Audio and Video Codecs

While the cellular revolution has made voice connectivity ubiquitous in the developing world, data services are largely absent or are prohibitively expensive. In this paper, we present Hermes 1 , a point-to-point data connectivity solution that works by modulating data onto acoustic signals that are sent over a cellular voice call. The main challenge is that most voice codecs greatly distort signals that are not voice-like; furthermore, the backhaul can be highly heterogeneous and of low quality, thereby introducing unpredictable distortions. Hermes modulates data over the extremely narrow-band (approximately 3kHz bandwidth) acoustic carrier, while being severely constrained by the requirement that the resulting sound signals are voice-like, as far as the voice codecs are concerned. Hermes uses a robust data transcoding and modulation scheme to detect and correct errors in the face of bit flips, insertions and deletions; it also adapts the modulation parameters to the observed bit error rate on the actual voice channel. Through real-world experiments, we show that Hermes achieves approximately 1.2 kbps goodput which when compared to SMS, improves throughput by a factor of 5× and reduces the cost-per-byte by over a factor of 50×.

Voice over Internet Protocol (VoIP) may provide good services through Vehicular ad hoc networks
(VANETs) platform by providing services to many application scenarios range from safety to comfort.
However, VANETs networks introduce many challenges for supporting voice with QoS requirements. In
this paper, our study is based on Inter-Vehicle voice streaming rely on multi-hop fashion. For this task, a
performance evaluation of various audio CODECs will be analyzed by mean of simulations.
Furthermore, we test the impact of network environment on QoS metrics. To achieve good results,
CODECs behaviour is tested by using mobility information obtained from vehicular traffic generator. The
mobility model is based on the real road maps of an urban environment. Focusing on inter-vehicular
voice traffic quality, we provide simulations results in terms of both user level (MOS) metrics and
network level (such as Losses). According to this performance evaluation, we show that G.723.1 CODEC
worked well in the urban VANET environment.

This paper describes a method to randomly generate vectors of symbol probabilities so that the corresponding discrete memoryless source has a prescribed entropy. One application is to Monte Carlo simulation of the performance of noiseless variable length source coding.

Migration from IPv4 to IPv6 is still visibly slow, mainly because of the inherent cost involved in the implementation, hardware and software acquisition. However, there are many values IPv6 can bring to the IP enabled environment as compared to IPv4, particularly for Voice Over Internet Protocol (VoIP) solutions. Many companies are drifting away from circuit based switching such as PSTN to packet based switching (VoIP) for collaboration. There are several factors determining the effective utilization and quality of VoIP solutions. These include the choice of codec, echo control, packet loss, delay, delay variation (jitter), and the network topology. The network is basically the environment in which VoIP is deployed. State of art network design for VoIP technologies requires impeccable Interior Gateway routing protocols that will reduce the convergence time of the network, in the event of a link failure. Choice of CODEC is also a main factor. Since most research work in this area did not consider a particular CODEC as a factor in determining performance, this paper will compare the behaviour of RIP and OSPF in IPv4 and IPv6 using G.711 CODEC with riverbed modeller17.5.

In video compression class projects, students may observe some strange behaviors when using video codecs. Some performance metrics from a mediocre codec such as motion JPEG-2000 (or simply JPEG2000) may have exceptionally high values at certain compression ratios as compared to other high performing codecs. This strange behaviors may be overlooked by instructors and students may never understand why this is happening. In this paper, we will first highlight the strange behaviors. We will then use experiments to systematically determine the root cause. Our experiments show that, if one uses a previously compressed and decompressed video in some compression experiments, then it is highly likely that some strange behaviors will show up. Some advice will be provided to instructors, tutors, and students on how one can prevent such behaviors from occurring.

IEEE 802.11 is the most popular standard for WLAN networks. It offers different physical transmission rates. This paper focuses on this multi transmission rate of 802.11 WLANs and its effect on speech quality. In non-adaptive systems, when the physical layer switches from a higher transmission rate to a lower one, different than the one that the VoIP flow needs, the switching may result in congestion, high delay and packet loss, and consequently speech quality degradation. However, there are some algorithms that adapt the transmission parameters according to the channel conditions. In this study we demonstrate how choosing parameter (different codec and packet size) can affect the voice quality, network delay and packet loss. Further, this study presents a comparison between adaptive and non-adaptive methods. The adaptive method has also been evaluated for different congestion level from perceived speech quality point of view.

Esse artigo analisa o modo como a tensão entre os mecanismos que promovem a circulação do filme e os parâmetros midiáticos de sua identidade é articulada por meio de uma dupla codificação, simultaneamente algorítmica e jurídica. Para tanto, ele recorre às obras Download Finished (Sven König & !mediengruppe bitnik, 2006) e Artisa Sem Ideia (Cezar Migliorin, 2005).

The file size and picture quality are factors to be considered for streaming, storage and transmitting videos
over networks. This work compares Cinepak, Intel, Microsoft Video and Indeo Codec for video
compression. The peak signal to noise ratio is used to compare the quality of such video compressed using
AVI codecs. The most widely used objective measurement by developers of video processing systems is
Peak Signal-to-Noise Ratio (PSNR). Peak Signal to Noise Ration is measured on a logarithmic scale and
depends on the mean squared error (MSE) between an original and an impaired image or video, relative to
(2n-1)2.
Previous research done regarding assessing of video quality has been mainly by the use of subjective
methods, and there is still no standard method for objective assessments. Although it has been considered
that compression might not be significant in future as storage and transmission capabilities improve, but at
low bandwidths compression makes communication possible.

A compressão de dados é um dos factores que mais contribuiu para o grande crescimento das tecnologias da informação e da comunicação. Sem compressão, a maioria dos produtos tecnológicos de consumo e entretenimento, que são hoje banais, nunca teria chegado a existir, destacando-se, entre muitos outros, o DVD, as câmaras fotográficas digitais, os leitores de MP3, o YouTube e o streaming de vídeo, as redes sem fios e a televisão digital. De facto, as tecnologias de compressão multimédia permitem representar a informação de uma forma mais eficiente, reduzindo os grandes volumes de espaço de armazenamento que ocupa e, portanto, a largura de banda que consome para se transmitir nas redes e na Internet. Este livro tem como objectivo principal apresentar uma introdução fundamentada, clara, acessível e abrangente aos conceitos e tecnologias relacionadas com métodos, algoritmos e normas internacionais de compressão de informação multimédia. Ao longo do livro são abordados, entre outros, os seguintes temas: Conceitos e normas sobre compressão de informação multimédia, incluindo texto, imagens, áudio e vídeo; Entropia e algoritmos de compressão sem perdas: Shannon-Fano, Huffman, Aritmética e LZW; Psicoacústica, métodos e normas de compressão de áudio, incluindo voz (ADPCM, vocoders) e áudio de alta-fidelidade (transformadas, MP3, AAC, MP4 e Dolby AC-3); Teoria rate-distortion, métodos de compressão de imagem (DPCM, transformadas DCT, waveletet Haar, quantificadores lineares e não-lineares); Normas internacionais JPEG e JPEG2000; Compressão de vídeo, redundância espacial e temporal, quantificação de vídeo para aplicações multimédia e de videoconferência; Normas MPEG.