Audio and Video Codecs

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.

Recordings for acoustic research should ideally be made in a lossless format. However, in some cases pre-existing data may be available in a lossy format such as mp3, prompting the question in how far this compromises the accuracy of acoustic measurements. In order to determine whether this is the case, we compressed 10 recordings of read speech in different compression rates (16-320 kbps), and reconverted them to wav in order to examine the effect of compression on commonly used suprasegmental measures of fundamental frequency (f0), pitch range and level. Results suggest that at compression rates between 56 and 320 kbps, measures of f0 and most measures of pitch range and level remain reliable, with mean errors below 2% and often better than that. The skewness of the distribution of f0 measurements , however, shows much greater measurement errors, with mean errors of 6.9%-7.6% at compression rates between 96 kbps and 320 kbps, and 44.8% at 16 kbps. We conclude that mp3 compressed recordings can be subjected to the acoustic measurements tested here. Nevertheless, the indeterminacy added by mp3 compression needs to be taken into account when interpreting measurements.

The International Journal of Computer Networks & Communications (IJCNC) is a bi monthly open access peer-reviewed journal that publishes articles which contribute new results in all areas of Computer Networks & Communications.The journal focuses on all technical and practical aspects of Computer Networks & data Communications. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced networking concepts and establishing new collaborations in these areas.

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 wireless communication system, limited bandwidth and power is the primary restriction. The existing wireless systems involved in transmission of speech visualized that efficient and effective methods be developed to transmit and receive the same while maintaining quality of speech, especially at the receiving end. Speech coding technique is a material of research for the scientific and academic community since the era of digitization (digital). Amongst all elements of the communication systems (transmitter, channel and receiver), transmission channel is the most critical and plays a key role in the transmission and reception of information. The quality of speech at receiver end decides by channel conditions. Modelling a channel is a multifarious task. A number of techniques are adopted to alleviate the effect of the channel. Adaptive Multi Rate is one of the techniques that neutralize the deleterious effect of the channel on speech. This technique utilizes variable bit rate that dynamically switches to specific modes of operation depending upon the channel conditions. For example, Low bit rate mode of operation is selected in adverse channel conditions, this helps to provide more error protection bits for channel coding and vice versa. Therefore, in this paper, application of Code Excited Linear Prediction (CELP) source codec on speech followed by AMR codec is studied. Further, higher the bit rate used, the better is the quality of speech. In this paper apart from speech codec about AMR is also studied that why the AMR is proposed for the GSM, how the bits rates are reduced in AMR, operation of AMR and other applications of AMR.

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.

Conference control has been studied for years but most researches focus on homogenous A/V collaboration. There is no conference control framework for integration of multiple videoconferencing systems such as H.323, SIP and AccessGrid. In this paper, we propose a web-serviced based scalable conference control framework for such heterogeneous collaboration system. Based on XGSP A/V Web-Services framework, we propose a Global Multimedia Collaboration System (Global-MMCS), which can integrate multiple collaboration systems, and support various collaboration clients and communities. Now the prototype is being developed and deployed across many universities in USA and China.

In order to fit the small cross-center-biased characteristic of the real world video sequences, an improved version of the well-known cross diamond search algorithm is proposed in this paper. The new algorithm uses a small cross-shaped search patterns in the first two steps to speedup the motion estimation of stationary and quasi-stationary blocks. Experimental results show that this new cross-diamond search algorithm could achieve much higher computational reduction as compared with Diamond Search (DS) and Cross Diamond Search (CDS) while similar prediction accuracy is maintained, and it is especially suitable for videoconferencing sequences. Keywords—Motion estimation, new cross-diamond search, cross-center biased characteristic.

Este artigo contém informações e conhecimentos da tecnologia Voz sobre IP. Com ênfase na codificação do sinal de voz, no estudo dos parâmetros de QoS abordando os  Serviços Diferenciados (DiffServ), e os tipos de Redes de Comutação. Métodos de avaliação de codec, dividindo em testes subjetivos e objetivos

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.

In the last decade mobile phones have been evolved rapidly . Previously the main objective of these devices is a voice call , nowadays they provide increasingly powerful services such as (Web browsing, Playback Video, Gaming, SMS text messaging, etc…). Using these rich services mobile phone, that is powered from battery, become consuming more and more energy especially when dealing with video services. This paper presents implementation of playing back local and downloaded video with different CODECs in mobile phone. Moreover the paper will presents measurements and analysis of power consumption, CPU and RAM usage resources Measurements conducted on mobile phones based on Symbian platform. The results show that different CODECs as well as CPU&RAM resources affected directly to battery consumption during playback video in mobile phone. J2ME is the programing language that will be adopted.

https://sites.google.com/site/ijcsis/vol-11-no-5-may-2013

El consumo de contenidos audiovisuales a través de la red ha experimentado un espectacular crecimiento desde que aparecieron los primeros portales de vídeo en torno al 2005. Este desarrollo está directamente relacionado con la ampliación de la capacidad de transmisión de las redes. A principios de siglo las conexiones alcanzaban un ancho de banda de 56 Kbps que, si lo comparamos con el de las actuales, podremos comprobar que ha multiplicado exponencialmente su capacidad en un periodo de tiempo relativamente corto.

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).

In digital forensics, recovery of a damaged or altered video file plays a crucial role in searching for evidences to resolve a criminal case. This paper presents a frame-based recovery technique of a corrupted video file using the specifications of a codec used to encode the video data. A video frame is the minimum meaningful unit of video data. Many existing approaches attempt to recover a video file using file structure rather than frame structure. In case a target video file is severely fragmented or even has a portion of video overwritten by other video content, however, video file recovery of existing approaches may fail. The proposed approach addresses how to extract video frames from a portion of video to be restored as well as how to connect extracted video frames together according to the codec specifications. Experiment results show that the proposed technique successfully restores fragmented video files regardless of the amount of fragmentations. For a corrupted video file containing overwritten segments, the proposed technique can recover most of the video content in non-overwritten segments of the video file.

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×.

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.