Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.

Wavelet based scalable video codec

Abstract: With the increase in network speeds for both wired and wireless networks, video streaming has become a reality. But different networks are characterized by different network speeds and latency. Also different kinds of devices are used with different networks. These devices range from Desktop/Laptop computers with very high resolution and very high processing power to Mobile Phones with small displays, low processing power and limited battery life. A single common non-scalable video feed for all these different profiles of networks and devices is impractical as streaming a high quality feed may not be possible over GPRS type of networks and streaming a low quality feed may not be visually acceptable to a user with high network bandwidth and high resolution display. To overcome this problem multiple video feeds are used, each catering to a different profile of users. Content creation process becomes complex as multiple files have to be created usually using a different codec and file format. Deployment involves replication of hardware and software infrastructure and associated overhead of maintenance, hence increased cost.

Screen image sequence compression method utilizing adaptive block size coding and hierarchical GOP structure

Abstract: To compress screen image sequence in real-time remote and interactive applications, a novel compression method is proposed. The proposed method is named as CABHG. CABHG employs hybrid coding schemes that consist of intra-frame and inter-frame coding modes. The intra-frame coding is a rate-distortion optimized adaptive block size that can be also used for the compression of a single screen image. The inter-frame coding utilizes hierarchical group of pictures (GOP) structure to improve system performance during random accesses and fast-backward scans. Experimental results demonstrate that the proposed CABHG method has approximately 47%–48% higher compression ratio and 46%–53% lower CPU utilization than professional screen image sequence codecs such as TechSmith Ensharpen codec and Sorenson 3 codec. Compared with general video codecs such as H.264 codec, XviD MPEG-4 codec and Apple’s Animation codec, CABHG also shows 87%–88% higher compression ratio and 64%–81% lower CPU utilization than these general video codecs.

High sampling rate audio signal modeling

Abstract: In this paper, we study the issue of the high sampling rate audio modeling for lossless audio coding. We propose a cascade LMS structure to successfully model all high sampling rate audio signals. This cascade structure predictor, not only performs better than its counterpart FTR linear prediction coding (LPC) technique in modeling general audio signals, but also displays a faster convergence and smaller mean square error (MSE) than conventional LMS predictor and low-order stages cascade LMS predictor, while the complexity of the proposed predictor remains simple. The simulation results show that the proposed structure gets better prediction gain compared with Monkey's audio codec and MPEG-4 ALS codec provided by Technology University of Berlin (TUB) for real high sampling rate audio test set. Other adaption algorithms can be used for the single stages.

Wireless videophone schemes

Abstract: A range of 5-11.36 kbps videophone codecs are proposed and the 11.36 kbps codec 1, the 8.52 kbps codec 2 and the 8 kbps codec 2a are embedded in the intelligent re-configurable systems 1-3. After sensitivity-matched binary Bose-Chaudhuri-Hocquenghem (BCH) forward error correction (FEC) coding the data rate associated with codec 1 and codec 2a became 20.32 kbps, while that of codec 2 was 15.24 kbps. When using codec 1 in system 1 and coherent pilot symbol assisted 16-level quadrature amplitude modulation (16-PSAQAM), an overall signalling rate of 9 kBd was yielded. Over lower quality channels the 4QAM mode of operation had to be invoked, which required twice as many time slots to accommodate the resulting 18 kBd stream. In a bandwidth of 200 kHz, similarly to the Pan-European GSM mobile radio system's speech channel, using systems 1 and 3 for example, 16 and 8 videophone users can be supported in the 16 QAM and 4QAM modes, respectively.

Perbandingan Kualitas Layanan Wireless VOIP pada Codec G.711, G.723 dan G.729

Abstract: Voice over Internet Protocol (VoIP) is a technology that enable voice message transmission over data network (internet protocol). Codec is algorithm or special computer program to reduce number of bytes. The usage of appropriate codec at implementation VoIP is one thing determining in attainment of quality VoIP communications. This research implement and analys the usage of codec G711, G723 and G729 at protocol H323 for VoIP service. Examination by through simulation using ns- 2. Codec G723 has smaller delay value and jitter value than codec G711 and G729. The result from simulation is codec G723 has smaller bit rate value, so more efficient for implementation at network that having not big bandwidth or network capacity. In communications process usage of