2,203 research outputs found
Unequal Error Protected JPEG 2000 Broadcast Scheme with Progressive Fountain Codes
This paper proposes a novel scheme, based on progressive fountain codes, for
broadcasting JPEG 2000 multimedia. In such a broadcast scheme, progressive
resolution levels of images/video have been unequally protected when
transmitted using the proposed progressive fountain codes. With progressive
fountain codes applied in the broadcast scheme, the resolutions of images (JPEG
2000) or videos (MJPEG 2000) received by different users can be automatically
adaptive to their channel qualities, i.e. the users with good channel qualities
are possible to receive the high resolution images/vedio while the users with
bad channel qualities may receive low resolution images/vedio. Finally, the
performance of the proposed scheme is evaluated with the MJPEG 2000 broadcast
prototype
Ontology based approach for video transmission over the network
With the increase in the bandwidth & the transmission speed over the
internet, transmission of multimedia objects like video, audio, images has
become an easier work. In this paper we provide an approach that can be useful
for transmission of video objects over the internet without much fuzz. The
approach provides a ontology based framework that is used to establish an
automatic deployment of video transmission system. Further the video is
compressed using the structural flow mechanism that uses the wavelet principle
for compression of video frames. Finally the video transmission algorithm known
as RRDBFSF algorithm is provided that makes use of the concept of restrictive
flooding to avoid redundancy thereby increasing the efficiency.Comment: 7 pages, 2 figures, 4 table
Scalable Speech Coding for IP Networks
The emergence of Voice over Internet Protocol (VoIP) has posed new challenges to the development of speech codecs. The key issue of transporting real-time voice packet over IP networks is the lack of guarantee for reasonable speech quality due to packet delay or loss.
Most of the widely used narrowband codecs depend on the Code Excited Linear Prediction (CELP) coding technique. The CELP technique utilizes the long-term prediction across the frame boundaries and therefore causes error propagation in the case of packet loss and need to transmit redundant information in order to mitigate the problem. The internet Low Bit-rate Codec (iLBC) employs the frame-independent coding and therefore inherently possesses high robustness to packet loss. However, the original iLBC lacks in some of the key features of speech codecs for IP networks: Rate flexibility, Scalability, and Wideband support.
This dissertation presents novel scalable narrowband and wideband speech codecs for IP networks using the frame independent coding scheme based on the iLBC. The rate flexibility is added to the iLBC by employing the discrete cosine transform (DCT) and iii the scalable algebraic vector quantization (AVQ) and by allocating different number of bits to the AVQ. The bit-rate scalability is obtained by adding the enhancement layer to the core layer of the multi-rate iLBC. The enhancement layer encodes the weighted iLBC coding error in the modified DCT (MDCT) domain. The proposed wideband codec employs the bandwidth extension technique to extend the capabilities of existing narrowband codecs to provide wideband coding functionality. The wavelet transform is also used to further enhance the performance of the proposed codec.
The performance evaluation results show that the proposed codec provides high robustness to packet loss and achieves equivalent or higher speech quality than state-of-the-art codecs under the clean channel condition
Scalable and perceptual audio compression
This thesis deals with scalable perceptual audio compression. Two scalable perceptual solutions as well as a scalable to lossless solution are proposed and investigated. One of the scalable perceptual solutions is built around sinusoidal modelling of the audio signal whilst the other is built on a transform coding paradigm. The scalable coders are shown to scale both in a waveform matching manner as well as a psychoacoustic manner. In order to measure the psychoacoustic scalability of the systems investigated in this thesis, the similarity between the original signal\u27s psychoacoustic parameters and that of the synthesized signal are compared. The psychoacoustic parameters used are loudness, sharpness, tonahty and roughness. This analysis technique is a novel method used in this thesis and it allows an insight into the perceptual distortion that has been introduced by any coder analyzed in this manner
Scalable Video Coding
International audienceWith the evolution of Internet to heterogeneous networks both in terms of processing power and network bandwidth, different users demand the different versions of the same content. This has given birth to the scalable era of video content where a single bitstream contains multiple versions of the same video content which can be different in terms of resolutions, frame rates or quality. Several early standards, like MPEG2 video, H.263, and MPEG4 part II already include tools to provide different modalities of scalability. However, the scalable profiles of these standards are seldom used. This is because the scalability comes with significant loss in coding efficiency and the Internet was at its early stage. Scalable extension of H.264/AVC is named scalable video coding and is published in July 2007. It has several new coding techniques developed and it reduces the gap of coding efficiency with state-of-the-art non-scalable codec while keeping a reasonable complexity increase. After an introduction to scalable video coding, we present a proposition regarding the scalable functionality of H.264/AVC, which is the improvement of the compression ratio in enhancement layers (ELs) of subband/wavelet based scalable bitstream. A new adaptive scanning methodology for intra frame scalable coding framework based on subband/wavelet coding approach is presented for H.264/AVC scalable video coding. It takes advantage of the prior knowledge of the frequencies which are present in different higher frequency subbands. Thus, by just modification of the scan order of the intra frame scalable coding framework of H.264/AVC, we can get better compression, without any compromise on PSNR
Distributed video coding for wireless video sensor networks: a review of the state-of-the-art architectures
Distributed video coding (DVC) is a relatively new video coding architecture originated from two fundamental theorems namely, Slepian–Wolf and Wyner–Ziv. Recent research developments have made DVC attractive for applications in the emerging domain of wireless video sensor networks (WVSNs). This paper reviews the state-of-the-art DVC architectures with a focus on understanding their opportunities and gaps in addressing the operational requirements and application needs of WVSNs
Adaptive Variable Degree-k Zero-Trees for Re-Encoding of Perceptually Quantized Wavelet-Packet Transformed Audio and High Quality Speech
A fast, efficient and scalable algorithm is proposed, in this paper, for
re-encoding of perceptually quantized wavelet-packet transform (WPT)
coefficients of audio and high quality speech and is called "adaptive variable
degree-k zero-trees" (AVDZ). The quantization process is carried out by taking
into account some basic perceptual considerations, and achieves good subjective
quality with low complexity. The performance of the proposed AVDZ algorithm is
compared with two other zero-tree-based schemes comprising: 1- Embedded
Zero-tree Wavelet (EZW) and 2- The set partitioning in hierarchical trees
(SPIHT). Since EZW and SPIHT are designed for image compression, some
modifications are incorporated in these schemes for their better matching to
audio signals. It is shown that the proposed modifications can improve their
performance by about 15-25%. Furthermore, it is concluded that the proposed
AVDZ algorithm outperforms these modified versions in terms of both output
average bit-rates and computation times.Comment: 30 pages (Double space), 15 figures, 5 tables, ISRN Signal Processing
(in Press
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