplitudes, phases and frequencies) are transmitted to allow signal reconstruction. The sinusoidal parameters are sorted according to energy content and perceptual significance. The most significant parameters are transmitted first, allowing the use of only a small set of the parameters for signal reconstruction. The proposed scheme incurs a low delay and uses a 20 ms frame length. The results presented show the advantages gained for scalable audio coding by sorting the parameters.
School of Engineering-Department of Computer and Communications Engineering Lebanese International University (LIU) PO Box: 146404 Mazraa, Beirut, Lebanon
Modified Vector Base Amplitude Panning implementation on smart devices
2013 13th International Symposium on Communications and Information Technologies (ISCIT), 2013
ABSTRACT This paper describes an extension to Vector Base Amplitude Panning (VBAP) [7]. An audien... more ABSTRACT This paper describes an extension to Vector Base Amplitude Panning (VBAP) [7]. An audience targeting vector is introduced to the VBAP model to adjust the audio playback system according to the position of the audience. With this modification, it is possible to create virtual sound sources at positions that are better suited for the location of the target audience. Mobile applications such as mobile teleconferencing can take advantage of the surround sound technology to enhance the audio quality for users. In mobile surround, surround sound is produced by mobile telephone handsets, the locations of both the handsets and the audiences may vary during the play back. Therefore, we developed this vector based approach to allow us to move the loudspeakers and the audience independently from each other.
This paper describes the use of sorted sinusoidal parameters to produce a fixed rate, scalable, w... more This paper describes the use of sorted sinusoidal parameters to produce a fixed rate, scalable, wideband audio coder. The sorting technique relies on the perceptual significance of the sinusoidal parameters. Sinusoidal coding permits the representation of a given signal through the summation of sinusoids. The parameters of the sinusoids (the amplitudes, phases and frequencies) are transmitted to allow signal reconstruction. The sinusoidal parameters are sorted according to energy content and perceptual significance. The most significant parameters are transmitted first, allowing the use of only a small set of the parameters for signal reconstruction. The proposed scheme incurs a low delay and uses a 20 ms frame length. The results presented show the advantages gained for scalable audio coding by sorting the parameters
This paper extends a scalable to lossless compression scheme to allow scalability in terms of sam... more This paper extends a scalable to lossless compression scheme to allow scalability in terms of sampling rate as well as quantization resolution. The scheme presented is an extension of a perceptually scalable scheme that scales to lossless compression, producing smooth objective scalability, in terms of SNR, until lossless compression is achieved. The scheme is built around the Perceptual SPIHT algorithm, which is a modification of the SPIHT algorithm. An analysis of the expected limitations of scaling across sampling rates is given as well as lossless compression results showing the competitive performance of the presented technique.
International Journal of Engineering Research and Applications
The Transport Control Protocol (TCP) is used to establish and control a session between two endpo... more The Transport Control Protocol (TCP) is used to establish and control a session between two endpoints. The problem is that in 802.11 wireless environments TCP always considers that the packet loss is caused by network congestion. However, in these networks packet loss are usually caused by the high bit error rate, and the wireless link failures. Researchers found out that TCP performance in wireless networks can be highly enhanced as long as it is feasible to identify the packet loss causes; hence appropriate measures can be dynamically applied during an established TCP session in order to adjust the session parameters. This paper proposes an endto-end adaptive mechanism that allows the TCP session to dynamically adjust the RTO (Retransmission Timeout) of a TCP session; the server will have to adjust the timers based on feedbacks from clients. Feedbacks are piggybacked in the TCP Options header field of the ACK (Acknowledgment) messages. A feedback is an approximation of the time needed by the wireless channel to get the errors fixed. The mechanism has been validated using numerical analysis and simulations, and then compared to the original TCP protocol. Simulation results have shown better performance in terms of number of retransmissions at the server side due to the decrease in the number of timeouts; and thus lowest congestion on the wireless access point.
The Transport Control Protocol (TCP) is used to establish and control a session between two endpo... more The Transport Control Protocol (TCP) is used to establish and control a session between two endpoints. The problem is that in 802.11 wireless environments TCP always considers that the packet loss is caused by network congestion. However, in these networks packet loss are usually caused by the high bit error rate, and the wireless link failures. Researchers found out that TCP performance in wireless networks can be highly enhanced as long as it is feasible to identify the packet loss causes; hence appropriate measures can be dynamically applied during an established TCP session in order to adjust the session parameters. This paper proposes an end-to-end adaptive mechanism that allows the TCP session to dynamically adjust the RTO (Retransmission Timeout) of a TCP session; the server will have to adjust the timers based on feedbacks from clients. Feedbacks are piggybacked in the TCP Options header field of the ACK (Acknowledgment) messages. A feedback is an approximation of the time n...
This paper discusses the application of the Set Partitioning In Hierarchical Trees (SPIHT) algori... more This paper discusses the application of the Set Partitioning In Hierarchical Trees (SPIHT) algorithm to the compression of audio signals. Simultaneous masking is used to reduce the number of coefficients required for the representation of the audio signal. The proposed scheme is based on the combination of the Modulated Lapped Transform (MLT) and SPIHT. Comparisons are also made with the Discrete Wavelet Transform (DWT) based scheme. Results presented reveal the compression achieved as well as the scalability of the proposed coding scheme. The MLT based scheme is shown to have compression performance that is superior to the DWT based scheme.
Multi-rate extension of the scalable to lossless PSPIHT audio coder
Interspeech, 2003
Scalable and perceptual audio compression
This paper discusses the design and implementation of a scalable audio compression scheme that sc... more This paper discusses the design and implementation of a scalable audio compression scheme that scales up from lossy to lossless compression. Scalable audio compression has been of interest in the audio compression community for some time, with the most obvious attempt at obtaining a solution coming in the form of the MPEG-4 standard [1]. At the same time the increase in bit rates in both mobile communications [2] and the internet's broadband technology means that audio compression algorithms with higher bit rates than currently used, such as MPEG's mp3 [1], can be employed to obtain higher quality. However, the new increased data rates are not necessarily constant, this is especially the case when considering the internet. As such, scalable schemes that can scale to lossless compression have become rather interesting from an application point of view. The scheme presented in this paper achieves lossless compression that is comparable with the state of the art whilst maintain...
Methods and apparatus for combining session acceleration techniques for media oriented negotiation acceleration
Method and apparatus of voice mixing for conferencing amongst diverse networks
2011 IEEE International Conference on Multimedia and Expo, 2011
A video rate control mechanism based on a modified LMS algorithm is presented. The control mechan... more A video rate control mechanism based on a modified LMS algorithm is presented. The control mechanism is used within a real-time IP multicast based system. Simulation results show that such a control mechanism allows the system to adapt to the varying network conditions of receiving clients.
This paper presents a scalable to lossless compression scheme that allows scalability in terms of... more This paper presents a scalable to lossless compression scheme that allows scalability in terms of sampling rate as well as quantization resolution. The scheme presented is perceptually scalable and it also allows lossless compression. The scheme produces smooth objective scalability, in terms of SNR, until lossless compression is achieved. The scheme is built around the perceptual SPIHT algorithm, which is a modification of the SPIHT algorithm. Objective and subjective results are given that show perceptual as well as objective scalability. The subjective results given also show that the proposed scheme performs comparably with the MPEG-4 AAC coder at 16, 32 and 64 kbps.
This paper presents an audio coder based on the combination of the Modulated Lapped Transform (ML... more This paper presents an audio coder based on the combination of the Modulated Lapped Transform (MLT) with the Set Partitioning In Hierarchical Trees (SPIHT) algorithm. SPIHT allows scalable coding by transmitting more important information first in an efficient manner. The results presented reveal that the Modulated Lapped Transform (MLT) based scheme produces a high compression ratio for little or no loss of quality. A modification is introduced to SPIHT which further improves the performance of the algorithm when it is being used with uniform M-band transforms and masking. Further, the MLT-SPIHT scheme is shown to achieve high quality synthesized audio at 54 kbps through subjective listening tests.
This paper proposes a technique for scalable and also provides for lossless compression. It reduc... more This paper proposes a technique for scalable and also provides for lossless compression. It reduces smooth objective scalability, in terms of SegSNR, from lossy to lossless compression. The proposal is built around the perceptual SPIHT algorithm, which is a modification of the SPIHT algorithm and is introduced in this paper. Both objective and subjective results are reported and demonstrate that the proposed method performs comparably with the MPEG-4 AAC coder at 16, 32 and 64 kbps, yet also achieves a scalable-to-lossless architecture.
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Conference Presentations by Mohamad Raad
Papers by Mohamad Raad