Topic
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.
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TL;DR: A novel extension to wavelet transformed image and video codec based on SPIHT is presented for the purpose of surveillance where the detail information regarding a scene is not an essential requirement and a pre–processing scheme, called Zero–Shifting, has been proposed using which calculation of transformed coefficients becomes more consistent.
Abstract: A novel extension to wavelet transformed image and video codec based on SPIHT is presented for the purpose of surveillance where the detail information regarding a scene is not an essential requirement. A pre–processing scheme, called Zero–Shifting, has been proposed using which calculation of transformed coefficients becomes more consistent. This modification leads to a significantly higher PSNR and better visual quality at a specific bit rate with little additional computational complexity. We achieved 1.5–4.2 dB more gain at the lowest possible bit rate without using entropy coder, which provides only 0.3–0.6 dB increase in gain at the cost of higher computational complexity and reduced speed of the SPIHT codec.
1 citations
01 Oct 2016
TL;DR: An encoder based H.265/HEVC 4K slim codec that shares the prediction unit of encoder except an entropy decoder and saves the total size by 40 % compared to an independent codec with encoder and encoder separately.
Abstract: H.265/High Efficiency Video Coding (HEVC) is the latest next generation video compression standard posterior to H.264/AVC. However, despite its superior coding efficiency to the previous video coding standards, the complexity to implement it is an obstacle to overcome. Especially, combining the separate encoder and decoder has a disadvantage on the aspect of the size and power consumption. To solve these problems, we design an encoder based H.265/HEVC 4K slim codec. The decoder within this codec shares the prediction unit of encoder except an entropy decoder. The proposed shared prediction unit architecture saves the total size by 40 % compared to an independent codec with encoder and encoder separately. The size of logic is 2.8 M gates with 120 kB internal SRAM and the power consumption of this slim codec is within a level of encoder. The function of slim codec is verified on our designed the Xilinx Virtex-7 platform and the 4K UHD codec chip operating at 600 MHz is going to implement on a 28 nm CMOS process in this year.
1 citations
Journal Article•
TL;DR: Experimental results show that the real-time processing of video can be realized while the high image quality and compression efficiency are preserved.
Abstract: The hardware structure and the software task flowchart of MPEG-4 codec based on TMS320DM642 are briefly introduced.Then the transplantation and optimization of video codec on DM642 is mainly discussed.Experimental results show that the real-time processing of video can be realized while the high image quality and compression efficiency are preserved.
1 citations
TL;DR: A hybrid parametric-perceptual speech codec architecture is presented and the signal quality achieved using the hybrid codec is compared to the quality of some standard speech codecs.
Abstract: A hybrid parametric-perceptual speech codec architecture is presented. The basic CELP parametric codec structure is enhanced using the perceptual coding method. The objective of the codec hybridisation is obtaining significant improvement in the perceived signal quality. Two hybrid architectures are proposed. The first one encodes voiced parts perceptually in the CELP residual signal. The second one divides the signal into voiced signal components that are encoded using the perceptual algorithm, unvoiced components that are encoded parametrically and transients remaining unencoded. The signal quality achieved using the hybrid codec is compared to the quality of some standard speech codecs.
1 citations
NEC1
TL;DR: In this paper, an adaptive transform coding (ATC)-based Hi-Fi audio coding method for asynchronous transfer mode (ATM) networks is proposed, which assigns higher/lower priority to most/least significant bits, rather than to lower/higher frequency bands, as is commonly done in video coding.
Abstract: The authors propose an adaptive transform coding (ATC)-based hi-fi audio coding method for asynchronous transfer mode (ATM) networks. To maintain robustness despite cell loss, a novel layered coding scheme has been employed in ATC. In this approach, higher/lower priority is assigned to most/least significant bits, rather than to lower/higher frequency bands, as is commonly done in video coding. The number of bits for most/least significant bits is determined on the basis of two different coding-rates. Subjective test results for 192 kbps/96 kbps coding show that, even under conditions of cell loss, it is possible to maintain a level of quality equivalent to 192 kbps coding quality. The proposed approach appears to be promising for high-quality hi-fi audio signal transmission in ATM networks. >
1 citations